snom ONE Technical Manual - Atcom

Deploying the 

snom ONE IP Telephone System 

Technical Manual 

VoIPon www.voipon.co.uk sales@voipon.co.uk Tel: +44 (0)1245 808195 Fax: +44 (0)1245 808299

Copyright © 2010 snom technology, Inc. 

All rights reserved. 

snom® is a registered trademark of snom technology AG and its affiliates in the European Union, USA, 

Japan, South Africa, Australia, China, and certain other countries and regions. Unless specified otherwise, 

all trademarks, in particular product names, are legally protected trademarks of snom technology AG. Other 

mentioned trademarks or registered trademarks are the property of their respective manufacturers or owners. 

Product specifications are subject to change without notice. 


BRIEF CONTENTS 

Part I—Getting Started 

1 Getting Started ....................................................................................................................1 

Part II—Administering the System 

2 System Settings .................................................................................................................15 

3 Administering the Domain ................................................................................................49 

4 Trunks ..................................................................................................................................67 

5 Dial Plans ............................................................................................................................91 

6 Extensions ........................................................................................................................ 101 

7 Service Flags ................................................................................................................... 111 

8 Auto Attendant ................................................................................................................ 121 

9 Hunt Groups .................................................................................................................... 135 

10 Agent Groups .................................................................................................................. 141 

11 Paging............................................................................................................................... 157 

12 Conferencing ................................................................................................................... 163 

13 IVR Node .......................................................................................................................... 171 

14 Calling Card Account .................................................................................................... 179 

15 Pre-Pay Feature .............................................................................................................. 183 

16 Email ................................................................................................................................. 187 

17 Audio and Greetings ...................................................................................................... 197 

18 Buttons ............................................................................................................................. 209 

19 Plug and Play ................................................................................................................... 219 

20 Music on Hold ................................................................................................................. 241 

21 Call Detail Records (CDRs).......................................................................................... 251 

Part III—The User Interface 

22 Web Interface .................................................................................................................. 261 

23 Star Codes ...................................................................................................................... 279 

24 Voicemail .......................................................................................................................... 295 

25 Cell Phones ..................................................................................................................... 303 

Appendix A. Working with CSV Files ................................................................................. 315 

Appendix B. SIP Overview ................................................................................................... 331 

Appendix C. snom ONE and Exchange ............................................................................. 351 

Glossary ...................................................................................................................................... 357 

Index 369 



CONTENTS 


1 Getting Started 1 

About snom ONE . ................................................................................1 

Memory/Hardware Requirements . ......................................................2 

USB Device . .......................................................................................2 

Warnings .....................................................................................2 

Standards Conformance . .................................................................2 

Disposal . ......................................................................................3 

Installing the Software . ..........................................................................3 

Windows Installations .....................................................................3 

Logging In . .........................................................................................5 

Login Levels . .................................................................................5 

Activating Your License . ........................................................................7 

Upgrading the Software . ........................................................................7 

Manual Software Upgrades ..............................................................8 

Restarting the System . ..........................................................................8 

Restarting From Windows (Service Control Manager) ............................8 

Restarting From a Command Line . .....................................................9 

When the System Fails to Restart . ................................................... 10 

Technical Support . ............................................................................. 10 

Documentation . ................................................................................. 11 


2 System Settings 15 

Directory Structure . ............................................................................ 17 

Directory Tree. ............................................................................. 17 

Global Configuration File ............................................................... 17 

Folder Overview . .......................................................................... 18 

Folder Details . ............................................................................. 19 

Configuring System Settings . ............................................................... 23 

General . ..................................................................................... 23 

Administrator Login. ...................................................................... 24 

Appearance . ............................................................................... 25 

Performance. ............................................................................... 28 

SIP Settings . ............................................................................... 29 

Ports . .............................................................................................. 31 

HTTP Ports . ................................................................................ 31 

SIP Ports. ................................................................................... 31 

RTP Ports . .................................................................................. 33 

SNMP . ....................................................................................... 34 

TFTP . ........................................................................................ 35 

Logging . .......................................................................................... 36 


vi 

Deploying the snom ONE IP Telephone System 

General Logging . ......................................................................... 36 

Specific Events . ........................................................................... 37 

SIP Logging . ............................................................................... 38 

Configuration . ................................................................................... 39 

Save/Restore a Backup . ................................................................ 40 

Request Configuration . .................................................................. 40 

Reload Configuration Files ............................................................. 41 

Schedule a Reboot . ...................................................................... 41 

Certificate . ....................................................................................... 41 

Access List . ...................................................................................... 43 

Controlling User Page Appearance . ....................................................... 45 

General Control . .......................................................................... 46 

Redirection Control ...................................................................... 47 

Mailbox Control . ........................................................................... 47 

Email Control .............................................................................. 48 

3 Administering the Domain 49 

Naming the Domain . ........................................................................... 49 

Changing the Name of the Domain . .................................................. 50 

Assigning a Domain Administrator . ......................................................... 50 

Default Domain . ........................................................................... 52 

Configuring the Domain . ...................................................................... 53 

Accounts Page Display . ................................................................. 58 

Recording Defaults . ...................................................................... 60 

Email Settings . ............................................................................ 60 

Midnight Events . .......................................................................... 61 

Domain Address Book . ........................................................................ 63 

Creating a Domain Address Book ................................................... 63 

Searching for Contacts . ................................................................. 65 

4 Trunks 67 

Trunk Types . ..................................................................................... 68 

SIP Registrations ......................................................................... 68 

SIP Gateway . .............................................................................. 69 

Outbound Proxy . .......................................................................... 69 

Inbounds Calls . ................................................................................. 70 

How the System Identifies a Trunk ................................................... 70 

How the System Routes a Call to the Proper Extension . ............................. 72 

Basic Routing . ............................................................................. 72 

When Regular Expressions are Involved . ........................................... 73 

Regular Expressions and Direct Inward Dialing (DID) . ........................... 75 

Outbound Calls . ................................................................................ 77 

Caller-ID . ................................................................................... 77 

Generating the ANI . ...................................................................... 77 

Representing the Source ............................................................... 78 

Creating Trunks . ................................................................................ 79 

Configuring Trunks . ............................................................................ 80 

General Settings . ......................................................................... 80 

Outbound Settings . ...................................................................... 84 


Contents vii 

Inbound Settings . ......................................................................... 86 

Connecting Branch Offices Together. ..................................................... 86 

Using a Gateway Trunk . ................................................................. 87 

Using a SIP Registrations Trunk (SIP Tie Line) . ................................... 88 

5 Dial Plans 91 

Parts of a Dial Plan . ............................................................................ 91 

Key Components . ......................................................................... 91 

Wildcard Patterns . ....................................................................... 94 

Regular Expression Matching . ......................................................... 94 

Building a Dial Plan ............................................................................ 95 

Creating a Dial Plan . ..................................................................... 95 

Configuring the Dial Plan ............................................................... 95 

Sample Dial Plans . ............................................................................. 97 

Typical Dial Plan. .......................................................................... 97 

A Simplified Dial Plan . ................................................................... 97 

Dial Plan with Prefix in Front of the Number . ....................................... 97 

A North American Dial Plan ............................................................ 97 

Sending Star Codes on a Trunk . ...................................................... 98 

Forced Matching . ......................................................................... 98 

Inter-Domain Dialing . .......................................................................... 99 

Domains with Non-Overlapping Extensions . ....................................... 99 

Domains with Overlapping Extensions . .............................................. 99 

6 Extensions 101 

Preparing the User Account. ............................................................... 101 

Working with the Default Extensions . .............................................. 101 

Importing Multiple Extensions (Bulk) . .............................................. 103 

Creating a Few New Extensions .................................................... 103 

Before Registering the Extension . ........................................................ 106 

Administrator-Only Settings . ......................................................... 106 

Registration Settings . .................................................................. 106 

Permission Settings . ................................................................... 109 

7 Service Flags 111 

Setting up a Service Flag Account . ...................................................... 112 

Linking a Service Flag to an Account . ................................................... 114 

Method 1: Night Service . ............................................................ 115 

Method 2: Message-Only and Voicemail Options . ............................. 116 

Setting/Clearing a Service Flag . .......................................................... 117 

Setting a Manual Service Flag from the Phone . ................................. 118 

Setting a Manual Service Flag from the Web Interface . ....................... 118 

Service Flags and Buttons ................................................................. 118 

Using a Button to Activate a Service Flag . ....................................... 118 

8 Auto Attendant 121 

How an Auto Attendant Works . ........................................................... 122 

Welcome Message . .................................................................... 122 

Processing User Input . ................................................................ 123 


viii 


Ringing an Extension . .................................................................. 123 

Blacklisted Callers and Anonymous Calls ........................................ 124 

Multiple Messages—Time-Based Configuration . ................................ 124 

Setting up an Auto Attendant . ............................................................. 124 

Auto Attendant Behavior . ............................................................. 125 

Timeout Handling . ...................................................................... 127 

Night Service ............................................................................ 128 

Dial-by-Name . ........................................................................... 128 

Direct Destinations . .................................................................... 129 

Nesting Auto Attendants . ................................................................... 130 

Welcome Greetings . ......................................................................... 131 

Overview . ................................................................................. 131 

Recording the Greeting . .............................................................. 132 

Filing System and Naming Conventions . .......................................... 133 

9 Hunt Groups 135 

How the Hunt Group Works . .............................................................. 136 

Setting up a Hunt Group ................................................................... 137 

Identity . .................................................................................... 137 

Ring Stages . ............................................................................. 138 

Behavior . .................................................................................. 138 

Night Service ............................................................................ 140 

10 Agent Groups 141 

How the Agent Group Works . ............................................................. 142 

The Message-Music Cycle .......................................................... 143 

Setting up an Agent Group Account . .................................................... 144 

Identity . .................................................................................... 144 

Behavior . .................................................................................. 145 

Connecting Caller to Agent . ......................................................... 149 

Preventing Lengthy Periods in Ringback or Queue . ............................ 150 

Mapping Telephone Keypad Numbers with Extensions . ....................... 150 

Night Service ............................................................................ 152 

SOAP Interface . ......................................................................... 152 

Logging In and Out . .......................................................................... 152 

Method 1: Buttons . ................................................................... 153 

Method 2: Star Codes . .............................................................. 153 

Monitoring Agent Groups . .................................................................. 154 

Monitoring from the Web Interface . ................................................ 154 

Monitoring Using Buttons on Phone ............................................... 156 

Monitoring from the WAC (Web-Based Console) . ............................. 156 

11 Paging 157 

Types of Paging . .............................................................................. 157 

Unicast Paging . ......................................................................... 157 

Multicast Paging . ....................................................................... 158 

Setting up a Paging Account . ............................................................. 158 

Assigning Multicast IP Addresses to IP Phones . ................................ 161 

12 Conferencing 163 


Contents ix 

Scheduled Conferences . ................................................................... 164 

Creating a Scheduled Conference Account ..................................... 164 

Scheduling a New Conference . ..................................................... 166 

Modifying Scheduled Conferences . ................................................ 167 

Moderating Scheduled Conferences . .............................................. 167 

Ad Hoc Conferences . ....................................................................... 168 

Creating an Ad Hoc Conference Account . ....................................... 168 

Calling into an Ad Hoc Conference . ............................................... 169 

Moderating an Ad Hoc Conference . ............................................... 169 

13 IVR Node 171 

How the IVR Node Works . ................................................................. 172 

Setting up an IVR Node Account . ........................................................ 173 

Identity . .................................................................................... 174 

Settings . .................................................................................. 174 

Recording Messages . ....................................................................... 175 

Record the Message Directly . ....................................................... 175 

Upload a WAV File . .................................................................... 176 

Linking an External Application Server to an IVR Node . ............................. 176 

Sample SOAP Message .............................................................. 176 

14 Calling Card Account 179 

Setting up a Calling Card Account . ...................................................... 180 

Identity . .................................................................................... 181 

Behavior . .................................................................................. 181 

15 Pre-Pay Feature 183 

Setting up the Pre-Pay . ..................................................................... 183 

Creating the Rates Table ............................................................. 183 

Assigning a Dollar Amount to an Extension . ...................................... 185 

Methods of Access . .......................................................................... 185 

Extension . ................................................................................. 185 

Personal Virtual Assistant . ............................................................ 186 

Calling Card Account (PIN Access) . ............................................... 186 

16 Email 187 

Configuring the SMTP Server . ............................................................ 187 

System-Wide SMTP Settings . ....................................................... 187 

Domain-Specific SMTP Settings . ................................................... 188 

Adding Email Accounts to the System .................................................. 188 

Receiving Email Notifications from the System . ....................................... 191 

Types of Email Notifications . ......................................................... 191 

Customizing Email Notifications . .................................................... 191 

CDRs to Email . ................................................................................ 192 

CDRs for Trunk Activity ............................................................... 193 

CDRs for Your Extension Only. ...................................................... 193 

CDRs for All Extensions on Your Domain . ........................................ 194 

CDRs for All Extensions on All Domains . ......................................... 194 

Call Recording to Email . .................................................................... 194 


x 


Blacklist Activity to Email . .................................................................. 195 

System-Initiated . ........................................................................ 195 

17 Audio and Greetings 197 

Greetings . ...................................................................................... 197 

Personal Greetings . .................................................................... 197 

Auto Attendant Greetings . ............................................................ 198 

Agent Group Greetings . .............................................................. 200 

Recorded Phone Calls . ..................................................................... 201 

User-Initiated Recordings . ............................................................ 201 

System-Initiated Recordings . ........................................................ 202 

File System . .............................................................................. 202 

Notes . ..................................................................................... 204 

Audio Prompts ................................................................................ 204 

Uploading New Languages . .......................................................... 205 

Customized Ringtones . ..................................................................... 206 

18 Buttons 209 

Creating a Button Profile ................................................................... 211 

Point-and-Click Method . .............................................................. 211 

CSV Method . ............................................................................ 213 

Assigning Button Profiles to Users . ...................................................... 214 

Assigning a Profile to a Single User . ............................................... 214 

Massive Update of Profiles . .......................................................... 215 

Key System Configuration . ................................................................. 215 

Setting up a Key System . ............................................................. 216 

Multiple Identities and Button Profiles . .................................................. 217 

19 Plug and Play 219 

Finding snom ONE . .......................................................................... 219 

Plug and Play Checklist . .................................................................... 220 

Configuring the Administrator Settings . ........................................... 220 

Configuring the Domain Settings . .................................................. 221 

Creating an Authentication Password ............................................. 222 

Setting up the Domain Address Book ............................................. 223 

Configuring the Button Profiles . ..................................................... 223 

Creating the Extension(s) . ........................................................... 223 

Performing Plug and Play . .................................................................. 224 

DHCP, Option 66 . ...................................................................... 224 

SIP SUBSCRIBE (Multicast) . ........................................................ 225 

Mass Deployment . ...................................................................... 226 

Manual Method (HTTP) . ............................................................... 226 

Resetting the Phone . ........................................................................ 229 

Using the Web Interface . ............................................................. 229 

Using the Phone ........................................................................ 229 

Using the Phone GUI . ................................................................. 229 

Registering Numerous Extensions to One Phone . .................................... 230 

Overriding Plug and Play Defaults . ....................................................... 231 

Changing Default Settings from Admin > Settings > PnP . ................... 231 


Contents xi 

Changing Default Settings Using a Customized XML File . ................... 233 

Troubleshooting . .............................................................................. 239 

20 Music on Hold 241 

Setting up Music on Hold .................................................................. 242 

Editing MoH Sources . ................................................................. 243 

Creating WAV Files . ......................................................................... 243 

Configuring Paging/Music on Hold . ...................................................... 245 

Download and Installation ............................................................ 245 

Setting up Paging . ...................................................................... 246 

Configuring Music on Hold . .......................................................... 248 

XML Configuration File . ............................................................... 249 

21 Call Detail Records (CDRs) 251 

Automatically Generated CDRs . .......................................................... 251 

“Extension” CDRs ...................................................................... 252 

“IVR” CDRs .............................................................................. 252 

“Trunk” CDRs . ........................................................................... 253 

CDRs for Third-Party Software Vendors . ............................................... 254 

CDR to CSV . ............................................................................ 255 

CDR to Email . ........................................................................... 256 

CDR to Simple TCP . ................................................................... 256 

CDR to SOAP/XML . ................................................................... 258 


22 Web Interface 261 

Logging in . ..................................................................................... 261 

Extension Settings . ........................................................................... 262 

General Settings . ....................................................................... 262 

Redirection Settings . .................................................................. 265 

Email Settings . .......................................................................... 268 

Instant Message . ........................................................................ 270 

Call Lists . ....................................................................................... 271 

Mailbox . ................................................................................... 271 

Missed Calls . ............................................................................ 271 

Call Log . .................................................................................. 272 

Contacts . ....................................................................................... 273 

Personal Address Book . .............................................................. 273 

Domain Address Book . ................................................................ 275 

Upload Domain Address Book to Phone . ......................................... 276 

Extension Status . ............................................................................. 276 

23 Star Codes 279 

Basic Star Codes . ............................................................................ 280 

Redial a Number (*66) . ................................................................ 280 

Call Return (*69) . ....................................................................... 281 

Intercom (*90) . .......................................................................... 281 

Transferring Calls . ............................................................................ 281 


xii 


Transfer Calls . ........................................................................... 281 

Transfer a Call Directly to Someone’s Voicemail . ............................... 282 

Call Park (*85) and Call Park Retrieve (*86) . .................................... 283 

Call Pickup (*87) . ....................................................................... 283 

Conferencing . ................................................................................. 284 

Three-Way Conferences . ............................................................. 284 

Conference (*53) . ...................................................................... 285 

Anonymous Calls . ............................................................................ 285 

Block Caller-ID (*67) . .................................................................. 285 

Reject Anonymous Calls (*88) . ...................................................... 286 

Forwarding Calls . ............................................................................. 286 

Hot Desking (*70) ...................................................................... 286 

Call Forward All Calls (*71/*72). .................................................... 287 

Call Forward on Busy (*73/*74) . .................................................... 287 

Call Forward on No Answer (*75/*76) . ............................................ 288 

Do Not Disturb (*78) . .................................................................. 288 

Set Night Mode for Domain Accounts (*80) . ..................................... 289 

Call Center Features . ........................................................................ 289 

Agent Log in/Log out (*64/*65) . .................................................... 290 

Call Barge (*81) . ........................................................................ 290 

Call Teach Mode (*82) ................................................................ 290 

Listen In (*83) . ........................................................................... 291 

Miscellaneous . ................................................................................ 291 

Show Account Balance (*61) . ....................................................... 291 

Wakeup Call (*62) . ..................................................................... 291 

Request Call Details (*63) . ........................................................... 291 

Clean up an Extension (*84) ......................................................... 292 

Add to White List (*91) . ............................................................... 292 

Add to Black List (*92) . ............................................................... 292 

Call Record (*93/*94) ................................................................. 292 

24 Voicemail 295 

Setting up the Voicemail Account . ....................................................... 295 

Changing the PIN . ...................................................................... 295 

Record Your Name . .................................................................... 296 

Record a Personal Welcome Message . ........................................... 296 

Accessing the Voicemail System . ........................................................ 296 

Forwarding a Message . ..................................................................... 297 

Leaving Voicemail Messages . ............................................................. 297 

Composing a New Message . .............................................................. 298 

Personal Greetings . .......................................................................... 298 

Recording/Activating Personal Greetings . ........................................ 298 

Hearing Your Personal Greetings . .................................................. 299 

Advanced Features . .......................................................................... 299 

Send a Voicemail Message Without Ringing Extension(s) . ................... 299 

Transfer a Call Directly to Someone’s Voicemail . ............................... 299 

Listen to Voicemail on Cell Phone . ................................................. 300 

Voicemail Notification . ....................................................................... 300 


Contents xiii 

Via Email . ................................................................................. 300 

Via Cell Phone . .......................................................................... 300 

Voicemail Feature Codes . .................................................................. 300 

Send Voicemail to Email Account (*95/*96) . ..................................... 301 

Go to Voicemail (*97) . ................................................................. 301 

Go to Group Mail (*97) . ............................................................... 301 

Clear Message Waiting Indicator (*99) . ........................................... 302 

25 Cell Phones 303 

Cost Savings . ................................................................................. 304 

Callback . .................................................................................. 304 

Calling Card Account . ................................................................. 305 

Configuring the Cell Phone . ............................................................... 306 

Add a Cell Phone to Your Account . ................................................ 306 

Safeguard Against Loss-of-Signal Events . ....................................... 308 

Ring My Cell Phone When Voicemail Arrives . ................................... 308 

Feature Codes ................................................................................ 309 

Retrieve Call From Cell Phone (*51) . .............................................. 309 

Move Current Call to Cell Phone (*52) . ........................................... 310 

Call Cell Phone of the Extension (*00) . ........................................... 310 

Personal Virtual Assistant . .................................................................. 310 

Appendix A. Working with CSV Files . ......................................................... 315 

Structural Requirements . ................................................................... 315 

Structure . ................................................................................. 315 

Parameters . .................................................................................... 316 

Extensions . ............................................................................... 316 

Agent Groups . ........................................................................... 317 

Hunt Groups . ............................................................................ 317 

Conference Account . .................................................................. 318 

Paging . .................................................................................... 318 

Service Flags ............................................................................ 318 

Converting an Excel File to CSV . ......................................................... 319 

Changing the Field Separator and Validating the File . ......................... 323 

Creating a Domain Address Book . ....................................................... 324 

Address Book Parameters . ........................................................... 324 

Converting the Address Book to CSV . ............................................ 324 

Changing the Field Separator and Validating the File . ......................... 326 

Importing the CSV File . ..................................................................... 327 

Importing a CSV File for the Extension Account ................................ 327 

Importing CSV Files for Domain Address Books . ............................... 328 

Troubleshooting . .............................................................................. 329 

Appendix B. SIP Overview . ....................................................................... 331 

What is SIP? . .................................................................................. 331 

Background . ............................................................................. 331 

What is a SIP Session? . .............................................................. 331 

SIP Components . ............................................................................. 331 


xiv 


User Agent Clients . .................................................................... 332 

User Agent Servers. .................................................................... 332 

Back-to-Back User Agent (B2BUA) . ............................................... 333 

SIP Language. ................................................................................. 333 

Similarities with HTML and SMTP . ................................................. 333 

Types of Messages . .................................................................... 334 

Content of Messages . ................................................................. 335 

Describing the SIP Session . ............................................................... 336 

SIP Call Proces . .............................................................................. 337 

Session Establishment ................................................................ 337 

Session Termination . ................................................................... 339 

REGISTER Request . ................................................................... 339 

CANCEL Request . ..................................................................... 341 

snom ONE Settings and SIP Traces . .................................................... 344 

Viewing SIP Traces . .................................................................... 344 

Logfile Syntax . ........................................................................... 344 

Troubleshooting with Logfiles . ....................................................... 345 

Trunk and Extension Settings . ....................................................... 345 

Appendix C. snom ONE and Exchange . ...................................................... 351 

Configuring Exchange for snom ONE ................................................... 351 

Create a New Unified Messaging Dial Plan ...................................... 351 

Associate the Dial Plan with the Unified Messaging Server . ................. 352 

Enable Mailbox Recipients for Unified Messaging . ............................. 353 

Configuring the snom ONE Server . ...................................................... 353 

Create a new trunk to connect to Exchange: . .................................... 353 

Add the Exchange Gateway to Your Current Dial Plan . ....................... 354 

Set the snom ONE External Voicemail System . ................................. 355 

Glossary . ................................................................................................ 357 

Index . ................................................................................................... 369 


Part I 

Getting Started 




GETTING STARTED 

Chapter 

1 

Welcome to snom ONE, your state-of-the-art IP telephone system. Administrating a VoIP system can be a 

daunting task for administrators unfamiliar with VoIP. This guide is designed to help you plan and configure 

snom ONE Voice over IP (VoIP) deployments ranging in complexity from a small office using the Internet 

for SIP trunks to a large, multi-national conglomerate of snom ONE systems. 

snom ONE is the answer to all your IP telephone system needs. 

About snom ONE 

snom ONE is a SIP-based IP telephone system which is implemented in a back-to-back user agent (or 

B2BUA), so all traffic between two endpoints traverses the system (Figure 1-1). This gives the system 

complete control of the call state and allows it to participate in all call requests. It also allows it to perform a 

number of functions that are impossible using the SIP proxy model. A few such functions are forking calls to 

the cell phone, returning them to the system, if needed, and connecting phones remotely to the system. The 

B2BUA also facilitates advanced media features such as call recording, barge, listen in, and whisper. 

Answering 

SIP UA 

Originating 

SIP Endpoint 

Figure 1-1. B2BUA Architecture 

Originating 

SIP UA 

Answering 

SIP Endpoint 

The system is easily portable to most operating systems and requires only a small amount of memory. Due 

to the need for speed, snom technology does not use a traditional database to store its information. Doing 

database lookups can be slow in a very busy system. It is much faster to have a data structure in memory 

with the necessary information to authenticate a phone. 

The snom ONE is a software-based solution and executes several measures to preserve call quality: 

■ A high-scheduling priority is used with applications that are real-time critical. 

■ The snom ONE measures CPU usage and rejects calls if the load becomes too high. 

■ When too many calls are coming in during a short period (call bursts), the system also rejects calls. 

While these measures are important, the performance of the software depends largely on the hardware being 

used. 


2 


Memory/Hardware Requirements 

Memory is usually not a problem today, but influencing factors include the number of voice mails, the 

amount of logging information that is stored on the system, and the number of CDRs. If you want to keep 

2 weeks’ worth of information and you have heavy traffic, you will need enough memory to store that information. 

At minimum, any system needs at least 40 MB of memory. For 20 extensions, 50 MB is recommended, 

and for 150 extensions, 150 MB. 

10 Extensions 20 Extensions 150 Extensions 

Memory* 40 MB 50 MB 150 MB 

Hard Disk** 100 MB 100 MB 150 MB 

* Memory requirements will increase according to voicemails, CDR duration, etc. 

** Hardware requirements are significantly impacted by recordings, as they consume lots of disk space. 

CPU requirements can often be a challenge. The ideal scenario will be to have a dual-core CPU, wherein the 

entire system is bound to one of the cores (processor affinity mask), and the operating system and other nonreal-time 

applications use the other core. In this scenario, the operating system will use one core whenever 

possible. 

(Virtualization is also an option.) 

When determining hardware requirements, you need to factor in the number of parallel calls, the number of 

registrations that will be kept alive on the system, and transcoding issues (i.e., will the CPU need to translate 

the codecs?). If the CPU only has to pass the packets through, the CPU performance is practically limited by 

the I/O-subsystem. 

Hard disk space is also dependent on whether you will be recording calls and the length of time those 

recordings will be kept on the system. At minimum, 100 MB will be needed, but call volume and other 

variables will determine whether 100 MB is really sufficient. 

Although the hard disk requirement for the installation is minimal (less than 20 MB), it is essential that you 

keep an eye on the system once it has been in production for a while. Additional languages, recorded calls, 

and log files can fill the disk quickly. 

USB Device 

The snom ONE software has been loaded onto the 1 GB USB drive that was shipped with this book. This 

section contains information about the USB drive. 

Warnings 

The USB drive is for indoor use or storage with a temperature range between +0°C and +70°C. Not for 

outdoor use! Do not use or store product in rooms with high humidity (for example, in bathrooms, laundry 

rooms, damp basements). Do not immerse product in water and do not spill or pour liquids of any kind 

onto or into any part of it. Do not use product in surroundings at risk for explosions and do not use the 

handset in such surroundings (paint shops, for example). Your device may contain small parts. Keep them 

out of the reach of small children. 

Standards Conformance 

This device is CE and FCC-certified and meets European and U.S. health, safety, and environmental standards. 

Unauthorized opening, changing, or modifying the device will cause the warranty to lapse and may 


Chapter 1: Getting Started 3 

also result in the lapse of the CE and FCC certifications. In case of malfunction, contact authorized service 

personnel, your reseller, or snom technology. 

Disposal 

This device is subject to European Directive 2002/96/EC and may not be disposed of with general household 

garbage. The separate collection and proper disposal of electrical and electronic devices serves to protect 

human health and the environment, as well as to provide a vehicle for using and reusing resources prudently 

and rationally. If you do not know where you may dispose of the device at the end of its lifespan, contact 

your municipality, your local waste management provider, or your seller. Disposal of electrical and electronic 

products in countries outside the European Union should be done in line with local regulations. Please contact 

local authorities for further information. 

Installing the Software 

The snom ONE software has been loaded onto the USB drive that was shipped with your order. Follow the 

instructions below to complete the installation. 

Windows Installations 

1. Put the USB drive in your computer. The software will be listed with your standard disk drives. 

2. Double-click the snomone folder. 

3. Double-click the windows folder. 

4. Double-click setup.exe. 

5. Click Next at the welcome wizard. 

6. Choose an installation location or accept the default, then click Next. 


4 


If you are installing the software on a 64-bit system, the default directory path will be as follows: 

7. Click Next to start the installation. 

The installation will take a minute to install. 

8. Click Close when the installation is complete. 


Logging In 


Once the software has been installed, the snom ONE will run as a service. To access it, bring up a browser 

from the machine on which the software has been installed and enter http://localhost. 

Note: If there is a firewall running on the system, be sure that it allows access to HTTP as well as the other 

ports of the system (see page 31). 

The login screen shown below will be displayed: 

Figure 1-2. Login Levels 

Login Levels 

snom ONE includes three main login levels—administrator, domain administrator, and user— that provide 

access to different web interfaces. The login levels are shown in Figure 1-2, and the web interfaces for each 

login type are shown in Figure 1-3: 


6 

System 

Administrator 

Domain 

Administrator 

User 


Figure 1-3. snom ONE Web Interfaces 

The Automatic login type will default to the highest login level of the person logging in. Administrators 

and domain administrators who want to log in as a user must enter their extension number and choose User 

Login from the dropdown. Instructions for each login type are detailed below. 

Login Types 

■ For users: 

— Account: Extension number (e.g., 509) 

— Password: Extension password 

— Login Type: Automatic 

■ For system administrators: If you are logging in as an administrator, use the following account 

information to log in: 

— Account: admin 

— Password: The password field is blank by default. 

— Login Type: Automatic. 

Note: When either a system or a domain administrator logs in as a user, User Login must be 

selected from the dropdown. 

■ For domain administrators: If you are logging in as a domain administrator, use the following account 

information to log in: 



— Login Type: Automatic 

■ To access the WAC: 



— Login Type: Console Login 


Activating Your License 


Your snom ONE license will need to be activated before you can register extensions to the system. The activation 

code is located on the back cover of this book near the Product key barcode. To activate your license, 

complete the following steps: 

1. Click Admin. 

2. Click the Settings tab, then click License as shown below: 

3. Enter your activation code into the Code box (the activation code is located on the back cover of this 

book near the Product key barcode). 

Upgrading the Software 

Software upgrades can be downloaded from wiki.snomone.com. To avoid interrupting calls, perform system 

upgrades during off hours. If you need to upgrade the system when the system is active, you can send a page 

announcing the upgrade, or in cases where only a few people are involved, barge into the call and inform the 

participants of the upgrade. The Maximum Number of Calls setting can also be used see restrict call activity 

(see page 28). Configure this setting to 0 so that no new calls can be established on the system. To 

check the status of the system, navigate to Admin > Status > Graphs. A busy system will show active calls 

on the system graphs. 

Another way to determine how many active calls are on the system is to check the call list (from the domain 

settings, click Status > Calls). 


8 


Manual Software Upgrades 

Manual upgrades can be done by replacing the pbxctrl.exe file. Before the upgrade, you might want to 

make a backup of the working directory in the event you want to revert to the former software version. 

1. Determine the version number of the current software by going to Admin > Status. 

2. Navigate to the working directory and rename the pbxctrl.exe file so that it corresponds with the 

old version number (e.g., pbxctrl-2.0.3.1713). Keep the file in the directory so that you can 

later determine which software version is associated with which file. This step can be done while the 

system is running. 

3. Go to wiki.snom.com and download the new executable (click the Existing Installations Only 

link). 

4. Put the file into the snomONE working directory. 

5. Rename the new file pbxctrl.exe. 

6. Restart the system. 

Restarting the System 

Various system configurations require a system restart. You can do this by either restarting the computer or 

restarting the system. When restarting the system, use one of the following methods. 

Restarting From Windows (Service Control Manager) 

1. Right-click My Computer. 


2. Click Manage. 

3. Click Service Applications, then click Services. 

4. From the right-hand pane, click snomONE PBX. 

5. Click Restart. 

Restarting From a Command Line 

The system can be stopped and restarted from the command line: 

■ net stop snomone 


VoIPon www.voipon.co.uk sales@voipon.co.uk Tel: +44 (0)1245 808195 Fax: +44 (0)1245 808299 

.

10 


■ net start snomone 

When the System Fails to Restart 

If you get an error message when restarting the service, you may need to modify the log_filename parameter 

in the pbx.xml file. The value of the log_filename parameter requires a dollar sign ($) and should be 

modified if it does not contain a dollar sign; otherwise, your system’s performance will become unstable. The 

following sample shows a problematic value in the log_filename field. 

To change the value, open the pbx.xml file using an XML editor. Click in the text and replace it with 

log-$.txt. 

Technical Support 

Technical support for snom ONE is available through the wiki, the online help system, and the forum. If 

you cannot find an answer to your question through these sources, you can also open a ticket with snom 

ONE support. Each support option is detailed below. 

Wiki 

The view the knowledge data for snom ONE, go to wiki.snomone.com. 

Online Help 

To access online help, click the Help link at the top of each page. 


Forums 

To join the forum, go to forum.snom.com. 

Submit a Ticket 


To submit a ticket or view existing tickets, go to support.snom.com, and log in or register a new account: 

Documentation 

The following typographic and usage conventions occur in this book: 

Typeface Description Example 

Bold Names of tabs, settings, Navigate to Admin > Settings > Log- 

and buttons 

ging. 

monospace File and directory names, ■ The default recording name is 

examples of program 

$r/$d/$t-$i-$u-$n.wav 

code, command strings, ■ Place the file into the 

and user input. 

recordings directory. 

■ A typical dial plan would include 

the string ([0-9]*)@.* 

as pattern and sip:\1@\ 

r;user=phone as the 

replacement. 

monospace Names of parameters 

■ Modify the cdr_field_ 

italic 

separator parameter. 

Feedback and comments can be sent to feedback@snom.com. 


12 



Part II 

Administering 

the System 




SYSTEM SETTINGS 

Chapter 

2 

The snom ONE telephone system encompasses three different accesss levels—administrator, domain administrator, 

and user—each providing a different web interface (depicted earlier on page 6). The system administrator 

has control of the entire system, while the domain administrator has control of only the domain. 

This setup enables the administrator to assign responsibilities related to user accounts and other non-system- 

related settings to another individual without compromising system configurations. This chapter discusses 

only administrator-level settings. Configurations that can be made by the domain administrator are discussed 

in Chapter 3. 

Admin 

Settings 

Admin 

Domain 

Settings 

Domain Admin 

Domain 

Settings 

User 

User 

Settings 

User 

Settings 

User 

Settings 

Figure 2-1. snom ONE Web Interfaces (Admin, Domain Admin, and User) 

To access the administrator settings: 

1. Log in as admin. The following screen will be displayed. 


.

16 


2. Click Admin. The new screen includes the four tabs shown below. These tabs can be seen only by 

administrators. These setting are briefly outlined in Table 2-1 and are discussed in detail throughout 

this chapter. 

Table 2-1. Menu Tree of Administrator Tabs 

Tab/Link Name Description 

Settings 

General This page allows you to configure basic global settings (IVR, tone, and web languages), 

including performance, password, and SIP settings. 

License This is where you activate your license code. 

Ports The ports that are used for the system are specified here. 

Logging The settings on this page are designed to help you debug issues. 

Configuration From this page, you can restore the system’s configuration, including recordings and 

customized files in the tftp server directory. You can also request configuration 

information from a service provider, reload configuration files if you want to change 

them without restarting the system, or schedule a system reboot. 

Certificate From this page, you can install a certificate and private key to ensure secure communication. 

The certificate will be checked by clients that need to trust you. The private 

key will be used for encrypting messages. 

MoH The settings on this page allow you to specify which music on hold sources will be 

available on the system. 

PnP From this page, you can provide parameters to change the generation of plug and 

play files. 

Access You can control which IP addresses may access the snom ONE service here. 

User Page 

Control 

This page lets you control which settings will be displayed on the user’s web interface. 

Domains From the Domains tab, the administrator can view and edit the domain. 

Email 

General This page allows you to configure the mail client settings that will be used by the 

system when sending out emails. 

Messages The settings on this page allow you to specify which events warrant an email notification 

and who should receive that notification. Examples include call rejection due 

to CPU load and call disconnection because of one-way audio. 

Texts This page allows you to change the text of system-generated email. 

Status 


Tab/Link Name Description 

Chapter 2: System Settings 17 

General This page shows you the status of your system (e.g., the release of software you are 

running, the type of software, the expiration date, etc.). 

Logfile This page displays SIP messages, which can be used for troubleshooting. 

Call Log A list of calls that have been made through the system is displayed. 

Calls A list of active calls is displayed. 

Directory Structure 

Directory Tree 

While most of the system settings can be configured from the web interface, the administrator will need to 

access the file system at times. This section provides a brief overview of the files that are part of the system 

and explains why you may need to access them. 

All files related to the system are located in C:\Program Files\snom\snomONE, which is the system’s default 

working directory. The main program that runs the software is pbxctrl.exe, and the global configuration 

file is pbx.xml (Figure 2-2). The folders inside the snomONE directory contain files that hold information 

about accounts, trunks, audio prompts, features, and many other aspects of the system. The majority 

of folders are composed during installation. However, a few folders (e.g., cdre, cdri, cdrt) are composed 

only after the phone communicates with the system (i.e., once a call is made or a star code is used). 

Figure 2-2. snom ONE Directory Structure 

Global Configuration File 

For settings that cannot be changed through the web interface, the global configuration file can be used (Figure 

2-2). This file is a standard XML file, encoded in UTF-8. Unlike the majority of files that can be modified 

through the web interface, the global configuration file contains many hidden settings which cannot be 

changed this way. Changes to this file must be made through either the XML file or by handcrafting a URL 

that contains the change. With either method, the three important components are the address of the snom 


18 


ONE system, the name of the setting, and the value. In the URL example shown below, the address is pbx, 

the setting is setting, and the value is being changed to 123. 

Important: Always make a backup before modifying the file! 

http://pbx/reg_status.htm?save=save&setting=123 

Once the URL is set, press Enter. This is equivalent to pressing the Save button when in admin mode. 

A typical use case for modifying the global configuration file would be to change the format of CDRs (applicable 

only for CDRs that are exported via simple TCP). To change the CDR format of a system running 

on a localhost, you would use something similar to 

http://localhost/reg_status.htm?save=save&cdr_format=$w$5d$25m$2o$20F$20T$20 

R$20r$15c$20f$15v 

where cdr_format is the parameter. 

Caution: The above link will change your CDR format if you execute it while the system is running on the 

localhost! 

Note: If you want a specific delimiter between the fields, use a comma (e.g., $w,$5d,$25m,$2o,$20F 

,$20T,$20R,$20r,$15c,$20f,$15v). Be careful when choosing a delimiter. If the names or URIs 

contain those characters, there will be a conflict. 

Folder Overview 

Table 2-2 lists the folders that are located in snom/snomONE: 

Table 2-2. Folder Details 

Folder Name Contents 

accesslist IP addresses that have been blocked from accessing the system 

acds Account information about each agent group 

adrbook Contacts that have been added to the domain address book 

attendants Account information about each auto attendant 

audio_en Audio prompts for auto attendant, IVR, etc., and ring tones 

audio_moh Music on hold and universal tones like ringback 

button_lists Button profiles that are on the domain 

buttons Individual buttons that are part of the button profiles 

callingcards Account information about each calling card 

cdr CSV CDRs, if any exist 

cdre* CDRs that contain information about the extension 

cdri* CDRs that indicate the role played by the IVR 

cdrt* CDRs that contain information about the trunk 

colines Central office lines on the trunk 

conferences Account information about conference accounts 

dial_plan Dial plans that are on the domain 

dial_plan_entry Individual pattern and replacement values for each dial plan 

domain_alias Information about the domain when it is associated with an alias 

domains Information about the current state of the domain 

email_templates Custom email templates 


Folder Name Contents 


extensions Information about each extension (name, number, MAC, etc.) 

generated** Files that are generated from templates that exist within the system binary 

hoots Multicast paging information 

html Customized HTML files 

hunts Account information about hunt group accounts 

ivrnodes Information about each IVR node account 

messages Voicemail messages 

mohs Available music on hold sources 

pnp_parms Configuration files that are involved in plug and play 

recordings Recorded calls and greetings 

registrations Static registrations 

schedules User-created conferences 

srvflags Account information about service flag accounts 

tftp Firmware and/or provisioning files for the phones 

trunks Information about each trunk on the system 

user_alias Information about each account associated with an alias 

users Information about each account (extension, auto attendant, etc.) 

wipers Conference recordings 

* The cdre, cdri, and cdrt folders are generated after the phone contacts the system. 

** The generated folder is created when the phone contacts the system for its configuration files during plug and play. 

Folder Details 

Each folder in the snomONE directory contains a list of XML files similar to the following: 

Each XML document reflects various XML constructs and concepts: 


20 


Each folder in the snomONE directory is detailed below. 

■ accesslist—The accesslist folder contains IP addresses that have been blocked from accessing 

the snom ONE service. IP addresses can be blocked/unblocked by the administrator in 

Admin > Settings > Access List. 

■ acds—The acds folder contains an XML file for each agent group. Each file includes the possible 

parameters for the account and any values assigned to those parameters. 

■ adrbook—The adrbook folder contains contacts that have been added to the domain address 

book. Each contact has a corresponding XML file. 

■ attendants—The attendants folder contains an XML file for each auto attendant. Each file 

includes the possible parameters for the account and any values assigned to those parameters. 

■ audio_*—The audio_* directory contains audio prompts that are used by the auto attendant 

and IVR node. These prompts affect the language settings that are available from the web interface 

(see page 23). The files within the directory contain many types of prompts. Following is a summation 

of the main prefixes. 

Prefix Description 

aa- Auto attendant 

bi- Built-in numbers 

co- Conference 

code- Feature-code related 

ex- Extension-related 

mb- Mailbox 

pb- ACD-related 

wk- Time-related 

Note 1: The directory suffix corresponds with the language of the prompts themselves (i.e., English 

prompts are contained in audio_en). See page 204 for instructions on installing audio prompts 

for additional languages. 

Note 2: The ringback.wav and busy.wav files are also included in this directory. These files 

influence the ringback and busy tones that a caller hears during voicemail prompts and IVR interaction 

(see page 23 for more on tone language). 

■ audio_moh—The audio_moh folder contains DTMF tones and music on hold sources. 

■ button_lists—The button_lists folder contains the button profiles that are available on 

the domain. 

■ buttons—The buttons folder contains the types of buttons (e.g., private, speed dial, DND) 

that have been set in the button profiles. 



■ callingcards—The callingcard folder contains an XML file for each calling card account. 

Each file includes the possible parameters for the account and any values assigned to those parameters. 

■ cdr—The cdr folder contains the CSV CDRs, if there are any. 

■ cdre—The cdre folder contains CDRs that hold extension-related information. A CDR is generated 

for each call. 

■ cdri—The cdri folder contains CDRs that hold IVR-related information. A CDR is generated 


■ cdrt—The cdrt folder contains CDRs that hold trunk-related information. A CDR is generated 



22 


■ colines—The colines folder contains information about co-lines that are created on the system. 

■ conferences—The conferences folder contains an XML file for each conference account. 

Each file includes the possible parameters for the account and any values assigned to those parameters. 

■ dial_plan—The dial_plan folder contains an XML file for each dial plan on the system. 

■ dial_plan_entry—The dial_plan_entry folder contains the pattern and replacement 

values for each entry in the dial plan. 

■ domain_alias—The domain_alias folder contains the alias name of each domain. Each alias 

has an XML file. 

■ domains—The domains folder contains details about the domain, such as star codes, password, 

and PIN requirements. 

■ email_templates—The email_templates folder contains customized email templates. 

■ extensions—The extensions folder contains an XML file for each extension. Each file includes 

the possible parameters for the extension and any values assigned to those parameters. 

■ hoots—The hoots folder contains an XML file for each paging account. Each file includes the 

possible parameters for the paging account and any values assigned to those parameters. 

■ html—The html folder is used for customizing configuration files, such as ringtones.xml and 

pnp.xml. You can download these files from 

wiki.snomone.com and then place them into the html folder. Once the snom ONE is restarted, 

the new configurations will take effect. 

■ hunts—The hunts folder contains an XML file for each hunt group. Each file includes the possible 

parameters for the account and any values assigned to those parameters. 

■ ivrnodes—The ivrnodes folder contains the details of all IVR node accounts. Each file includes 

the possible parameters for the account and any values assigned to those parameters. 

■ messages—The messages folder contains voicemail messages. 

■ mohs—The mohs folder contains a WAV file for the sources of music on hold that are available on 

the system. 



■ pnp_parms—The pnp_parms folder contains all configuration files that are involved in plug and 

play. 

■ recordings—The recordings folder contains a WAV file for all greetings and recordings, 

whether created for personal or system use (e.g., auto attendants, agent groups, and hunt groups). 

See Chapter 17 for more information on recordings. 

■ registrations—The registrations folder is needed for the index file. 

■ schedules—The schedules folder contains conferences that were created by users. 

■ srvflags—The srvflags folder contains an XML file for each service flag account. Each file 

includes the possible parameters for the account and any values assigned to those parameters. 

■ tftp—The tftp folder can be used for customizing plug and play. For example, placing the 

snom_320_custom.xml file into the tftp folder will provide new settings for the phone. 

■ trunks—The trunks folder contains an XML files for each trunk. Each file includes the possible 

parameters for the trunk and any values assigned to those parameters. 

■ user_alias—The user_alias folder contains an XML file for every alias name that is associated 

with an account (a DID number is considered an alias). 

■ users—The users folder contains an XML file for every account on the system (DIDs are not 

considered a separate account). 

■ wipers—The wipers folder contains conference recordings. 

Configuring System Settings 

This section describes the general administrator settings that are located in Admin > Settings on the web 

interface. (You must be logged in as administrator.) 

General 

■ System Name: This field allows the administrator to set a name for the system. The name is used 

in several places to identify the system. For example, when the system sends out emails with system 

performance information, the system name will be placed into the subject line of the e-mail. 

The next four settings pertain to language and time zone. The settings made here by the administrator can 

be overridden by the domain administrator. 

■ Default IVR Language: This setting represents the language that is used for voicemail and IVR 

prompts. Only U.S.-English prompts are included in the installer, but additional languages can be 

installed. See “Audio Prompts” in Chapter 17 for more information on audio prompts and languages 

that can be downloaded. 

■ Default Tone Language: This setting allows you to control the ringback and busy tones that a 

caller hears during voicemail prompts and IVR interaction. The files that influence this setting are 

ringback.wav and busy.wav. These files are located in the audio_* directory. The * represents 

a country code (see Table 17-5 for a list of country codes). 


24 


■ Default Web Language: This setting controls the language used on the web interface. The file that 

influences this setting is lang.*.xml, where * represents a country (see Table 17-5 for a list of 

country codes). 

■ Time Zone: The Time Zone setting is used to set the local time zone. It influences the time that 

appears on a user’s mailbox messages, the telephone display, and voicemail. The system is capable 

of dealing with several time zones at once, making it possible for every user to select his or her own 

time zone. 

Administrator Login 

The system administrator’s login must be protected at all times, so it is imperative that a high-security password 

be used (see the Password Strength setting on page 25). The password is stored in a hashed format 

and cannot be read from the global configuration file. 

■ Username: This field sets the administrator’s user name. By default, the user name is admin, but it 

can be changed to anything (we recommend changing it). 

■ Password: This field sets the password for the user name from the previous setting. By default, 

there is no password, but be sure to set one. The administrator’s password can be reset in the event 

it gets lost (see instructions below). 

Resetting the Admin Password if Lost 

To reset a lost password, you will need to remove the existing password from the global configuration file 

(from the pw_pass parameter) then enter a new one through the web interface. This can be done in Notepad. 

Instructions are provided below: 

1. From a text editor, open the root file (pbx.xml). The file path is shown below: 


2. Search for the element tag. 


3. Delete the password. (If no text appears between the element tags, then the password field is empty.) 

4. Save the file. 


6. Set a new password through the web interface (Admin > Settings > General). 

Appearance 

■ Web Session Timeout (s): This field determines how long a web session will stay active before it 

times out. The duration is set in seconds, and the default value is 3600 (1 hour). Increase or decrease 

this setting depending on whether you want the system to log you off after a certain period 

of time. Once timed out, the main login screen will be displayed. 

■ Password Strength: This field is used to specify the types of passwords that are acceptable. The 

default is “Accept All Passwords,” but it is advisable to require medium or high-security passwords. 

Passwords should not be dictionary words. A combination of letters, digits, and symbols is advised. 

— Accept all Passwords: All passwords will be accepted. 


26 


— Medium Security: The score must be 120 or higher (see Table 2-3). 

— High Security: The score must be 200 or higher (see Table 2-3). 

Table 2-3. Password Scoring Method 

Character Type Points 

Digits 10 

Upper/lowercase letters 26 

Symbol 28 

■ Default CDR listing size: A Call Detail Record (CDR) is a record produced by the system that 

contains the details of calls that have terminated on the system. CDRs include the date and time 

the call started, the number that made the call, the number that received the call, and the call duration. 

Use this field to limit the number of CDRs that will be displayed in the web interface (CDRs 

can be memory consuming). The default is 30 CDRs. For more information on CDRs, see Chapter 

21. 

■ Keep CDR Duration: This setting defines the length of time CDRs are kept on the file system. 

CDRs can consume a large amount of disk space, so be careful when setting this field. If too many 

CDRs are on a busy system, the system will not start. The default is 7 days. The duration can be 

expressed in various time units: 

— Seconds: Enter an s after the number (e.g., 60s for 60 seconds) 

— Minutes: Enter an m after the number (e.g., 60m for 60 minutes) 

— Hours: Enter an h after the number (e.g., 24h for 24 hours) 

— Days: Enter a d after the number (e.g., 7d for 7 days) 

■ SOAP Trusted IP: SOAP (Simple Object Access Protocol) is an XML message-based protocol 

specification used for allowing computers to talk to each other over a network such as the Internet, 

typically as part of a Web service. SOAP can be used to pass almost any type of data between two 

applications. Because it is based on XML, SOAP is language- and platform-independent. The two 

applications could be written in different languages and could run on different operating systems. 

SOAP is often used to make Remote Procedure Calls (RPCs). When used to encode RPCs, SOAP 

is used as the request/response protocol. See Chapter 13 for a sample SOAP message. 

The SOAP Trusted IP field controls a SOAP request coming from random addresses and is available 

only if the license key contains a SOAP flag. Use this field to specify the IP address or the host 

names that are permitted to make SOAP request to the system (i.e., management system). If you 

want to allow multiple systems to send SOAP requests, use a space to separate the entries. 

■ CDR URL: The CDR URL field controls where the CDRs are written. The system can write 

CDRs to a SOAP destination and to a CSV file (the default separator for the values is a comma; 

however, it can be changed by modifying the cdr_field_separator parameter). To populate the 

CDR URL field, use the syntax shown in Table 2-4. CDRs can be viewed only after they have been 

collected and organized through an external call accounting application. 

Table 2-4. Syntax Used in CDR URLs 

Format CDR URL Syntax Description of Syntax 

ASCII cdr:192.168.1.2:10000 ■ 192.168.1.2 is the IP address 

■ 10000 is the TCP port of the 

CDR server 



Format CDR URL Syntax Description of Syntax 

CSV file:disk Disk can be any name, as it is not 

used by the program anyway. 

Email mailto:name@company.com 

■ Record Location: This field determines where system-initiated recordings will be stored on the 

system. By default, $r/$o/$a/$d-$t-$i-$n.wav is the string used in this field. For more information 

about recording calls and how to create a string for the Record Location field, see page 

201. 

■ Compress Recording: The system will compress recordings when this setting is enabled. Otherwise, 

recordings will be saved as 16 bits/sample. 

■ Hide aliased rows under domains page: This setting allows you to control whether the alias account 

of your domain will be displayed on the Domains page. When this setting is set to No, the 

list of domains will include the alias in addition to the primary domain, which may be confusing 

since it appears that more domains are on the system than there actually are. When this setting is 

set to Yes , only the primary domain(s) will be displayed. 


28 


Before (set to No): 

After (set to Yes): 

Performance 

From this section, you can set performance-related settings. 

■ Maximum Number of Calls: This setting defines the number of simultaneously calls allowed by 

the system. Because every call requires a certain portion of the available CPU, allowing too many 

calls will affect the quality of all ongoing calls. By limiting the number of calls on the CPU, you 

can reject calls that would otherwise potentially degrade overall performance. On modern PCs, 100 

or more calls can run on one computer; however, on an embedded system, you will probably have 

much less CPU power and thereby increase the probability of running out of CPU power if you 

allow that many calls. If this field is left blank, the key in the system will limit the calls, or the CPU 

limitation, when reached, will limit the calls. 

■ Processor Affinity Mask: This field allows the administrator to assign a particular CPU to the 

pbxctrl system process. When using a multiple-core CPU, the operating system has to assign processes 

to processors. By default, the operating system tries to balance out the load so that the overall 

system performance is as fast as possible. However, the disadvantage with this approach is that the 

whole process gets stuck for some time while moving the processes from one CPU to another. If 

during that time the CPU should play out media, it will come across as stuttering and be perceived 

as jitter coming from the system. In order to avoid this problem, bind the pbxctrl to one fixed 

CPU. Depending on the operating system, you can do this manually or you can ask the system 

processes to do this during startup. Changes to this field require a system restart. 



■ Maximum Duration of Call Recording: This setting allows you to establish an upper limit on call 

recordings. This is important since large recordings can cause problems with system performance. 

(Recordings left in voicemail boxes consume disk space and can have limitations placed on file size.) 

■ Max. size of a configuration backup file: This setting allows you to establish the maximum size, 

in bytes, of the backup file. The default is 1 MB. If your system has a lot of data to be backed up, 

increase this value. The backup feature through the web interface is used mostly in the appliances 

(in embedded systems where the file access is primitive). On other systems, it is advisable to use the 

OS file manager to do the backup. 

■ Max. number of concurrent registrations per extension: This setting controls how many user 

agents (i.e., phones) can be registered against an extension. This feature is useful if you want to 

restrict the number of phones that can be registered to an extension. If this field is left blank, the 

extension will have no limit. 

■ Minimum Registration Time (s): In a SIP environment, the registrar determines how long a user 

agent may be registered. Short registration times have a negative impact on the performance; however, 

it is critical that user agents stabilize quickly once they’ve lost their connection to the system. 

This setting defines the lower limit for the registration time. The default is 30 seconds. 

■ Maximum Registration Time (s): This setting is used to define the upper limit for the registration 

time. The default value is 360 seconds. See also Appendix B, “SIP Overview.” 

■ UDP NAT Refresh (s): If the registering user agent is behind NAT, the system uses this setting 

to control the registration. The system registers agents that use the UDP transport layer only for a 

short time so that the user agents will re-register quickly and keep the NAT bindings alive. Typically, 

the settings for UDP should be from 15 to 45 seconds since most NAT routers close ports 

after 60 seconds of inactivity. The default for this setting is 30 seconds. 

■ TCP/TLS NAT Refresh (s): This is similar to the UDP NAT refresh setting. Since TCP/TLS 

connections do not need to refresh the bindings as often, a value of a few minutes is okay in most 

situations. The default is set to 180 seconds (3 minutes). 

■ Maximum call duration: This setting establishes an upper limit for the call duration. By default, 

the setting is 2 hours (7200 seconds), but you can increase this setting if you lean toward longer 

phone calls. This setting is crucial for keeping your call list clean; for example, if one mailbox talks 

to another mailbox or if a call does not drop properly, the system can automatically clean this up. 

■ Maximum ring duration (s): This setting determines the length of time the system will wait before 

it disconnects a call. 

SIP Settings 

This section defines SIP-related settings. 


30 


■ Use Short SIP Headers: Some SIP headers have a short form (e.g., the From header gets shortened 

to F, the To header becomes T, and Via becomes V). Short headers have the advantage of 

saving space in the messages, reducing overall probability of running into problems with maximum 

message size in UDP. Although it is quite simple to support this, some devices do not support 

short headers. For this reason, the system offers both short and long. In order to maximize interoperability, 

keep the default value (long). If you are running into UDP packet fragmentation problems 

(message size above 1492 bytes), switch to the short header form. 

■ Listen to sip.mcast.net: SIP also has its own multicast group according to RFC 3261. A SIP 

device usually knows where to send requests, but during bootup and configuration, a user agent 

might want to locate the system with a multicast request. If this setting is turned on and you are 

using user agents with the multicast detection feature, you can just plug the devices into the network 

and they will get their configuration information automatically from the system if the system 

PnP settings have been configured (see Chapter 19). Changing the Listen to sip.mcast.net setting 

requires a system restart. 

■ Allow domain admin to change trunks: In hosted environments, the service provider might want 

to set up the trunks and hide this feature from domain administrators. If this feature is set to No, 

domain administrators will not be able to edit or see the trunks. 

■ Allow domain admin to change dial plans: If this feature is set to No, domain administrators 

cannot change the dial plans or even view them from the web interface. 

■ Allow domain admin to create or change accounts: If this feature is set to No, domain administrators 

cannot create, change, or view accounts. 

■ Allow domain admin to create or change ANI: If this feature is set to No, domain administrators 

cannot create or change the ANI (automatic number identification). The ANI is used when the 

system sends an outbound call on a trunk (see domain settings for more details). 

■ Loopback detection: This setting applies only to multi-domain environments (for single-domain 

environments, leave this setting on). When the system starts a call, that same call may come back 

to the system and create a loop. This may happen if the call is for another caller on the system in 

a multi-tenant environment. This is a dangerous situation because it might repeatedly initiate the 

same call, ending up in many calls that take a lot of resources. Therefore, the system must detect 

such a loop. In environments where an external SIP proxy routes the call from one snom ONE domain 

to another, a simple loopback detection that is based on the call-ID is too pessimistic. Therefore, 

in such environments, you might want to allow these calls and turn the loopback detection off 

(see the Try Loopback setting on page 96). 

■ Inband DTMF detection: When a user presses a key on the telephone, the system must be able 

to understand the action. In telephony systems, this mechanism is typically called DTMF. In VoIP, 

DTMF is usually sent “out of band” (RFC 2833) because it allows the system to easily detect the 

tones in a fail-safe manner. However, not all devices support this method, and when this is the case, 

the system must decode and analyze the media stream and perform this detection. Because this is 

erroneous and costs additional CPU performance, it is strongly recommended that you not use this 

feature and instead replace devices which do not support the out-of-band method with devices that 

do. In-band and out-of-band can be compared to a T1 line that uses in-band signalling and a PRI 

line that uses out-of-band signalling. 

■ Remote SIP management: This setting allows the provider to send commands to the system (for 

example, for rereading the configuration). It is useful in environments where the service provider 

controls the system from a centralized location. By default, this setting is off, but if you are using 

such an environment, this setting needs to be activated. 


Ports 


The Ports page allows you to control which networking resources the system will utilize when communicating 

with the outside IP world. When specifying ports, list the ports that you may bind to, either 

specifying a port number or explicitly specifying the IP address and the port, separated by a colon (e.g., 

192.168.1.2:8080). If you are binding to IPv6 addresses, you must put a square bracket around the 

IP address (e.g., [2001:db8::4]:5060). If you are only specifying the port number, the system will 

bind to all IPv4 and IPv6 addresses on the system. If you want to bind only to IPv4 sockets, use the form 

0.0.0.0:5060. If you want to bind only to IPv6 sockets, you can use [::]:5060. In general, you may 

bind to more than one socket; just separate the addresses by spaces. If you do not want to use the service, 

leave the field empty. If you change a port binding, you will need to restart the snom ONE service. We support 

the default SIP port. 

* Changes to HTTP and SIP settings require a system restart. 

HTTP Ports 

The HTTP and HTTPS ports are used to communicate between the built-in web server and the web browser 

(snom ONE does not rely on IIS or Apache for its web interface). The HTTP port is used for insecure, 

but lightweight, communication. The HTTPS port is used for secure, but a bit more expensive, communication. 

If you cannot reach the system on any port, change the ip_http_port and ip_https_port parameters 

in the global configuration file (the defaults are shown below). If you are running another service on your 

host or if you want to gain some additional security, you may change these ports to any other available port. 

The system will fail to start if it cannot bind to the port. 

■ HTTP Port: The default HTTP port is 80. 

■ HTTPS port: The default HTTPS port is 443. 

SIP Ports 

In this section, you can provide specific port information for the SIP protocol. SIP can run on UDP, TCP, 

or TLS. TCP and UDP send the SIP packets unencrypted and therefore are considered insecure. TLS is 

used for secure SIP communication since it encrypts the SIP signalling packets much like HTTPS encrypts 

HTTP traffic. The default SIP port per RFC 3261 is 5060 for SIP and 5061 for SIPS. The snom ONE software 

is listening for and transmitting SIP requests and responses on these ports. 


32 


* Changes to HTTP and SIP settings require a system restart. 

■ SIP UDP Ports: If you are using SIP over UDP, you need to set this field. The default port for 

UDP is 5060. Multiple ports are permitted (e.g., 5060 5064). 

■ SIP TCP Ports: If you are using SIP over TCP, you need to set this field. The default port for TCP 

is 5060. 

■ SIP TLS Ports: If you are using SIP over TLS (Transport Layer Security - Security over TCP), you 

need to set this field. The default port for TLS is 5061. 

■ Maximum number of SIP connections per second: This setting specifies the number of SIP 

conversations the system will respond to in 1 second. This setting is useful for deterring against SIP 

attacks. 

■ Maximum number of SIP connections: This setting limits the total number of SIP connections 

the system will support. This setting must be configured in busy environments where resource limitation 

is an issue. 

■ SIP IP Replacement List: This setting applies to a system that is used in a DMZ zone with NAT 

(e.g., to connect remote phones to a system that is not on a public IP address). In this case, when 

the system builds the remote SIP packets, it will use the public IP address of the router. The setting 

should include a list of local IP addresses and their replacements (e.g., 192.168.1.0/255.255 

.255.0/192.168.1.1 0.0.0.0/0.0.0.0/75.150.87.9). Whenever the system finds a 

local address in the list, it replaces the local address with the remote address, so the SIP messages 

from the system will look as if they were sent from the replaced IP address. The format of the list is 

LocalAddress/RemoteAddress [LAdr/RAdr]. Both the LAdr and the RAdr must be an IPv4 or IPv6 

address (e.g., 192.168.1.2/203.4.5.12). DNS addresses are not resolved here. 

Internet 

Firewall/ 

router 

75.150.87.9 

DMZ 75.150.87.9:5060, 1000-10999 

192.168.1.1 

192.168.0.7 


588 

589


■ IP Routing List: The IP Routing List setting is used to override the operating system IP routing 

table and is linked to the routing table (this setting will be consulted by the system before consulting/using 

the operating system). Whenever the system wants to know the IP address that is being 

used when sending a SIP packet, it steps through the list and looks for a match (using the netmask 

Mask) to a destination address (DAdr). If a match is found, the system uses the provided IP address 

(LAdr). The mask must be in the form of an IP address, e.g., 255.255.255.0. 

RTP Ports 

The Real Time Protocol (RTP) ports are used for sending and receiving media. Be sure to specify a reasonable 

port range so that you have enough ports for all open calls. A port range of 100 ports is not unusual. 

Most user agents send RTP media data from the same port on which they expect to receive data. This is 

useful when a user agent sends media from behind NAT. The system can use this mechanism to establish a 

two-way media path, even if the user agent is not able to determine its public IP address for media and is 

behind NAT. 

■ Port Range Start: This setting represents the starting RTP port that the system will use for media 

sessions. If the system is behind a firewall, these ports should be open. 

■ Port Range End: This setting represents the end RTP port that the system will use for the media 

sessions. RTP uses UDP for transport, whereas SIP can use UDP, TCP, and/or TLS. 

■ Follow RTP: Some user agents use different ports for sending and receiving. Although they will 

not be able to operate behind NAT, they are within the scope of the IETF standards. With this 

setting, these devices can be made compatible. By default, this flag is set to On. If you have trouble 

with devices that use different ports for sending and receiving, try turning this flag off. Some 

troublesome devices also have a flag that can be used to turn the usage of different ports off. 

This behavior can be controlled on a trunk level, as well. If only a specific trunk has this problem, 

use this setting only on the trunk level. 

■ Codec Preference: The Codec Preference setting allows you to select the codecs that will be supported 

on the system. The codecs that are allowed on the system are shown at the left. If you do 


34 


not want to use a particular codec, click the codec, then click Remove. This will move the codec 

to the right-side selection box, removing it from use. The system comes with recommended highquality 

codecs like G.711 µ-law (0), G.711 A-law (8), G.722 (9), G.726 (2), or GSM 6.10 Full- 

Rate (3). Codecs can be changed without restarting the service. G.729 is a royalty-based codec and 

requires a fee, and it is not enabled by default. 

■ Lock codec during conversation: In certain cases, the system can switch to a common codec 

(advertised by both end devices) to avoid the transcoding during the call setup. Even though this 

is legal from the protocol’s point of view, many devices still cannot change codecs midstream. To 

avoid this problem, you must enable this feature. 

Once this is set, the system will not switch the codec during the call setup. This may introduce 

transcoding, which is a CPU-intensive job. Default is off. 

■ Packet length (in ms): This is the ptime parameter in the session description protocol (SDP). 

The default is 20 ms. 

■ Multicast IP Addresses: Set this to an IP address if you want the system to send and receive multicast 

IP addresses on this network interface. If this is set to 20 ms, then the system will send out 

packets every 20 ms, which equals 50 packets per second. If both sides of the call are set to different 

ptimes, then the system will have to transcode them, which will degrade performance. 

■ Bind to specific IP address (IPv4): The system listens to SIP requests that are sent by the user 

agent from this IP address only. This is useful if you have a dual NIC machine and want to listen to 

SIP requests only on one interface. If this is left blank, then the system will listen to SIP requests 

on all the interfaces it sees in the machine. 

■ Bind to specific IP address (IPv6): The system listens to SIP requests that are sent by the user 

agent from this IP address only. 

SNMP 

Simple Network Monitoring Protocol (SNMP) is used to poll the system for information. Table 2-5 lists the 

information that can be made available to an SNMP management tool. 

■ SNMP Port: The SNMP port setting defines the port on which the system will listen for SNMP 

requests. By default, the port is 161. 

■ SNMP trusted addresses: This field lists the IP addresses that may send SNMP requests. If this 

setting is empty, the system will not accept any SNMP requests. Whenever a request is rejected, the 

system writes a log message. This field requires the IP address of the SNMP server. 

■ SNMP Community: An SNMP community is the group to which devices and management stations 

running SNMP belong. If you would like to change the community, you can do so from the 

web interface. It does not require a restart of the service. SNMP default communities are private 

(write) and public (read). The system, by default, is set to public. 


Table 2-5. Available SMNP Sensors 


OID Description Absolute Unit 

1.3.6.1.4.1.25060.1.1 Call legs Yes Calls 

1.3.6.1.4.1.25060.1.2 Registrations Yes Registrations 

1.3.6.1.4.1.25060.1.3 Messages Yes Minutes 

1.3.6.1.4.1.25060.1.4 Call attempts No Calls 

1.3.6.1.4.1.25060.1.5 Successful calls No Calls 

1.3.6.1.4.1.25060.1.6 Media CPU load Yes Value 0..100 

1.3.6.1.4.1.25060.1.7 Successful emails No Emails 

1.3.6.1.4.1.25060.1.8 Unsuccessful emails No Emails 

1.3.6.1.4.1.25060.1.9 Email alert flag Yes Value 0..1 

1.3.6.1.4.1.25060.1.10 SIP received packets No Packets 

1.3.6.1.4.1.25060.1.11 SIP sent packets No Packets 

1.3.6.1.4.1.25060.1.12 Allocated memory Yes Bytes 

1.3.6.1.4.1.25060.1.13 Calls Yes Calls 

1.3.6.1.4.1.25060.1.14.x Trunk calls Yes Calls 

1.3.6.1.4.1.25060.1.15.x Trunk status Yes SIP response code 

1.3.6.1.4.1.25060.1.16 Uptime Yes Seconds 

1.3.6.1.4.1.25060.1.17 Disconnects (no media) Yes Calls 

1.3.6.1.4.1.25060.1.18 Disconnects (media 

timeout) 

Yes Calls 

TFTP 

The TFTP ports are used for provisioning purposes. Many SIP devices use TFTP for automatic configuration. 

■ TFTP Port: The TFTP (Trivial File Transfer Protocol) port is on port 69 by default. If your 

machine has multiple network interface controllers (NICs), you may specify the IP address port to 

bind only to that port. 

■ Allow TFTP Write: Some devices write log files using TFTP, and this can be enabled with this feature; 

however, this feature makes it possible for users to write files that affect other devices, and this 

may introduce system instability and security concerns. Per the example above, you can also bind to 

a private IP address, which will make it more secure. 

■ NTP Port: The system can act as an NTP (Network Time Protocol) server for the network. If you 

want to run the NTP server on the system, use this field to provide a port number on which it can 

run. By default, the NTP Port field is blank so that the system doesn’t respond to NTP requests. 

The default port is 123, so if you populate it and restart the service, the system will respond to 

NTP. The phone gets its time from the NTP, so if the phone cannot reach the Internet, then a local 

NTP server is necessary. A popular, publicly available NTP server is pool.ntp.org. 


36 

Logging 


Logging is a powerful mechanism for tracking system activity. Once you have installed the system, the logging 

feature allows you to see how it works and how the system interprets the input to the system. 

General Logging 

■ Log Level (0-9): This field determines which log messages are included in the log. The range is 

between 0 and 9. Level 0 will show you only critical messages, while level 9 will show you all the 

available log messages of the system. Log level 9 should be used only for troubleshooting, as it creates 

additional load for the system and may generate extremely large log files. 

■ Log Length: This is the length of the internal log message buffer. This buffer is used to show the 

log messages in the web interface. 

■ Log Filename: To request that the system write log messages to a filename, specify that filename 

here. If you put a dollar sign into the log filename, the system will replace the dollar sign with the 

current day and automatically generate a new file everyday. This will ensure that the log files do not 

become too large over time. This is highly recommended; otherwise, do not log to a file at all. Also, 

be sure to delete old log files from time to time so that your file system does not get overloaded 

with too much logging information. 

One of the first log messages that you will see is the working directory. If the Log Filename setting 

does not contain a path, the system will write the log file to that directory. You can specify the 

directory during the installation process. 

Example: The setting log-$.txt will create a log file under the system working directory with 

the name log-yyyy-mm-dd.txt, where yyyy is the year, mm is the month, and dd is the date. 

If you have multiple drives, it is good practice to write the logs there (e.g., D:\logs). 

Warning: Please note that using a log file without a dollar sign as part of the name is a ticking time 

bomb. While the system may initially run fine, system instability is likely to occur as disk space 

becomes consumed. 

Warning: Be sure to lower the log level once the system is running, especially when you write the 

log messages to a file. Sooner or later, you will get a hard disk full error, which is a severe situation 

because the system will not be able to save runtime data. 


Specific Events 


You can enable or disable logging on a subsystem level (available subsystems are shown below): 

■ Log general events: These events are of general interest, for example, information about the working 

directory. 

■ Log SIP events: Events in this module relate to the SIP traffic of the system. 

■ Log media events: The system reports events about media processing, for example, a one-way 

audio RTP timeout. 

■ Log IVR events: This module logs events related to processing user input (e.g., all DTMF events, 

including the auto attendant and the mailbox). 

■ Log email events: If you want to troubleshoot the email server interaction, turn this module on so 

that you’ll see SMTP events. 

■ Log http events: This flag controls whether events in the internal HTTP server should be logged. 

■ Log registration events: When a device registers or deregisters, it appears in this module. 

■ Log SNMP events: SNMP events occur when an external SNMP agent requests information from 

the system (see also page 34). 

■ Log trunk events: Log events are related to the trunks (e.g., when a trunk registers for the first 

time or when an inbound trunk calls). 

■ Log SOAP events: This subsystem deals with SOAP input and output. 

■ Log TFTP and PnP events: This module includes events related to the built-in TFTP server. It 

also includes information related to plug and play. 

■ Log TLS events: This module logs TLS-related messages in the log file. 

■ Analyze audio levels (CPU intense!): This feature measures the audio levels on a call leg. The 

volume is measured in decibels (dB) relative to the maximum volume (0 dB is maximum loudness). 


38 

SIP Logging 


Logging is important when tracking down problems. The settings on this page are designed to help you 

troubleshoot. 

When the system receives or sends a SIP packet, it determines whether the packet will be logged and which 

log level the event will have. 

■ Log REGISTER: REGISTER packets deal with the registration of extensions or trunks, which 

can generate a lot of traffic. If you are not interested in the registration traffic, disable this setting 

(see Appendix B for more information on SIP messages). 

■ Log SUBSCRIBE/NOTIFY: SUBSCRIBE/NOTIFY deals with message waiting indications 

(MWIs), the LED state, and other used subscriptions. 

■ Log OPTIONS: OPTIONS can be used to keep the SIP connection alive. They are designed so 

that a user agent can see which options or methods the other user agent supplies. OPTIONS allow 

you to view the full SIP messages with all the SIP header information (when using log level 7). 

■ Log Other Messages (e.g., INVITE): All other packets usually belong to an ongoing call (e.g., 

INVITE, CANCEL, ACK, BYE). 

When you enable the logging for one of the previous categories, the SIP packets will be logged on log level 7. 

If your log level is below 7, the packets won’t show up in your log. 

■ Log Watch List (IP): The watch list filters the SIP packets by IP address. This feature is useful 

when you have a specific device that you want to watch in the system’s log. Enter the IP addresses 

into this field. 

■ Log Watch List: This setting defines the log level for the previous setting—Log Watch List (IP). 

The log level for the watch list is independent of the log level of the packets that are not on the list 

(and which are on level 7). This makes is possible to lower the log level and show the SIP packets of 

a specific device also on a lower log level. 

Example: If you want to log SIP messages from 192.168.1.113 only, configure the following settings: 

— Log Watch List(IP): Set the IP address (e.g., 192.168.1.113). 

— Log watch list: 7 

— Log Level: 8 

— Specific Events: Enable Log SIP events. This filters out all the events, since they are set at 8 

and log watch is set at 7. 

Turn off everything under SIP Logging. 


Retrieving SIP logging 


The SIP log is the most common logging that is requested. Use the following steps to turn on SIP logging: 

1. Navigate to Admin > Settings > Logging. 

2. Set Log Level to 7. 

3. Set Log Length to 300. 

4. Under “Specific Events,” enable the following settings: 

■ Log general events 

■ Log SIP events 

■ Log trunk events 

5. Under “SIP Logging,” enable Log Other Messages (e.g., INVITE). 

6. Click Save. 

7. Navigate to Admin > Status > Logfile, and click Clear. This will clear the page. 

8. Make the call. 

9. Navigate to Admin > Status > Logfile, and click Reload. 

Configuration 

This section shows you how to save and restore the system configuration, as well as load the configuration 

XML files. This will generate a complete backup of your system, including all temporary files, recordings, 

TFTP server files, etc. You can even schedule this task nightly so that you automatically have a fresh backup 

for disaster recovery. 

Important: The backup from the web interface is designed for smaller systems. If the working directory of 

the system is larger than 5 MB, it is best to use TAR or ZIP and move the backup to another storage place. 


40 


Save/Restore a Backup 

This backup will make a TAR backup of the whole configuration, including audio recordings. Because the 

file might get potentially large, you should perform this action in times when there is not too much going on 

in the system. This will prevent you from interfering with the service and will ensure that you get an integer 

snapshot of the system state. 

1. Use the “click here” link, as shown in the following image. 

2. When prompted, save the file on the local computer. 

Important: The size of the file is limited by the “Max. size of a configuration backup file” setting (see the 

performance settings on page 28). If the TAR file exceeds the configured size, the backup will fail and you 

will be returned with an empty file. Therefore, be sure to check whether the file size is okay after storing the 

backup. If there is a problem, consider making the file system backup manually. 

To restore the backup 

To restore a configuration, the file must be loaded through the web interface. As with saving the configuration, 

this may be time-intensive, so be sure to do this when no calls are on the system. The system restoral 

will first erase your existing configuration, so be careful with this step. 

1. Click Browse to find the file to be restored (the full path of the file is required). This file is the one 

you backed up in the previous section. 

2. Click Save to load the selected file. This will load the selected file and the system will have the new 

configuration. 

Warning: The Factory Reset button on this page will load a default configuration file from the file system. 

All data will be erased. 

Note: The global parameter (max_tcp_length) can accommodate the large TAR files. You will need to set 

this before running the backup/restore. Currently, this value is in bytes, so you will need to provide a value 

similar to 384000000 (for 384 MB). 

Request Configuration 

The system may receive its configuration information from a service provider. Please be careful when using 

this feature because it may overwrite your existing configuration without further warning. If you leave 

the URL empty, the system will use the default URL. Please note that your license key will be sent over the 

Internet (encrypted). 

If you have a web service that generates configurations dynamically, you can request the pull-down of a configuration 

from the web server. The system will initiate the web request on its own. This feature is useful in 

large-scale installations (many systems running as CPE devices) that involve a central configuration management 

database. 


Reload Configuration Files 


Configuration files ringtones.xml and pnp.xml can be customized for your system. These files can be 

downloaded from wiki.snomone.com and uploaded through the Browse option shown below. You can also 

place the files into the html folder and restart the system (downloads are available at wiki.snomone.com). 

See also “Customized Ringtones” on page 206. 

The files are detailed below: 

■ ringtones.xml: This file is used to describe the available Alert-Info or Call-Info header 

that should be sent in the different Alert-Info headers. Since some phone vendors handle these 

headers differently, this file addresses that. 

■ pnp.xml: This file is used to describe the files that are available for plug and play. This file also 

defines various parameters for different phone models and vendors (see Table 19-3). 

Schedule a Reboot 

To schedule software updates, you can request that the system reboot the server. (This applies to non-Windows 

servers only.) Note that during a reboot, all services will be shut down. You can schedule a restart on 

the next occasion, or you can instruct the system to wait until midnight. In any case, the system will restart 

only if there is no call active. 

Certificate 

The system supports HTTPS, TLS, and SRTP. These protocols require a digital certificate and a private key 

for secure communication (the private key will be used for encrypting messages). Certificates are used to 

indicate to your communication partner that you are who you claim you are. This is done using a third party 

who certifies your identity and issues you a certificate that comes with a domain name. The certificate will 

be checked by clients that need to trust you. Usually, certificates are used for web services; however, the same 

certificates can also be used for SIP services. The system can support multiple certificates, so you can have a 

certificate for each domain. 


42 

Purpose 


By using a certificate, you defend your installation against DNS redirection attacks. An attacker might get 

control over a DNS server (which you do not operate) and redirect all requests to their server. Although the 

attacker might be able to present the same certificate that you have, he does not have the private key that you 

used when you requested the certificate from the trusted third party. Therefore, the attacker will be unable to 

establish secure communication. This way, the user agent can check if the host that he contacted is really the 

desired host and deny the connection if the public and the private keys do not match. 

Getting a Valid Certificate 

Certificates are used in the system in two places. First, they are used to secure the traffic between the web 

browser and the web interface of the system. Second, they are used to secure the SIP traffic between the 

phone and the system’s signaling path. 

The system by default generates a certificate, referred to as a self-signed certificate. While this provides a reasonable 

encryption of the traffic, it does not ensure that the client is really talking to the server. For example, 

it could also talk to a person in the middle that is just relaying the traffic. This essentially means that the traffic 

is not private any more, and since most Internet browsers are very strict regarding checking of certificates, 

the user must explicitly accept the untrusted certificate. Also, some IP phones do only accept SIP traffic on 

connections that have valid certificates. While the user of a web browser can just click and accept the certificate, 

a user of a phone usually does not have such a choice and the connection just fails. 

Buying a Certificate 

When you buy a certificate, it must be known that you are really the one who is operating a server. Although 

the mechanisms for this process differ, all services require that you pay for the service and that your web 

browser is already set up to trust the certificate authority. This mechanism is suitable if you are operating 

a public service where it is not an option to load root certificates on many clients. You usually also need to 

specify which IP addresses are using this certificate for the service. 

Making Your Own Certificate 

If you have control over the clients, you may also generate your own certificates. For example, you can join 

the community at http://cacert.org and generate them there. You will need to load the root certificate into 

the clients that should talk to the snom ONE system. 

There are various other sites available which provide a similar service. You may also download the openSSL 

toolkit and compile your own certificate generator and set up your own trusted network. If you have already 

done this to secure your other office infrastructures (e.g., email or VPN), you can probably reuse the certificates 

for that. 

Certificate Size and Format 

Please use only 512 and 1024 bit certificates. The system currently has trouble handling certificates with 

other sizes. The security and the performance on these certificates is still reasonable. 

The format of the certificate must be base64-encoded. You must include the private key and the certificate in 

the upload. Please note that uploading the private key this way might be intercepted by an intruder. You can 

minimize this risk by using the localhost address from the local machine. 



In order to provide the key, just enter the ASCII string that you received from the trusted party, copy it into 

the text field, and press Save. The system will then present this certificate to HTTP and SIP connections that 

require secure communications. 

Access List 

If your system is under a denial of service attack, the system will automatically block the IP address that is 

attacking the system. You can also add IP addresses to the Access page and place a Block on them so that 

they will be unable to access the system. Just enter the IP address, the net mask, and the access type, and 

click Create. Enter only as much information as needed by the net mask; for example, if the net mask is 

“255.255.0.0,” enter 192.168.0.0. To delete an entry, click the Delete button. 

Changing the entry does not require a restart of the system. The changes take effect immediately. 

You may specify IP addresses with their netmask and their policy. If there is no match, packets will be accepted. 

There are several reasons why a system administrator might want to define who has access to the 

system: 

■ Protection against denial of service attacks: If you are operating the system on publicly available 

addresses, there is always the risk that someone will try to interrupt the service. Although the system 

has several protections against such attacks, it might be easier to rule out such attacks from the 

beginning. 


44 


■ Limiting the service to authorized addresses: You might also want to limit the service to specific IP 

addresses only. For example, while you might allow users to register their IP phones in the office, 

you might allow only selected users with their associated IP addresses to register their phones from 

home. 

The motivation for the list is to provide the firewall type of functionality within the system application and 

reduce the chance of unauthorized access to the system. 

The access control is not only limited to SIP, but it also applies to all other protocols on the system, including 

HTTP, TFTP, and SNMP. When the system acts as a client (for example, when performing DNS requests), 

the rules do not apply. Using 0.0.0.0 in the IP field specifies that everything will be accepted. In 

addition, if you are getting a lot of requests from a particular source, the system will automatically add them 

to the access control list and block them. 

How the Access List Functions 

When a packet reaches the system, the system checks the list of enabled and disabled addresses for a match. 

If the request is ignored, the system discards the packet without answering. When the system checks the list 

for matches, a match occurs if a “source address” matches a “check address” with the mask. More specific addresses 

are checked first, making it possible to define exceptions to the general rule. Also, the system checks 

IPv4 and IPv6 addresses separately. If there is a match, the system checks for the type. If the type is Allow, 

then the system accepts the packet. If the type is Block, then the system blocks that request. If there is no 

match in the list, then the request is accepted. If the list is empty, the access control is disabled. This is the 

default behavior after the installation of the product. 

For UDP-based requests, this is relatively easy—the request is just not answered. But because the UDP port 

is open, there is no ICMP request sent to the origin, which means if someone wants to attack the system, it 

might be possible for the attacker to figure out that there is an open port. But since the system just discards 

these messages, the damage is limited. 

For TCP ports, the situation is more complicated. In Linux, there is no way for an application to determine 

where a TCP connection is coming from until the connection is accepted. This is why the system first accepts 

the connection and then examines whether the connection was allowed or not. If the connection was 

not allowed, then it is turned down immediately. In Windows, there is a special system call that first checks 

where the connection is coming from. If the source is not enabled, then the system does not accept the connection. 

However, because the operating system has already answered the TCP connection request with an 

acknowledge, in Windows it will be obvious that an application is running on the ports. 

The behavior is similar to a firewall. However, especially for TCP, a firewall will keep the traffic completely 

out. Someone testing the system will not get a response for a TCP request if the source IP address is not 

listed. 

Example 

Table 2-6 shows a scenario in which all users in the LAN are given access, but access from the public Internet 

is not allowed, except for two employees working from home and a trunk that comes from a service provider 

with a small range of IP addresses. 

Table 2-6. Access List Scenario 


First 

entry 

Next 

entry 

Next 

entry 

Next 

entry 

Next 

entry 


Address Net Mask Type Description of Result 

127.0.0.1 255.255.255.255 Allow This will ensure that 

you can always access 

the HTTP interface 

from the local computer. 

192.168.0.0 255.255.0.0 Allow This will ensure that 

everyone in the LAN 

can access the system. 

0.0.0.0 0.0.0.0 Block This entry will disable 

all packets by default 

(enter this last; otherwise, 

you will be 

unable to access the 

system). 

213.1.2.3 255.255.255.255 Allow This will give the remote 

worker access to 

the system. Repeat the 

same entry for other IP 

addresses. 

12.23.34.45 255.255.255.248 Allow This entry is intended 

for the IP addresses of 

the ITSP. 

Controlling User Page Appearance 

The UserPage-Control link allows the administrator to block certain settings from users. These settings are 

shown in Figure 2-3 and correspond to the highlighted settings shown in Figures 2-4 through 2-7. 

Figure 2-3. Blockable User Settings 


46 

General Control 


The General Control settings allow you to prevent the user from viewing the password fields and certain 

monitoring settings. 

■ SIP password: By preventing the user from seeing the SIP Password setting, the administrator 

can ensure that users will be unable to change their SIP passwords. This can save the administrator 

time spent in dealing with registration problems that result from changed SIP passwords. 

■ Web password: By preventing the user from seeing the Web password setting, the administrator 

can ensure that users will be unable to change their web passwords. This will prevent the user from 

forgetting a changed password and the administrator from having to issue a new one. 

■ Calls, presence, PAC watch field: This setting impacts the following user settings: 

— Watch the calls of the following extensions: This setting allows users to monitor the status 

of other extensions. Some users will need this setting (a boss who is monitoring certain 

employees, a secretary who needs to monitor the boss’s extension, etc.). Other users, such as 

those working in a call center, will not need to monitor other people’s extensions. 

— Watch the presence of the following extensions: This setting allows you to see a person’s 

presence status (e.g., away, busy, DND) from the WAC and from some phones. 

— Watch the following accounts on PAC: This setting allows users to monitor extensions on 

the WAC. Not all users will need access at this level. 

■ User can select the address book: This setting is applicable to the address book of the snom 

phone. It allows the user to upload the address book from the system to the snom phone directory. 

Figure 2-4. Blockable User Settings (Settings > General) 


Redirection Control 


The Redirection Control settings allow you to prevent the user from viewing the redirection settings: 

■ Incoming anonymous calls: This setting allows users to control the treatment of anonymous 

callers (reject them, pretend to be busy, ask for name, etc.). In some situations, administrators may 

prefer to determine how anonymous callers are handled. 

■ Call forward on no answer timeout: This setting allows users to specify the number of seconds 

that should lapse before the call is directed to voicemail. In situations where voicemail is not used, 

such as call centers, this setting is not necessary. 

Figure 2-5. Blockable User Settings (Settings > Redirection) 

Mailbox Control 

The Mailbox Control settings allow you to prevent the user from viewing the following voicemail-related 

settings: 

■ Mailbox enabled field: This setting allows users to enable their voicemail. In situations where 

voicemail is not used, such as call centers, this setting is not needed. 

■ Time until mailbox picks up: This setting allows users to set their own mailbox pickup time and 

let their phone ring for longer periods of time before it goes to voicemail. 

■ Maximum number of messages: This setting allows users to set the number of message that will 

be saved in their mailboxes. If space is an issue, administrators may want to place their own limit 

on this setting and prevent the user from changing it. 

■ Mailbox escape account: This setting gives users the option to change the direction of a call once 

the caller presses 0 to speak with the operator (calls are redirected to the escape account). In many 

cases, users will not have the option of directing their calls to another extension and will not need 

this setting. 


48 


Figure 2-6. Blockable User Settings (Settings > Mailbox) 

Email Control 

The Email Control settings allow you to prevent the user from viewing the following email-related settings: 

■ Send a mailbox message by email: This setting allows users to send voicemail messages to their 

email accounts. In situations where voicemail is not used, such as in call centers, this setting is not 

needed. 

■ After sending an email: This setting allows users to specify how messages that have already been 

read should appear (marked as new/read or deleted). 

Figure 2-7. Blockable User Settings (Settings > Email) 


ADMINISTERING THE DOMAIN 

Chapter 

3 

The snom ONE uses the notion of a domain since all IP systems are based on domains. A domain is essentially 

a logical group of users on the same network or computer that is linked to the Internet (the name must 

be a resolvable entity on the Internet via DNS so that it can get its IP address). Email is a type of IP system 

that employs domains; for example, users belong to domains like yahoo.com, google.com, and aol.com. 

User Host 

jack google.com 

jpiccolo yahoo.com 

rnti aol.com 

The snom ONE telephone system is similar in that it relies on domains. The snom ONE supports multiple 

domains and can be used to facilitate multi-tenant use. 1 

Host 

pbx.companya.com 

pbx.companyb.com 

pbx.companyc.com 

Users within the same domain on the snom ONE can call each other without going through a trunk. They 

can also access the same address book and share features, such as call monitoring and call park/pickup. 

Naming the Domain 

By default, the system installation creates pbx.company.com and associates localhost as alias to it. 

Default Domain Alias 

pbx.company.com localhost 

The localhost domain will match all requests that cannot be matched to a domain name within the domain 

list. This makes it possible to run the system on changing IP addresses without changing the name of 

the domain. 

You may use several names for a domain; this is known as domain aliases. However, one of the names will be 

the primary (canonical) name for the domain. The system will use that name whenever it has to generate a 

name for the domain. 

Domain names may also be IPv4 addresses when you cannot change the DNS. However, if you are assigning 

IP addresses by DHCP, be careful with this method. Make sure that the host is always running on that 

IP address. You may mix IPv4 names with DNS addresses, and you may also later rename the domain names 

and reassign the primary domain name. 

1. snom ONE blue supports multi-tenant use for up to five companies. Multi-tenant use is not available for snom 

ONE yellow and snom ONE free . 


50 


Changing the Name of the Domain 

To change the name of the default domain so that it reflects your company domain: 

1. Navigate to Domains and click the Edit icon. 

2. Replace pbx.company.com with your own domain name. 

3. Enter an alias if desired. 


5. Click List to view the domain. The new domain will be displayed. 

Note: Setting up a domain on the system does not imply that the necessary DNS records have automatically 

been set up. If you can, set up your DNS so that users from other domains can find the group by standard 

DNS name resolution (e.g., pbx.company.com = 75.1.2.4, the IP address of the system). 

Assigning a Domain Administrator 

Just as the system is managed, so must the domain be. This can be the role of the administrator or a user 

who is assigned by the administrator (the administrator will always have access to both system and domain 

settings). Whether the administrator assigns someone or not is usually based on the size of the company and 

how much responsibility the administrator wants to delegate. The responsibilities of the domain administrator 

involve extension accounts, trunks, and dial plans, but the administrator can also limit many of these 

settings. 

The permission to manage the domain should be given only to those who are allowed to change passwords, 

create or delete accounts, change dial plans and rates, and so forth. Permission is given from the Permission 

tab in the user’s extension account: 


Chapter 3: Administering the Domain 51 

To log in to the domain, users will need to log in as Domain Administrator (with their extension credentials) 

as shown below: 

Once logged in, domain administrators will be able to access to the domain settings shown below. These 

settings are detailed in Table 3-1. 

Table 3-1. Menu Tree of Domain Tabs 

Web Interface 

Tab Description 

Settings 

Domain 

Settings 

This page allows you to configure the necessary settings for the domain. From this 

page, you can decide which dial plan, language, and music on hold source the domain 

will use. You can also enable features such as camp-on, trust caller-ID, and voicemail 

timeout. 

Star Codes This page lists the star codes that are used to control the system. From this page, you 

may define your own set of star codes. 

Address This page lists the names and contact information of those who are part of the domain. 

Book From this page, you can add individual entries to the address book, or you can upload 

a CSV file to add a large number of names. 


52 


Web Interface 

Tab Description 

Buttons This page provides a list of all the button profiles that are available on the domain. 

From this page, you can create additional button profiles. 

Accounts This tab provides a listing of the accounts that have been created on the domain. This 

includes extensions, hunt groups, agent groups, service flags, paging accounts, and any 

other account that exists on the domain. Accounts can be created, edited, and deleted 

from this page. 

Trunks This tab provides a listing of all the trunks that have been created on the domain. New 

trunks can be created at this page, and existing trunks can be edited and deleted. 

Dial Plans This tab provides a listing of all the dial plans that have been created on the domain. 

New dial plans can be created at this page, and existing dial plans can be edited and 

deleted. 

Status 

General This page shows you a snapshot of your system: release of software, type of software, 

expiration date, etc. 

Logfile This page displays the SIP messages, which can be used for troubleshooting. 

Call Log This page displays a list of calls that have been made through the system. 

Calls A list of active calls are displayed on this page. 

Default Domain 

The system comes with a preconfigured domain that includes ten 2-digit extensions and numerous domain 

accounts (Figure 3-1). To view the accounts, click Accounts > List. 

Figure 3-1. Default Accounts 



Note: A yellow triangle symbol indicates that the account is missing either the SIP password or the web 

password. 

Configuring the Domain 

From the Settings tab, you can configure many of the domain’s settings: 

■ Name of the Domain: This field allows you to give the domain a descriptive name. The name can 

contain special characters and spaces and may include upper and lowercase characters. This setting 

is used when the system sends an email with the call data records for the domain. This name is 

independent from the domain name (which is set by the system administrator). 

■ Default Dial Plan: This dial plan will be used by the domain, but users can override this with 

their own dial plan based on need. Most of the accounts in a domain will use the default dial plan. 

If you do not set a default dial plan, you will likely have problems placing outbound calls. Therefore, 

it is strongly recommended to choose a default dial plan. 

■ Default IVR Language: This setting can be configured by the administrator and is not needed on 

the domain level in single-domain environments. 

■ Tone Language: This setting can be configured by the administrator and is not needed on the 

domain level in single-domain environments. 

■ Default Web Language: This setting can be configured by the administrator and is not needed 

on the domain level in single-domain environments. 

■ Music on Hold Source: You can select the source of music that will be used when a call is put on 

hold. The party who is on hold hears the music that is chosen here. See also Chapter 20, “Music 

on Hold.” 

■ Timezone: The system can handle multiple time zones simultaneously, making it possible for 

users to select a time zone that corresponds with their geographic location. This setting influences 

the time on mailbox messages and the telephone display. Before the system can apply the time 

zone to the user’s extension, it needs a time zone configuration file (timezones.xml), which is 

encoded in XML and looks like the following: 

The name of the time zones is reflected in the lang_xx.xml file, where xx is replaced by the 

respective language (see Table 17-5 for a list of country code identifiers). 

■ Country Code: The country code plays a major role in interpreting telephone numbers. Three 

modes are available for this setting. 


54 


Table 3-2. System’s Interpretation of Telephone Numbers 

Modes Description Input 

As Is If this setting is left empty, the system does not change telephone numbers. 

In this case, you must make sure that users, PSTN gateways, and service 

providers use the same method of representing a number (for example, 10 

or 11 digits in the United States). Choose this mode when the other modes 

are causing problems. 

Leave empty. 

NANPA If you enter a 1 into the Country Code field, the system will interpret telephone 

numbers according to the NANPA scheme (North American Numbering 

Plan Administration), which means that international numbers will 

start with 011 and numbers that have 10 digits will be written in the (xxx) 

xxx-xxxx scheme (JavaScript takes care of this in the web interface). This 

scheme also applies to countries other than the United States, like Canada. 

Apart from the readability of numbers, this scheme has the added benefit 

of automatically converting differently represented numbers (11-digit or 

10-digit) into a global format. The area code must be 3 digits and is used for 

numbers that have only 7 digits. 

1 

ROW If you put another country code in this field (for example, 33 for France), Any country 

the system will interpret numbers according to a “rest of world” scheme, code (e.g., 

which means that international numbers start with 00 and national numbers 

start with 0. The area code is used to determine if the short number 

format can be used (local call). 

33) 

The Country Code setting corresponds to the Rewrite global numbers setting (page 85): 

■ Area Code: This is the telephone area code for the domain. This is generally NPA in the NPA- 

NXX-XXXX number (e.g., 978, 212, etc.). 

■ Default ANI: You can configure each domain with a default ANI (Automatic Number Identification). 

ANI is a service that tells the recipient of a telephone call which telephone number made the 

call. In most cases, the ANI is used in the From field in the SIP packets or the caller ID. 

■ Trust Caller-ID: This setting is used to specify whether the caller-ID should be used to authenticate, 

rather than asking users to enter their PIN numbers. 

■ Emergency Numbers: You can specify special destination numbers when an extension dials an 

emergency number. The number(s) that trigger the mechanism must be listed in this setting. If you 

have more than one emergency number, use a space between the numbers (e.g., 911 199). By 

default, this setting is blank (i.e., it is not set to 911). 



■ Use last 7 digits when cellphone matching: If this is checked, the system will use the last 7 digits 

to match cell phone numbers. This is useful in places where only 7 digits are used in the caller-ID. 

■ Send welcome email when an extension is created: If this setting is checked, new users will automatically 

receive a welcome email. 

■ Delete all aliases when deleting the primary account: Aliases will be deleted when primary accounts 

are deleted. 

From/To Headers 

■ From header when switching the domain: This setting allows you to set a new From header. 

It is used when the system switches the domain context for calling an extension. You may enter 

several dollar-escaped characters into this field using the values shown in Table 3-3. For example, 

the string $d: $n will produce an output like Company A: John Smith if the source domain 

name was set to Company A and the original content of the header was John Smith. If this setting 

is empty, the system will not change the header. 

Table 3-3. Parameters Used for Setting the From and To Header 

Value Output Example 

d The domain name of the calling 

(source) domain 

Company A 

e The domain name of the called (destination) 

domain 

called.domain.com 

n The original content of the header John Smith 

x Parameter 1 of the called extension 

y Parameter 2 of the called extension 

z Parameter 3 of the called extension 

x, y, and z are fields that can be used for additional information. 

■ To header when switching the domain: This setting allows you to set a new To header. It is 

used when the system switches the domain context for calling an extension. You may enter several 

dollar-escaped characters into this field using the values shown in Table 3-3. 

■ Change names in To/From-headers: Usually it is okay if the system changes the names in the 

To/From headers in the SIP packets. For example, this makes sense when the address book matches 

the caller-ID and the system should present the address book entry. However, if the system is 

used as a SIP device in an operator environment, you may want to make sure that the system does 

not change headers. In this case, you can turn this flag on. 

CDR Settings 

■ CDR URL: This setting allows you to configure where and how the CDRs are written for a specific 

domain. This setting will override the configurations that are made at the system level for the 

domain. The system supports several types of CDR generation formats, apart from writing them 


56 


into the /cdr directory in XML format (see the CDR URL setting on 

page 26). 

Billing Settings 

■ CMC authentication for outbound calls: This setting works in 

tandem with the “C” checkbox in the dialplan (page 91). When the 

“C” is checked, this domain setting will be used; otherwise, it will be 

ignored. This setting dictates at what point, after a CMC code has been 

entered into the phone, the system will begin looking for a match. 

When Use address book is selected, the system will automatically take 

the CMC from the address book record. 

■ Billing Tones: This field allows you to configure whether the domain 

users will hear a beep during the call when using the pre-pay feature of 

the system (pre-pay is discussed in Chapter 15). 

■ Credit for outbound calls: This setting allows you the place a limit on outbound calling for 

the entire domain. Once the extensions as a group have reached the maximum, outbound calls 

can no longer be placed. This setting can also be configured at the extension level (see Credit for 

outbound calls in page 107). 

Voicemail/Mailbox Settings 

■ Time until user’s mailbox picks up: This setting defines the number of seconds the system will 

wait before redirecting a call to a user’s mailbox when there’s been no pickup. You may override 

this setting in the settings for a specific extension. If you plan to use cell phones with the system, 

choose a long voicemail timeout (e.g., 30 seconds), as the call setup time for a cell phone call can be 

lengthy and the user might need to search for the cell phone. If you do not plan to use cell phones, 

you can choose a shorter timeout (e.g., 15 or 20 seconds). 

These next two settings allow you to specify memory capacity for voicemail messages. When configuring 

these settings, consider the size of your system and the information shown in Table 3-4. 

■ Voicemail Size: This setting determines the number of messages that can be stored 

in a user’s voicemail box. Although this setting can be overridden by the user (see 

the Maximum number of messages setting on page 267), a reasonable default 

should be set at the domain level (20 is a good default number). If this setting is 

left blank, the number of messages that can be stored in a user’s mailbox will be 

unlimited. 



■ Maximum Voicemail Duration: This setting specifies the length of a mailbox 

message. Typically, a mailbox message should be no longer than 2 minutes. If this 

setting is left blank, the length of any one message can be unlimited. 

Table 3-4. Voicemail and Disk Space Requirements 

Settings Disk Space Required* 

Number of Users Voicemail Size Voicemail Duration Each User All Users 

10 25 2 minutes 5 MB 50 MB 

10 100 2 minutes 20 MB 200 MB 

100 25 2 minutes 5 MB 5 GB 

100 100 2 minutes 20 MB 20 GB 

*1 minute of voicemail or call recording requires approximately 100 kb of disk space. Based on this 

measurement, you can record 165 hours of voicemail per gigabyte. 

■ Voicemail PIN Digits: This setting specifies the number of digits that will be 

required for the voicemail PIN. Although any number larger than one digit is acceptable, 

we recommend you require four or five digits. See also the PIN setting in 

the extension account (page 104). 

■ Require Entering Mailbox PIN: When this setting is enabled, users will be required to enter their 

PIN before retrieving voicemail messages. While retrieving messages should be simple, many office 

layouts may make it too easy for an unauthorized person to listen to another’s voicemail messages 

without authorization. This setting can be used to prevent that across the domain. 

■ Calling own extension number goes to mailbox: By default, the system will send a user to the 

voice mailbox if he dials his own extension. However, in some situations it is useful to have the 

system call the registered extension. 

■ Mailbox Escape Account: If the Mailbox Escape Account setting is set, a user who reaches the 

mailbox can press 0 to get to the account that you specify here (the user must press 0 before the 

beep). The number must be an internal number. 

■ Mailbox Direct Dial Prefix: This setting allows users to call directly into their mailboxes or transfer 

a caller directly to a user’s voicemail without having to wait for the voicemail timeout. Typically, 

this prefix is an 8, so by dialing 8511, the caller goes directly to the voicemail greeting. 

■ External Voicemail System: This setting is used if you want to use an external voicemail system 

(e.g., Microsoft Exchange 2007/2010). The setting requires a telephone number that can be dialed. 

You can include replacement fields which are used in the caller-ID representation for outbound 


58 


calls. The dial plan for the called extension will decide which trunk will be used to send the call. 

See Appendix C for more information on Exchange. 

■ Mailbox Explanation Prompt: If a user records a personalized message, the system can deliver another 

message after the prompt that explains that the caller may now leave a message. If this setting 

is enabled, the voicemail system will explain the caller’s options after the playback of the personal 

recording. For the standard greetings, the system always explains the options. 

■ Offer Camp On: This setting allows you to offer Camp On to callers, rather than limiting them 

to leaving voicemail. Callers will have the option to press 1 for a callback. Turn this off if you are 

using external voicemail systems such as Microsoft Exchange. 

■ Call forward on no answer timeout: This setting establishes the number of seconds that the system 

will wait before performing an timeout action. It is advisable that this setting be less than the 

voicemail timeout; otherwise, the call will not be forwarded but will go to voicemail instead. 

■ Park Reminder: When this setting is enabled, the system will call back an extension that has 

parked a call, as a reminder that a call is still parked. Specify the amount of time that should pass 

before the reminder is made. 

■ Play envelope information before playing the mailbox message: When this setting is enabled, 

envelop information will be offered to the user while retrieving voicemail messages. To retrieve the 

information, the user can press #5. (If this setting is not turned on and the user presses 0 to hear 

more options, the envelop information will be offered at that point.) 

Accounts Page Display 

■ Accounts page size: This setting determines how many accounts appear on one page. The options 

range from 25–1000 accounts (all accounts can also be viewed). 

■ Hide aliased accounts under accounts page: This setting allows you to hide all alias accounts. 

(Aliases are DID numbers and are used for direct dialing.) When this setting is set to Yes , aliased 

accounts will not be displayed: 



■ Display hyperlink for the disabled accounts in the accounts page: When this setting is set to 

Yes , disabled accounts will be hyperlinked to the actual account: 





Accounts can be disabled and enabled using the Disable and Enable buttons at the bottom of the 

Accounts page. Additional actions are shown in the dropdown. 

■ Display MAC address for the accounts in the accounts page: When this setting is activated, 

the MAC address for each account (if one is available) will be displayed. When the permanent or 

temporary assignment mode is used for obtaining MAC addresses, a question mark or asterisk will 

appear in the MAC address field. 

■ Display Email address for the account in the accounts page: When this setting is activated, the 

email address for each account (if one is available) will be displayed: 

■ Display cell phone for the account in the accounts page: When this setting is activated, cell 

phone numbers that have been configured to an account will be displayed: 


60 


All three fields can be displayed at the same time: 

Recording Defaults 

This section allows you to define domain-level call recording settings. Recorded calls will be saved in the recordings 

directory (see also Chapter 17). 

■ Record incoming calls from hunt group: This setting controls the recording on an extension for 

the calls coming from the hunt group for the entire domain. 

■ Record incoming calls from agent group: This setting controls the recording on an extension for 

the calls coming from the agent group for the entire domain. 

■ Record incoming calls from extension: This setting controls the recording on an extension for 

the calls coming from another extension for the entire domain. 

■ Record outgoing calls to internal numbers: This setting controls the recording on an extension 

for the calls going to another extension for the entire domain. 

■ Record outgoing calls to external numbers: This setting controls the recording on an extension 

for the calls going to external numbers for the entire domain. 

■ Record outgoing calls to emergency numbers: This setting controls the recording on an extension 

for the calls going to an emergency number. 

Email Settings 

The system can be configured to send out emails on important system events, but it needs an SMTP server 

to do this. From this page, you can configure the SMTP server settings (these settings can also be made by 

the administrator by going to Admin > Email). 


Midnight Events 


At midnight, the system can perform several maintenance tasks, including the generation and sending of 

CDR reports. 

■ Send daily CDR report to: The system sends daily CDR reports (sample shown in Figure 3-2) to 

the email addresses listed in this field. Use semicolons between multiple email addresses, since email 

addresses may contain spaces (e.g., Fred Feuerstein ; Carl Clements 

). To test this feature, click the Try button once you’ve entered an 

email address rather than waiting until midnight. Keep in mind that this is just a link, and you may 

have to save your list before clicking the link. The report shows a running list of the previous day’s 

calls: 

Figure 3-2. Sample CDR Report 

■ Reset DND Flags: DND is a major problem if users forget that they put their extensions on 

DND and then open a trouble ticket next morning because they’ve stopped receiving calls. When 

the Reset DND Flags has been enabled, the system resets all DND flags of the domain during 

midnight. 

■ Reset Hot Desking: Forgetting to deactivate a Hot Desking situation has similar consequences 

to forgetting to turn off DND. Because of this, the system allows you to reset the hot desking at 

midnight. 

■ Reset Block CID: This is another setting you may need to reset at midnight. 

■ Send emails to CDR email address for domain level blacklisted calls: When someone calls into 

the auto attendant, hunt group, or agent group and is put onto the blacklist, the system will notify 

the person who receives the CDR report. 

■ Log out agents from all ACDs: All agents will be logged out of all agent groups. 


62 

Provisioning Parameters 


When the system automatically generates files for the configuration of extensions, it needs a few settings 

from the domain. 

■ Default PnP Dialplan Scheme: This setting allows you to set auto-dial and configure timeoutbased 

dialing. These settings are used by the phone and not the system (see Chapter 5 for the dial 

plan that is used by the system). The PnP Dialplan Scheme setting allows you to tell the system 

how many numbers your extensions contain (either 2, 3, or 4 digits). This will initiate automatic 

dialing once the required number of digits have been entered. If your extensions are all 3 digits 

long and include only digits 2 through 7, you should choose 3-digit extensions [2-7] 

xx. Then if a user dials 201, the system will identify that an extension has been dialed and will not 

wait for more digits. If NA is set and Country Code 1 is used, then anytime someone dials 1, the 

system will know to dial that country. For this reason, extensions should not begin with a 1. 

— User must press enter: When this scheme is selected, the user of the extension must always 

press the confirm button on the phone. This is similar to a cell phone, where the user is required 

to press the green send button on the device. This scheme can prevent a lot of problems, 

as it is difficult to predict how many digits the phone must collect before it has enough 

digits for a complete number. 

— When the user is located in the North America area (www.nanpa.com), the length of the 

number is predictable for national calls and calls of other extensions in the domain. However, 

calls to international destinations or star codes are difficult to predict, so users must press the 

confirm button to start the call. 

— For other countries, the “Europe” scheme might be appropriate. As with the rest of the world 

outside of NANPA, the length of a telephone number is practically unpredictable and users 

must press the confirm button to start the call. The “Europe” scheme will dial automatically 

only if the destination is another extension. 

Note: If you want to create a customized dial plan, you can do this from either the snom phone or 

through the binary files of the system. In the case of the latter, you will need to request the appropriate 

file from snom and place it into the snom/snomONE/html directory (you may need to 

create the html directory). After you have modified the file, save it and reboot both the phone and 

the system. 

Example: In the event you wanted to exclude numbers higher than 5 from the default of [2-7] 

xx so that users wouldn’t be required to enter a 1 when dialing local numbers beginning with 604 

or 778, you would need to replace [2-7] with [2-5] in the file named snom_3xx_dialplan_ 

usa3.xml (assuming you are using a snom phone). 



■ Authentication User/Password: This setting makes it easier to carry out domain level settings of 

phones. Using this setting, the administrator can create identical usernames and passwords for all 

devices in a domain. This password will bypass users’ passwords. However, an extension@domain is 

still required. The default username is admin, and the default password is password. 

■ Authentication PIN: This setting allows you to override the user’s PIN, which is used on the 

phone itself. Administrative access to the phone allows you to reset the phone. 

Domain Address Book 

The domain address book allows users on the same domain to easily call each other and to see who is calling. 

Users can search for contacts from either the web interface or from their IP phones. From the address book, 

contacts can be whitelisted or blacklisted and CMC codes can be added to a contact’s information. 

Creating a Domain Address Book 

Adding Contacts Manually 

Enter the contact’s information into the Create New Entry form shown below: 

■ Number: The telephone number of the contact. This is not a SIP URI. 

■ Speed Dial: You may use a two-digit star code to speed-dial this contact. Note that these star codes 

cannot overlap with any of the standard star codes. 

■ CMC: The CMC identifies the customer in the CDR report and is used to expedite billing in 

offices that bill their clients for phone calls, such as law offices and other businesses that track time 

spent with clients. As long as the CMC has been added to the address book and the caller is calling 

from the phone number that is associated with that CMC, the CMC will automatically appear in 


64 


the CDR report. The CMC can also be entered directly into the display of the phone (by pressing 

the CMC soft key) if the person is calling from a number other than the number that is in the address 

book. 

■ Contact Type: The Contact Type dropdown allows you to blacklist or whitelist a contact. 

Once contacts have been added to the domain address book, their information will be displayed to the person 

they are trying to reach. 

Adding Contacts Using CSV Files 

Domain address books can also be compiled through CSV files (see Appendix A). When importing a CSV 

file, you have to option of clearing existing contacts out of the system before importing new ones. 

Adding Contacts Automatically 

Contacts who call into an agent group can automatically be added to the address book (see Add to the address 

book on page 146). 

Adding Contacts by Modifying DID Information 

When the system has been configured to automatically add agent group callers to the address book, the 

caller’s DID will be added to the list of address book entries. Using this information, the system can route 

repeat callers to the same agent. However, agents will be unable to identify the caller unless the contact’s 

information is added to the address book. 

1. Click the contact’s DID. 

2. Enter the contact’s information in the Edit Address Book Entry form. 


Searching for Contacts 

Web Interface 


Users can search for contacts by entering the contact’s first or last name into the search box: 

Phone Display 

If the administrator has enabled the User can select the address book setting (Admin > Settings > User 

Page-Control), users can also search through the domain address book from their IP phones: 


66 



TRUNKS 

Chapter 

4 

Trunks are used to route calls in and out of the system. When the system receives a call from an extension, 

it checks to see if another account is being called. It does this by looking at the To field of the SIP packet. 

If it’s not another account, then the system consults the dial plan to determine which trunk the call should 

be routed to (extension-to-extension calls do not require trunks). When the system receives an outside call, 

it looks at the IP address it came from to see if the IP address is included the trunks table, and then routes it 

accordingly. 

In the past, trunks were physical lines, and the number of calls that could be placed was limited by the number 

of lines. Figure 4-1 shows an example of a traditional trunk. In the example shown below, a telephone 

switch is connected to a corporate private branch exchange (PBX) through one physical T1 line. A separate 

line (or channel) is needed for each of the 23 calls that are on the PBX. 

Telephone 

Company 

1 2 3 4 5 6 7 8 9 10 11 12 

13 14 15 16 17 18 19 20 21 22 23 24 

T1 Trunk (24 channels) 

Figure 4-1. Traditional Trunking—Physical Lines (limited by lines) 

SIP trunks are virtual lines, so the number of calls that can be made is limited only by bandwidth, rather 

than actual channels. SIP trunks allow a system to provide access for many individual telephones instead of 

requiring individual lines for each phone. 

Trunks can be dedicated to incoming, outgoing, or two-way traffic. Incoming trunks carry calls into the system 

from an external network (i.e., PSTN, cellular, ITSP). Outgoing trunks carry calls away from the system 

to external networks. Two-way trunks can be used for both sending and receiving calls. Figure 4-2 shows a 

SIP gateway trunk connecting to the PSTN and a SIP registrations trunk connecting to the ITSP. 


PBX

68 

Cell Phone 

Access 


PSTN 

SIP 

Gateway 

Trunk 

LAN 

SIP Phones 

ITSP 

(data router) 

SIP 

Registration 

Trunk 

External 

Internal 

Figure 4-2. Using SIP Trunks for Communicating with External Networks 

There are three types of trunks with the snom ONE telephone system: SIP registrations, SIP gateway, and 

SIP proxy. All act as an entry and exit point to an external network. The type of network you are connecting 

to will determine which type of trunk you will need. 

Trunk Types 

SIP Registrations 

When connecting to an ITSP (Internet telephone service provider), you may be required to use a SIP registration 

trunk due to the fact that the ITSP does not need to know your public IP address. They will learn it 

through the registrations process (see Appendix B for information on SIP registration). 

The SIP registrations trunk is a popular type of SIP trunk and is easy to set up once the ITSP account has 

been established. It requires the system to register to the provider, just like an IP phone registers with the system. 

Like an extension that has been registered with the system, SIP registration trunks require a user name 

or account (usually the DID), a password, and the IP address or domain name of the SIP or proxy server. 

The advantage of the SIP registrations trunk is that the IP address is dynamically bound to the SIP registration, 

which allows the registration to be used from any IP address. Although service providers often assume 

that a trunk registration involves an IP phone or an ATA rather than a trunk, many phones can sit behind 

the snom ONE telephone system and share the resources of the trunk (Figure 4-3). However, most service 

providers limit the number of calls that can be made over a SIP trunk and use a charging method that simulates 

the charges accumulated by a physical legacy TDM trunk (i.e., the number of call paths). 

Although SIP trunks are virtual trunks and can theoretically have an infinite number of calls, SIP trunks are 

limited by the amount of bandwidth that is available to handle the calls. 


586 

587 

588 

Figure 4-3. SIP Trunks 

PSTN 

GSM 

Network 

Internet 

ITSP 

588 

Chapter 4: Trunks 69 

6031234567 

6031234566 

510 511 

6031234577 

6031234588 

A SIP registration trunk can also be used to connect two or more snom ONE systems together, as in a 

branch/head office configuration (see “Connecting Branch Offices Together” on page 86). 

SIP Gateway 

The gateway trunk is typically used to talk to a PSTN or cellular gateway, which could authenticate the call 

leg on the IP address in the system. However, some ITSPs that do not have a session boarder controller and 

require a public IP address use gateway trunks. A gateway is defined as a device that converts media from one 

network or protocol to another. This type of trunk is used for a media gateway. 

SIP gateways are used to transport public switched telephone network (PSTN) terminated signaling across 

an Internet protocol (IP) network. The gateway sits between the PSTN and the IP network (Figure 4-4). 

Unlike the SIP registrations trunk, the gateway model does not register—It just sends the traffic to the destination. 

In this model, the system uses the caller-ID of the system to indicate the extension that initiated the 

outgoing call (if that extension did not block the caller-ID). This model is typically used with PSTN gateway 

hardware located on customer premises, but it can also be used to link two snom ONE systems together as 

long as they are routable to each other, i.e., both are on public IP addresses or on the same private network. 

PSTN 

Media 

Gateway 

Figure 4-4. Example of SIP Gateway Trunk 

Outbound Proxy 

SIP Gateway Trunk 

The outbound proxy trunk can be used to communicate with any other type of SIP proxy or IP telephone 

system or to join two IP telephone system deployments together. The outbound proxy trunk is a direct con- 


70 


nection to the network and is similar to the gateway model. The difference is in the way anonymous calls are 

made and in how the proxy represents its own domain. As the name suggests, the proxy model assumes that 

you are talking to a SIP proxy or session border controller, while the gateway model assumes you are talking 

to a SIP user agent. However, the two models are quite similar. As a general rule, use the gateway trunk first, 

and if you have an issue, try changing the trunk type. 

PSTN 

SBC 

SIP Gateway Trunk 

Figure 4-5. Example of Outbound Proxy Trunk 

Inbounds Calls 

How the System Identifies a Trunk 

When a new call comes into the system, the system must determine whether the call is an internal extension 

or an outside call that is configured to a trunk. To do this, it first looks at the From field to see if it matches 

an extension. If it’s not an extension, then from a registration trunk, it looks for a line parameter in the SIP 

INVITE (see Appendix B for information on the SIP INVITE). The line parameter is set by the system 

when the trunk is registered (Figure 4-6). Although it is not a predefined SIP parameter, the line parameter 

is used by many SIP implementations to identify where the call is routed to. 

Figure 4-6. Line Parameter in REGISTER Request of SIP Trace 

Figure 4-7. Line Parameter in 200 Ok of SIP Trace 

Figure 4-8. Line Parameter in INVITE of SIP Trace 



For gateways and proxies, this method is not possible, so if the line parameter isn’t there to give clues on 

how to route the call, the system must continue searching the trunk. This is done by a recursive DNS-resolution 

of the outbound proxy of the trunk. As the system searches, it uses the priority shown in Figure 4-9. 

Note: The domain name localhost matches any domain name presented in the Request-URI. (The 

Request-URI is a Uniform Resource Identifier that lists the user or service to which the request is being 

addressed.) Also, if the From header identifies an extension on the system, the trunk identification will be 

cancelled and the system will assume that the call is from that extension, regardless of whether the extension 

is registered on the perceived IP address or not, and the system will try to authenticate the extension. The 

system determines which IP addresses and ports a trunk may send requests to. 

A match is based on 

the following priority: 

Proxy address of trunk 

DID number in domain of trunk 


Account name of trunk 

Domain name of trunk 


Username of trunk 


Port of proxy address of trunk 





Port of proxy address of trunk 



Figure 4-9. Identifying Gateway and Proxy Trunks 

Trunk Settings 

Extension Settings 

75.150.87.9:5060 


DID 

Port

72 


How the System Routes a Call to the Proper Extension 

Basic Routing 

Once the trunk has been identified and the inbound call is trusted, the system must determine where in the 

system to send the call. Two possibilities are available for this: 

1. The system first looks at the trunk’s Send call to extension setting. If the setting includes an extension, 

the call will be sent to that account. If the extension does not exist, the call will fail. 

If the setting includes regular expressions, the system will route the call accordingly (see “When 

Regular Expressions are Involved” on page 73). When using regular expressions in the Send call 

to extension field, it is important to provide a default extension in the event the replacement pattern 

does not produce an existing extension (in the example shown below, 777 is the fallback extension). 

2. If nothing was set in the trunk’s Send call to extension setting, the system must go to the individual 

extensions to determine whether the called number is associated with any account. If the number 

dialed was 8124351234, the system will see that extension 508 is associated with the incoming 

number and will route the call accordingly. If no extension is associated with the incoming number, 

the call will fail. 

The number shown to the right of the extension is known as a DID. This method of attaching a 

DID to an extension is a viable routing method only when the number of DIDs is small (e.g., four 

to ten DIDs). When a large number of DIDs are involved, using regular expressions in the trunk’s 

Send call to extension field is a better approach (page 73). 

Corresponding Log File and SIP Trace 

This section provides a few SIP traces that correspond to the settings shown in the previous section (“Basic 

Routing”). When a call has been routed to an extension through the Send call to extension setting, the 

system writes a log (Figure 4-10) with the message Trunk sends call to ... into the log 

file (log level 5). 

Figure 4-10. Log File Reflecting "Call to extension" Setting 

If the trunk’s Send call to extension field is empty and the system finds an extension that is associated 

with the number called, the called number will be reflected in the user portion of the Request-URI is 

8124351234 (Figure 4-11): 


Figure 4-11. SIP INVITE Reflecting DID Routing 

When Regular Expressions are Involved 


Regular expressions can be used for numerous call routing purposes, but they are especially helpful when you 

have a large number of DIDs. When you have a few DIDs, you can simply enter the DID number into the 

Account Number(s) setting in the extension that the DID is assigned to. Assigning a large number of DIDs 

to individual extension accounts, however, is cumbersome and could be expedited using regular expressions 

(direction inward dialing is discussed on page 75). 

A regular expression is a special character string that describes a search pattern that the trunk will use when 

looking for a match for an incoming number. The expression consists of the following key elements: 

■ Pattern: The pattern portion of the string defines the numbers that the system will use when 

searching for a match and how many. In the example shown in Figure 4-12 (expression 1), the 

system will consider all numbers (0 through 9) when identifying a match and will include only the 

first seven digits. In expression 2, the system will still consider all numbers but will include only the 

last three digits. 

■ Replacement: The replacement field tells the system which expression to use when dialing. In our 

example, expression 2 has been defined (as is indicated by \2), so if 9781234567 had been the 

incoming number, 567 would be dialed. 

■ The Request-URI (To header flag): This flag tells the system where to look. By default, the system 

will look at the Request-URI (“u”). However, if you want the system to look at the To header, use 

a t as shown in Figure 4-14 (some Internet service providers provide the destination information in 

the To header). SIP recommends the Request-URI. 

■ Delimiters (field separators): Each part must be separated by a delimiter (a unique character which 

is not used elsewhere in the setting string). In the example shown in Figure 4-12, an exclamation 

mark has been used. 

Note: Expressions are written without any spaces: ![0-9]{7}([0-9]{3}$)!\1! 

Delimiter 

! 

Numbers 

accepted 

Expression 1 

Number of 

digits 

([0-9]{7}) 

PATTERN 

Numbers 

accepted 

Expression 2 

Number of 

digits 

([0-9]{3}$) 

Figure 4-12. Regular Expressions String—Multiple Expressions 

Delimiter 

REPLACEMENT 

Expression 

used 

Delimiter 

! ! 

\2 


74 


Important: The example shown in Figure 4-12 simply shows that numerous expressions can be used in one 

string; however, a single expression in many cases would be sufficient. For example, to specify that only the 

last three digits of a number should be used for dialing, you could use !([0-9]{3}$)!\1 as shown in 

Figure 4-13: 

Delimiter 

! 

Numbers 

accepted 

PATTERN REPLACEMENT 

Expression 1 

Number of 

digits 

([0-9]{3}$) 

Delimiter 

! 

Expression 

used 

\1 

Figure 4-13. Regular Expressions String—a Single Expression 

Delimiter 

! 

This expression allows the trunk to find the correct extension when lots of DIDs are used on the system. As 

a safety net, you can also direct the trunk to send the call to a default extension in the event the replacement 

pattern does not produce an existing extension. The string shown in Figure 4-14 includes a default extension 

number (777), so if the trunk is unable to find an extension using the replacement field, extension 777 will 

be called. 

Delimiter 

! 

Numbers 

accepted 

PATTERN REPLACEMENT 

Expression 1 

Number of 

digits 

([0-9]{3}$) 

Delimiter 

! 

Expression 

used 

\1 

Delimiter 

! 

R. 

URI 

Figure 4-14. Regular Expressions String—Default Extension 

Regular Expression Syntax 

The syntax used in expressions is explained in Table 4-1. 

Table 4-1. Syntax Used in Expressions 

Syntax Function 

! 

[ ] 

{ } 

( ) 

\2 

* 

$ 

Delimiter (separates each part) 

Lists the digits that are accepted 

Defines the number of digits 

Stores the numbers that match 

Indicates which expression to call 

Wild card (any length and character) 

Tells the system to read from end of string 

Samples of regular expressions are shown in Table 4-2. 

t 

Delimiter 

! 

Default 

to 

777 


Delimiter 

!

Table 4-2. Examples of Regular Expressions 

Example 1 ![0-9]{7}([0-9]*)!\1!t!100 


This example is common in Europe. It is used to strip the main number of the system so 

that the remaining numbers can be used to identify the extension. If the extension is not 

found, the call will be sent to the auto attendant. This example assumes that the number 

starts with 7 digits (e.g., 0228123) and that the auto attendant is located on extension 100. 

Example 2 !([0-9]{4}$)!\1!t!100 

This example always uses the last 4 digits of the number, regardless of how long the number 

is. This example assumes that the number of digits is always the same. 

Example 3 100 

This example is common in U.S. offices, where it is typical to send all calls to an auto attendant. 

This example assumes that the auto attendant is located on account 100. 

Example 4 !1([0-9]*)!\1!u!100 

This example can be used to strip the first digit from a DID number (default destination 

would be 100). 

Regular Expressions and Direct Inward Dialing (DID) 

Direct Inward Dialing (DID) allows users on your system to have their own (or even multiple) local numbers 

assigned to them. With DIDs, anyone outside the company can call the user’s unique phone number 

and reach the user on the same phone as those who dial the user’s extension number internally. Before you 

can use DID numbers, you will need to purchase them from a DID service provider, then link the DID 

numbers to the extension accounts. The easiest way of doing this is by entering the DID number into the 

Account Number(s) setting of the extension that will be using the DID. 

While this method is easy to use, it is too cumbersome to employ for a large number of DIDs. You will need 

to use an alternate method, depending upon the numbering scheme of the DID numbers themselves. When 

the ending digits of the DIDs form a definite pattern that can lend itself to correlate with extension numbers, 

you simply need to enter a single string into the trunk’s Send call to extension field. The only requirement 

here is that the ending digits must begin with a 4 or higher (extension numbers should stay within the 

range from 4xx through 7xx to keep from overlapping with Direct Destinations numbers). 

If the last 3 or 4 digits of a block of DIDs begin with a 1, 2, or 3 (e.g., 100, 101, etc.), the ending digits cannot 

be used as the extension numbers. You will need to use the workaround shown in Option 2. 

Option 1: Clear pattern with a start number of 4 or higher 

If you have a batch of DID numbers with a usable pattern on the ending digits, you can easily use regular 

expressions in the Send call to extension setting. The system will replace the 10-digit DID with the extension 

number. Table 4-3 displays a list of DIDs that can be easily converted to 3- and 4-digit extensions 

using regular expressions. 

Table 4-3. Direct Inward Dialing Numbers with Usable Patterns 

DIDs and 3-Digit Extensions DIDs and 4-Digit Extensions 

978 521-1400 978 521-4100 

978 521-1401 978 521-4101 


76 


DIDs and 3-Digit Extensions DIDs and 4-Digit Extensions 

978 521-1402 978 521-4102 

978 521-1403 978 521-4103 

978 521-1404 978 521-4104 

978 521-1405 978 521-4105 

Follow the steps below: 

1. Create the regular expression. 

2. Place the string into the trunk’s Send call to extension setting. 

Using the example shown below, the system will use either the last 3 or 4 digits of the 10-digit number, 

depending upon which string is used. (If the system cannot find a matching DID, the call will 

be sent to extension 777.) 

Option 2: Defined pattern with a start number lower than 4 

When DID ending patterns are in the 100, 200, or 300 range, you should not create extensions based on 

these numbers (extensions should begin with 4 or higher). As a workaround, you will need to create aliases 

for the extensions, and to save time, this should be done when you create the extensions. Once this is complete, 

a regular expression must be entered into the Send call to extension field so that the system will 

replace the 10-digit DID with the alias number: 

1. Decide which alias numbers to use. 

DID Alias 

978 521-1100 100 

978 521-1101 101 

978 521-1102 102 

978 521-1103 103 

978 521-1104 104 

978 521-1105 105 


2. Enter the aliases into the corresponding extension accounts: 


3. Enter a regular expression into the Send call to extension setting, telling the system to replace the 

10-digit DID with the alias number that is associated with the extension account. 

Outbound Calls 

Caller-ID 

The caller-ID is shown in the called party’s telephone display and is also used by the carrier for billing purposes. 

This is typically the From field of the SIP packet. When the system sends a call to a trunk, it presents 

the source of the call, usually the caller-ID of the calling extension. However, in cases of redirected calls, it 

can get more complicated. Here, the original caller-ID should be in the display of the called party, while for 

billing purposes, a local identity on the system must be used. 

Also, in many cases, it is necessary to translate a local number (e.g., sip:123@localhost) into a global 

telephone number (e.g., sip:9781234567@itsp.com;user=phone). The user=phone parameter 

indicates that the domain name should be ignored when presenting this number. This number is also known 

as the ANI (automatic number identification). 

Generating the ANI 

Before the system can generate an outbound call on a trunk, it must have an ANI. An ANI is an administrative 

number provided by the system to identify the caller to the receiver and is not necessarily the 

originating number. To get the ANI, the system first checks the SIP URI in the To/From header and the 

identity headers (P-Preferred-Identity, P-Asserted-Identity or Remote-Party-ID) for the 

user=phone flag. When this flag is set, those numbers will be used as the ANI. 

Figure 4-15. ANI and user=phone Flag 


78 


However, if the SIP URI does not contain the user=phone flag, the system looks for the ANI from the following 

places (in the order listed here): 

1. Extension account (ANI setting): If the local account has an ANI number, the system uses this 

number. Every extension or any other account that has a dial plan also has an ANI number field. By 

entering this information, it is clear how this account will be presented to the outside world. 

2. Domain (Default ANI setting): 

3. Trunk (Prefix setting): If the trunk’s ANI has not been set, the system will check to see whether a 

prefix has been set in the trunk. When this is the case, the system will add the fixed number to the 

front of all calls leaving the trunk. In many cases when you have an ITSP, you may want a prefix 

number to be appended to the front of all calls. The prefix could be an account number, a least-cost 

routing number, or a number to indicate an outside line on another system (e.g., 9 as shown below). 

4. Trunk (Trunk ANI setting). 

As a last resort, the system uses the DID number of the trunk, when there is one. This is a typical 

scenario in the NANPA area, where a trunk has a primary number associated with it. When someone 

calls the number, the caller will typically reach an auto attendant that processes the DID number. 

Representing the Source 

The system differentiates between the following two numbers when presenting a number on the trunk. 

■ Display number: This number is the number that the user should see on the display of the phone. 

In the case of a redirected call, this number would be the original caller-ID that the system sees on 

the incoming call. 

■ Network number: This number is the number that the provider wants to see for billing purposes. 

Over the last few years, providers have developed different methods of representing these two numbers: 

■ RFC 3325 describes a way to represent these two numbers. In most cases, it makes sense to use the 

P-Asserted-Identity header. In this case, the From header in the INVITE represents the 

display number, while the P-Asserted-Identity header has the network number. A similar 



representation can be done with the P-Preferred-Identity header. The system changes only 

the name of the header from P-Asserted-Identity to P-Preferred-Identity. The rest 

remains the same, as in the first method. 

■ No Indication: This method just discards the display number and uses only the network number 

in the From header. This method is a fallback when the provider cannot handle any other method. 

The disadvantage here is clearly that any redirection information gets lost. 

■ Remote-Party-ID: This method is described in a draft that expired years ago; however, there is still 

a lot of equipment outside that is supporting this method. In this case, the From header in the 

INVITE represents the network number, while the P-Asserted-Identity header is the display 

number. 

■ RFC 3325, but don’t hide: This method should not be used. It has been used in environments 

where the fields have gotten mixed up, and it is creating even more confusion. 

■ To strip the first digit from a DID number, you can use the pattern !1([0-9]*)!\1!u!100 

(default destination would be 100). 

Creating Trunks 

To view trunks, click on the trunks tab. This page shows you the trunks that are available and the status of 

each trunk. 

1. From the domain where the trunk will be used, click the Trunks tab. 

2. Enter a name for the trunk. You can name the trunk anything you prefer (e.g., company name, 

department, etc.). The name must consist of alphanumeric characters and may contain spaces. 

3. From the dropdown list, choose a trunk type. (For a description on each trunk type, see “Trunk 

Types” on page 68.) 

4. Click Create. 


80 


Configuring Trunks 

General Settings 

Trunks have a large number of settings. For an explanation of each setting, match the numbers in the image 

with the numbers below the image. 

1. Name: Once a trunk has been created, the name and type of the trunk can be changed at any time. 

2. Type: See “Trunk Types” on page 68. 

3. Direction: Traffic can be limited to either inbound traffic or outbound traffic. 

■ Inbound and outbound: Typical setting. 

■ Inbound only: Limiting traffic to inbound traffic prevents users from using the trunk for unauthorized 

calls. 

■ Outbound only: Enable this setting if you are using a trunk for outbound traffic only. It makes 

it easier for the system, as the trunk will not try to match inbound traffic to this trunk. 

4. Trunk Destination: This setting is useful when connecting to an Exchange server. Depending upon 

the setting, the SIP headers will be altered. If NBE is selected, additional options are made available 

(2, 4, 8, or 12 ports). 



5. Display Name: The display name is used for display purposes in the trunk listings page. This setting 

is the friendly name of the SIP From field and can be changed at any time. 

6. Account: The account is typically the main DID on the trunk. 

7. Domain: The domain is typically the upstream registration server or SIP server. It can be an IP address 

or a domain name. 

Example: sip.itsp.com 

If the service provider is using DNS SRV records, it could be itsp.com. 

8. Username: The username and password are used for authentication purposes (not all registrars require 

a different username for authentication). 

9. Password: The password needs to match the password from the trunk provider. 

10. Password (repeat): Confirmation of the password. 

11. Proxy Address: This setting defines where requests made of this trunk will be sent. When this field 

is set, requests will be sent to the specified address. Otherwise, the dial plan replacement field will be 

used to route the request. 

Example: sip.messagenet.it:5061 

The outbound proxy field follows the definitions of RFC 3263 (“Locating SIP Servers”). You may 

use the fully qualified domain name (FQDN) for a SIP server. If you add a colon with the port number 

after the FQDN DNS, a resolution will be used; otherwise, the system will try DNS NAPTR 

and DNS SRV first. You can force the SIP connection to use TCP by supplying an outbound proxy. 

Example: sip:hostname:5060;transport=tcp 

Important: The outbound proxy is an important setting. If an outbound proxy is not used, then the 

system will assume that SIP requests on this trunk can come from any location. This will make it difficult 

for the system to match incoming SIP requests. Unless you want to receive requests from any 

location in the Internet, you should specify an outbound proxy. 

12. CO (central office) Lines: CO-lines can serve multiple purposes on the system. They can be used 

in a way similar to the CO-lines of traditional TDM-based PBX systems. Although snom ONE runs 

calls through SIP trunks over computer networks instead of connecting to public telephone lines (at 

least directly), CO-lines are still useful for determining which calls are active between the system and 

the outside world. It is a way to channelize a trunk. 

Each trunk may have several CO-lines. Because users can subscribe to the state of CO-lines, each 

CO-line name must be unique within a domain. For example, four CO-lines might be set up on 

trunk 1 with the names co1 co2 co3 co4, while trunk 2 might contain co5 co6 co7 co8 

(the list must be delimited by spaces). The system will reject names for a CO-line that are already 

in use by accounts or other CO-lines in the same domain. The CO-lines that you establish can be 

monitored from the domain’s Account list. If a CO-line is in use, it will display the extension and the 

outside phone number to which it is connected. 

Limiting inbound and outbound traffic: CO-lines can be used to limit the number of calls that can 

be assigned to a trunk. When the CO-line setting is used, the system will reserve one line for each 

call. When all lines are in use, the system will reject further calls that attempt to use the CO-line. For 

example, if you have a SIP trunk that can handle only three simultaneous calls, you should assign 

three CO-lines (e.g., co1 co2 co3). If a fourth simultaneous call attempt is made, the system will 

play a fast busy signal. If you have more than one trunk, you can set up a failover behavior so that 

the system will try to use a different trunk for the call once the first trunk has all CO-lines in use. 


82 


Reserving lines exclusively for inbound or outbound traffic may be appropriate at times. If you put a 

:i after the line name, the system will use that line only for inbound traffic (e.g., line1:i). If you 

put a :o after the line name, the system will use that line only for outbound traffic. If there is no attribute 

set after the colon, the line will be available for both inbound and outbound traffic. 

Monitoring CO-lines: CO-lines are required for many basic functions. In most small offices, transfers 

are done by pressing the Hold key and lighting the lamps on the other phones. In order to do this, 

the phone must be able to display which call is on which CO-line. 

Phone 

Buttons 

Button 7 

Button 8 

Button 9 

Button 10 

Figure 4-16. CO-Lines Linked to Phone Buttons 

CO lines 

From a system point of view, the status of a CO-line is similar to the status of an extension: It may be idle, 

connected, ringing, on hold, or it might be holding a parked call. Therefore, the mechanism that shows the 

status of a CO-line is similar to the mechanism that shows the status of an extension. This implies that the 

name of the CO-line must not clash with the name of an extension or any other account on the system. 

(In order to see the CO-line status, the user agent needs to subscribe to the status of the CO-line. See the 

description of the phones on how to do this.) 

13. Permissions to monitor this account: Enter the extensions that are permitted to monitor this account. 

14. Override codec preference: Use this setting to specify particular codec preferences for the trunk. By 

default, the system uses the codec preference of the system, which has the advantage of allowing the 

system to negotiate codecs in such a way that transcoding can be avoided. (When default mode is 

being used, no codecs will be displayed on the left side of the codec selection box.) 

If it is necessary to enforce specific codecs on a trunk, choose a codec from the list of available codecs 

and click the Add button, then repeat as needed. Be sure to position the codec with the highest preference 

first. To delete a codec, click the Remove button. To move a codec up or down in the preference 

list, use the Up and Down buttons. 

15. Lock codec during conversation: This prevents the end point from engaging a conversation on one 

codec and then requesting a codec change later. For example, if a carrier wants the G.729 codec to be 

used, it can prevent someone from using G.711, for instance. 


CO1 

CO2 

CO3 

CO4


16. Proposed Duration (s): Use this setting to specify the registration duration. Although the registrar 

typically defines the duration of the registration (according to the IETF standards), some providers 

follow the proposed duration of the trunk. The values range from 20 to 3600 seconds. One hour 

(3600 seconds) is a reasonable value. 

Because some providers will accept the proposed duration without taking care to refresh the NAT 

binding, it is necessary to force the system to renew the registration in a shorter period of time 

(through the Keepalive Time setting below) and ignore the service provider’s registration. 

17. Keepalive Time: Keepalive time is the time required to keep the registration live before it reregisters. 

This setting helps with trunks which connect to ITSPs that do have a good session border 

controller. This setting may cause a lot of traffic to your service provider, so be sure not to enter an 

unnecessarily short time. Typically, this setting can be left blank. 

18. Send email on status change: Once a trunk has been registered, it is critical that status changes 

are brought to the attention of the appropriate individual. For example, if the service provider goes 

off line and the trunk looses the registration, the system will send an email notification to the email 

account that is specified. 

19. Strict RTP Routing: This setting is needed because the IETF allows “RTP traffic send ports” to be 

different from “RTP receiving ports,” creating an extremely NAT-unfriendly situation. While most 

implementations today use the same port number for sending and receiving RTP, some gateways still 

insist on strict IETF compatibility. In such a case, this setting should be enabled. We recommend 

you keep this flag disabled (“No”) unless asymmetrical ports are required. 

20. Avoid RFC 4122 (UUID): Not all registrars ignore RFC 4122 (UUID) when it does not support 

it, so it is necessary to explicitly suppress the UUID in register requests. SIP implementations that 

are not able to deal with UUID report a “Bad Request” to indicate that they were not able to process 

the request. The UUID is used to indicate that a registration replaces another registration, which is 

useful for avoiding double registration after a system restart. 


84 


21. Generate unique extension number: This setting is used for emergency calls, so that the 911 call 

center can directly call the extension without having to go through the auto attendant. When this 

setting is set on the trunk, the system will use the EPID of the extension to represent where the call 

is coming from (if it does not exist yet, the system will allocate a new EPID). 

22. Accept Redirect: Use this setting if your trunk should respect redirect codes. By default, this introduces 

significant security risks because the system cannot determine if these redirects introduce 

additional costs (redirection to expensive numbers). Therefore, enable this setting only if you are sure 

that your service provider or SIP gateway does not abuse this feature. This flag is especially important 

when you use the system with Microsoft Exchange or other Microsoft products, such as the speech 

server. Enable this setting also when you have a trunk that comes from another system; this will 

make it possible to call from one system to another. 

23. Interpret SIP URI always as telephone number: When a call comes into the system, the system 

needs to know how to interpret the number. In SIP, the URI contains a domain name; however, in 

most cases, the domain name should be ignored when interpreting the URI coming from this trunk 

(because the user portion is just a telephone number). Usually, this is indicated by the parameter 

“user=phone,” but there are some service providers that do not set this flag. By turning on the “Interpret 

SIP URI always as telephone number,” you make the system believe that this flag was set on the 

trunk call. 

24. Requires busy tone detection: When an analog PSTN gateway (e.g., FXO) is being used, hangup 

detection can be an issue. In FXO, the hangup signal might be just a tone that needs to be detected. 

Unfortunately, there is no international standard for the disconnect tone. Incoming international 

calls might give you a disconnect tone that the system has to recognize. Of course, if the PSTN gateway 

is capable of detecting this, the task should be left to the PSTN gateway. However, as a fallback, 

you may also configure the system to perform the hangup detection. The disadvantage of doing this 

is that it costs additional CPU resources and it might randomly disconnect calls, for example, if the 

other party is playing back a tone that sounds like a busy tone. The best way to avoid these kinds of 

problems is to use a digital line, e.g., a SIP trunk. 

25. Trunk requires out-of-band DTMF tones: When this setting is enabled, the trunk will use 

RFC 2833. 

Outbound Settings 

26. Prefix: This setting is used as a representation of the system when making outbound calls. The 

system will add the prefix to the front of all calls leaving the trunk. (See also “Generating the ANI” 

on page 77.) 

27. Global: This setting is used only in multi-domain environments. When it is enabled, calls that come 

in on this trunk are permitted to jump into other domains if there is a match on a global alias name. 



(If the direction is not only inbound, then other domains may use this trunk in their dial plans for 

outbound calls.) When this setting is disabled, the system does not search for tel:-alias names. 

28. Trunk ANI: You can configure each trunk with an ANI (Automatic Number Identification). ANI 

is a service that tells the recipient of a telephone call the telephone number of the person making the 

call. In most cases, the ANI is used in the From field in the SIP packets or the caller ID. 

29. Remote Party/Privacy Indication: This setting tells the system how to present the caller-ID on the 

trunk (see “Caller-ID” on page 77). 

30. Rewrite global numbers: When you are using a trunk, you might have to represent the telephone 

number in a specific format. For example, in the NANPA area, you might want to use 10 or 11 digits 

to represent a national number. 

If you are using several trunks, you can represent the same number in different styles depending on 

the trunk. The Rewrite global numbers setting corresponds to the Country Code setting (see page 

53): 

31. Failover Behavior: When the trunk receives an error code, it might send the call back to the dial 

plan and continue the matching process. The system continues the dial plan with the next higher 

priority, ignoring entries with lower or same priority. This is useful when the trunk is just a “trial” to 

place the call, for example, when several PSTN gateways are available for terminating the call and 

one gateway does not accept any more calls. Another example is when you first try to route the call 

via a peer-to-peer call using ENUM or other location methods and only if such resolution does not 

result in a connection fallback to a PSTN call. The setting allows four behaviors: 

— No failover: This is the default behavior. In this case, the caller will receive the error code as a 

result of the attempted call. 

— On 5xx and 408 error codes: This will trigger failover only when a 5xx or 408 class error 

code is received. PSTN gateways typically return 5xx class error codes when all channels are 

in use; however, this mode will allow you to switch to the next PSTN gateway when a line is 

busy and will not trigger the failover. 

— On all error codes: In this case, all error codes will trigger the failover process. Note that 

call redirect will also be treated as an error code, as the redirection destination can easily be 

abused to route calls through expensive routes. 

— Always, except for busy response: Even if the remote party is busy, the system will try alternate 

routes. 

If you are using failover, you have the option to specify a request timeout value for the trunk. By 

default, this is 32 seconds (the SIP default request timeout). However, in many cases, it makes 


86 


sense to specify a shorter value. After the request timeout, the system will internally generate a “408 

Request Timeout” response code and process it according to the failover rules. 

32. Is Secure: This setting is used to indicate that outbound calls on the trunk can be treated as secure 

calls. For example, when the trunk goes to a local PSTN gateway, you might decide to treat this call 

as a secure call. Incoming calls with the SIPS scheme will ask the snom ONE system to ensure that 

the call be kept secure end-to-end. 

33. ICID (RFC 3455): This setting is used in the IMS environment and is sent in the P-Charging- 

Vector header of the SIP packets on the trunk. It is essentially a token that identifies the trunk. If 

your provider uses this header, it should be placed into this field. 

Inbound Settings 

34. Explicitly list addresses for inbound traffic: This setting allows you to explicitly state trusted 

inbound proxy addresses and enables you to identify the origin of the request. It is ideal for VPN environments 

or when routing calls between offices, as it allows you to create a single trunk and specify 

the IP addresses of the other locations. The addresses can be either IP addresses (with an optional 

port number behind) or a DNS A or AAAA address. 

35. Send call to extension: This setting is discussed in “Inbound Calls” on 

page 70. 

36. Assume that call comes from user: This setting is used for trunks that accept redirects. The settings 

must be an extension in the domain of the trunk. This setting is necessary in order to determine 

what dial plan to use, and it is also necessary to charge a user on the system for the call. For regular 

trunks, you should leave this field empty. 

37. Ringback: This feature was introduced to deal with network operators who were not able to deal 

with early media. Although using the 180 Message simplifies the signaling in forking calls scenarios, 

it creates additional delay when the called party picks the handset up and the first samples on the 

conversion may not be transported. We strongly recommend leaving the flag set to the Media state 

(which is default) and asking the operator to fix the problems with early media. 

Connecting Branch Offices Together 

This section describes how to connect two snom ONE systems together so that calls between the two locations 

will be routed through VoIP rather than through the local PSTN. Two different scenarios are presented: 

■ Two snom ONE telephone systems with public routable IP addresses (this model is also suitable for 

two systems on a private network with no NAT issues) 

■ Two snom ONE telephone systems: A head office site and a remote site 


Using a Gateway Trunk 


SIP gateway trunks are the best method for connecting two snom ONE telephone systems, as long as both 

systems either have public routable IP addresses or are on a private network with no issues with NAT. Likely 

candidates for this configuration are the classroom environment and organizations with a large private network 

(as long as the system has an interface directly on a public IP address). Figure 4-17 shows two remote 

sites connected by two gateways. Callers from either system can dial the other system using four-digit dialing. 

5xx 

Extensions 

Site 1 

209.251.11.111 

Public IPs 

6xx 

Extensions 

Figure 4-17. Connecting Branch Offices Using a Gateway Trunk 

Site 2 

151.195.11.111 

Create a trunk for each branch office, using the instructions that are shown below. 

Branch Office 1 

1. From the Type dropdown, select SIP Gateway. 

2. Enter the IP address into the Domain field: 

Branch Office 2 

1. From the Type dropdown, select SIP Gateway. 

2. Enter the IP address into the Domain and Outbound Proxy fields: 


88 


Using a SIP Registrations Trunk (SIP Tie Line) 

The SIP registration trunk can be used to connect two branch offices when one of the offices has a private 

IP address. For example, a corporate head office has a pubic IP address, and a remote branch office has only 

one public IP address and requires that the system be on a private address but still wants to have connectivity 

back to corporate (Figure 4-18). 

Site 1 

192.168.1.x 

5xx 

Extensions 

Private IP 

Public IP 

6xx 

Extensions 

Site 2 

151.196.11.111 

Figure 4-18. Connecting Branch Offices Using a SIP Tie Line 

1. Create an extension that will be used by the trunk to route the calls: 

2. Create a SIP registrations trunk for the branch office, entering the extension that will be used to 

route the calls (e.g., 525). 



Once the offices have been connected, the headquarters will need to dial the extension that the trunk is 

routed to. Once the call hits the branch office, it will be sent to the auto attendant so that the headquarters 

can dial the remote extensions. The trunk has to register to the extension just as a phone would, so the 

system at the branch office will need to register the trunk to the system at the headquarters, the same way an 

ITSP would need to be registered. Then anyone in the world can dial the DID, much like they would dial 

that extension. 


90 



DIAL PLANS 

Chapter 

5 

When an extension dials a number that is not available on the local snom ONE telephone system, the system 

must search for a trunk on which to route the call. To do this, it consults the dial plan. The job of the 

dial plan is to use pattern and replacement strings to route calls out different trunks. Since extensions map 

to dial plans, dial plans should be set up before extensions; however, if only one dial plan will be used and it 

will be the default dial plan, then this is not as important. 

Dial Plan 

Trunk 

Trunk 

Multiple dial plans can exist on the same domain, but an extension can be mapped to only one dial plan. 

Administrators can assign the same dial plan to multiple extensions or create multiple dial plans and assign 

them to different users. A “restrictive” dial plan might be used to handle only local calls for phones in common 

areas, and an “international” dial plan might be used for international calling by executives. 

Parts of a Dial Plan 

Key Components 

A dial plan consists of the following key components: 

■ Preference: The preference number is used to establish the preference order that will be used by 

the system when searching for a trunk on which to route a call. The system processes the available 

trunks beginning with the lowest preference value. (Identical preference values can be used for different 

trunks within the same dial plan; however, the system will randomly choose a trunk when 

more than one trunk meets the given criteria, so you should preference them.) 

■ Trunk: The trunk setting allows you to designate trunks that will be used by the dial plan. Trunks 

that are available for the domain are listed in the dropdown menu, and additional trunks can be 

added. Once trunks have been designated for a particular dial plan, the trunk that meets the pattern 

and replacement criteria for a particular call will be used to process the call. 

■ C and P: The C and P checkboxes determine how a call is handled once the user dials out onto 

the trunk. If the C checkbox is selected, the user will be prompted to enter a CMC code, but will 

not be required to. If the P checkbox is selected, the user will be required to enter a PIN number. 


92 


This can be used to prevent unauthorized callers from making international calls. The C and P setting 

works in tandem with the CMC authentication for outbound calls domain setting (see page 

56). The CMC code can also be set from the domain address book or the personal address book. 

■ Pattern: The pattern field is used to interpret digit sequences that will be used to decide on routing. 

In most cases, simplified expressions consisting of literals, prefixes, and fixed patterns will suffice. 

— Literals: If you want to match a specific number, just enter the exact number (e.g., 911). 

The literal will be the first match in the expression. 

— Prefixes: Prefixes allow you to route numbers that begin with identical prefixes through the 

same trunk. For example, by specifying the common prefix 9011 in the pattern field, all international 

numbers will be routed through the same trunk. To indicate that a prefix is being 

using, an asterisk must be entered after the prefix (e.g., 9011*). 

— Fixed patterns: If you use an x in a pattern, the system will treat it as a wildcard for 0-9. 

For example, 978xxxxxxx will route all calls beginning with the area code 978 through the 

same trunk. 

■ Replacement: The replacement string is used to convert digit sequences into a dial string for outbound 

calling. It is used in the To header as well as in the Request-URI. 

— Simple patterns such as literals, prefixes, and fixed patterns do not require a replacement 

string; the system will automatically use sip:\1@\r;user=phone as the replacement. 

If you use a prefix in front of a star, the system will insert that prefix before the match. For 

example, 1* will put a 1 in front of the match. 

Following are some sample patterns and replacement strings. 

Example 1 Pattern: 91* 

Replacement: 1* 

Input: 919781234567@domain.com 

Output: 19781234567@domain.com 

Example 2 Pattern: 978xxxxxxx 

Replacement 

Input: 9781234567@domain.com 


Example 3 Pattern: 911/411 

Replacement: 

Input: 911 


Example 4 Pattern: xxxxxxx 

Replacement: 234xxxxxxx 

Input: 1234567 



Chapter 5: Dial Plans 93 

Following is a sample dial plan. The trunks are listed in the order in which the system will look at each 

trunk. The notes that follow the dial plan describe the effect of the pattern and replacement fields. 

Notes 

A. Not Allowed: This entry prevents anyone from calling 900 numbers. 

B. Entries B and C are used to route calls that are forked to cell phone number 9783741234. 

Whenever someone dials this number manually or if the system forks the call to the cell phone (or 

even if *00511 is dialed), then the system will try to route the call out the GSM gateway, then to 

the GSM phone. This method will prevent a hit on minutes. 

C. See entry B above. 

D. Entries D, E, and F: These entries are used in tandem to ensure that a call goes through even if 

you lose your Internet connection. The idea is if a call is sent to ITSP1 and it fails, it will be sent 

through the Sangoma trunk and if that fails because your internet is down, then the call will be 

sent out the GSM gateway. 

Important: Failover behavior must be enabled on the trunk. 

E. See entry D above. 

F. See entry D above. 

G. Sangoma: This entry is for international dialing. When this trunk is used, the user will be required 

to enter a PIN. 

H. ITSP2: This entry is used for dialing 1888 calls. 

I. All remaining calls will use this trunk. 


94 

Wildcard Patterns 


Wildcard patterns are a simple way to match a pattern in the system. This simple pattern-matching scheme, 

though not as powerful as the extended regular expression (ERE) pattern matching, can be used to solve 

most dialing issues. Table 5-1 lists typical usages of wildcard patterns. 

Table 5-1. Wildcard Patterns and Usage 

Wildcard Definition 

* A star (*) matches any string of any length. Note: 9* matches anything starting with a 9 

but *9 matches anything ending with a 9. 

? A ? matches any character and only one. 

$ A $ matches a single number and only one. 

% A % matches any number (multiple digits or empty string). 

[ ] A range enclosed in brackets matches that range. Note: 11[02] matches the pattern 110 

and 112. 

\ A \ matches the character that follows. 

Wildcard patterns can also be used in other places. To grant permission to all accounts, use a star (*). To 

limit the accounts, use a pattern such as 4* to allow all extensions that start with a 4. To grant permission to 

only one extension, place a single extension in the field. 

■ In the Permissions to monitor this account field: This field is included in all account types (extensions, 

auto attendants, hunt groups, paging accounts, service flags, etc.). 

■ In the Accounts that may record a message field: This field is included in agent groups, IVR 

nodes, and auto attendant accounts and is used to grant permission to record a greeting. 

■ In the Source field in Paging accounts: Use this field to specify which extensions can send a page. 

■ In the Extensions that may change status field: This field is used to change the status of a service 

flag. 

Regular Expression Matching 

The regular expression matching algorithm is a very flexible algorithm that follows the NAPTR algorithm of 

RFC 2915. For an exact description, please refer to this document (www.ietf.org). 

The pattern string of the dial plan is surrounded by a ^ and a $ to ensure that the whole string is matched. 

The system uses the username and the hostname. The port number, parameters, and the scheme are not 

included for the comparison. 

If there is a match, the system will generate the resulting destination from the replacement string. The string 

may include references to matching groups in the pattern string. These matches are referred by the group 



number (starting with 1). Additionally, the matching string r may be used to insert the registrar name. For 

examples of algorithms, see “Sample Dial Plans” on page 97. 

Building a Dial Plan 

Creating a Dial Plan 

Every domain has its own dial plans, and any number of dial plans is allowed on a domain. The number of 

entries allowed on a dial plan is also unlimited. 

1. To create a dial plan, click the Dial Plans tab. 

2. Enter a name for the dial plan. The name can be any descriptive name. Spaces, numbers, and capital 

letters are permitted. 


The dial plan has been created. Proceed to the next section to configure it. If you want to delete a dial plan, 

click Delete (be careful, as all dial plan data will be lost). 

Configuring the Dial Plan 

Once the dial plan has been created, trunks can be added. (Trunks must be created before they can be added 

to a dial plan. For more information on trunks, see Chapter 4.) 

1. To add trunks to the dial plan, click Edit, as shown below: 


96 


The following screen is displayed: 

2. Set the Global setting. To make this dial plan accessible to all domains on the system, click yes. 

3. From the Trunk dropdown list, choose a trunk. (Depending on the number of trunks that have been 

created for this domain, the dropdown list will vary.) In addition to trunks, other choices are shown 

below: 

■ Not Allowed: This option is used to identify a pattern of numbers that should be denied by the 

system. When this mode is used, the system will stop processing the dial plan when that pattern 

is encountered. This makes it easier to define outgoing calls for which the called party is charged 

the cost of the calls by the telephone carrier, instead of the calling party (e.g., 900 numbers or 

expensive numbers like Afghanistan, which should never be called). 

■ Call Extension: This option is used to send a call to a registered extension. In this mode, the replacement 

field must contain an extension number. In most cases, you can just enter the extension 

(e.g., 123). But you can also use the rules for generating a replacement pattern, including 

the \1...\9 replacement strings. This is useful if you have a remote FXO or GSM gateway 

(which is behind a remote NAT device) that you want to route calls to. This allows you to take 

advantage of snom ONE and SBC. 

■ Try Loopback: This setting is not applicable to single-domain environments. In multi-domain 

environments, this mode is used to send a call from one domain to another domain (on the 

same system) without using an external SIP proxy. In this mode, the system will first check to 

see whether there is a match of the pattern and then calculate the destination using the replacement 

pattern. If the destination matches a telephone number (starting with “+”) on the local 

system, then the PBX will send the call back to the system. 

Note: Before you can use the Try Loopback mode, you must disable the Loopback detection setting 

(see page 30), and you must have at least one UDP socket (not bound to any specific address or 

bound to the loopback address 127.0.0.1 or [::1]). 

4. Enter a preference value for the trunk (default is 100). 

5. Enter a pattern that the system will use when interpreting digit sequences. 

6. Enter a replacement string that will be used to convert digit sequences into a dial string for outbound 

calling. 


8. Repeat Steps 2 through 7 for each trunk that needs to be added to the dial plan. 


Sample Dial Plans 

Typical Dial Plan 


A typical dial plan would include the string ([0-9]*)@.* as pattern and sip:\1@\r;user=phone as 

the replacement. The pattern string has one group [0-9]* (which is referred in the replacement string as 

\1). That means that if the pattern is matched against the value 2121234567@test.com, it will store 

2121234567 in the first group and the result will be sip:2121234567@test.com;user=phone (the 

user=phone indicates to the recipient that the number is a telephone number). 

A Simplified Dial Plan 

In many cases, you may just want to route all numbers to an outside trunk. This can be done easily by using 

the pattern *. You don’t need to fill anything into the replacement field (the system will do this automatically). 

Dial Plan with Prefix in Front of the Number 

If you use a pattern like 1* in the replacement field, the system will automatically put a 1 in front of the 

match that it found in the pattern field. In the case when you use the pattern 9* and dial the number 

92121234567, the system will automatically convert that into a 12121234567. 

A North American Dial Plan 

If you use the snom ONE telephone system in the fixed-length dial plan of North America, you may use a 

dial plan similar to the one that follows: 

■ 9[911|411]—The first pattern matches the emergency number and the service number explicitly 

and sends it to the local gateway. It is a good idea to have an entry for these important numbers so 

that they don’t accidentally get routed to the wrong gateway. 

■ 011*— The second pattern matches all international numbers and sends them to a special trunk, 

which is intended to save costs for international calls. 

■ 1*— The third pattern deals with all domestic calls. The fixed-length pattern was used so that 

the system can actually tell when this number is complete. You could add another pattern like 

91978xxxxxxx and send those calls to another trunk if you have negotiated a flat rate with your 

local PSTN service provider. 


98 


Sending Star Codes on a Trunk 

Extended regular expressions can also be used when sending star codes on a trunk. 

When passing a star (*) to the carrier, a backslash is required before the star (e.g., \*67); otherwise, the system 

will treat the star and the numbers that follow as a star code. In the sample shown above, a backslash has 

been placed before the Block Caller-ID (*67) star code, so when the trunk is used, the system will replace the 

\*67 with a *67, causing the caller-ID to be blocked. (The d in the latter part of the string tells the system 

to replace it with the domain. If an r had been used, the system would replace it with the registrar.) 

Important: When using star code patterns in dial plans, you must remove the code from the standard star 

codes and/or any customized codes. 

If you’d rather not use star codes in the pattern field but still want to have the dial plan activate a star code, 

you can use a workaround. You can create a “stand-in” star code that will be used to correlate with the 

desired star code. In the example shown below, users will need to enter 99 before entering the actual phone 

number. The system will replace it with the *67 star code. This method allows you to retain the *67 star 

code as part of the default list of star codes; however, users will need to be informed ahead of time so they 

that will know to enter 99 before entering the phone number. 

Forced Matching 

The following dial plan illustrates the use of the caret. It’s being used to force the system to begin matching 

from the start of user input. 


Inter-Domain Dialing 

Domains with Non-Overlapping Extensions 


If you have domains with non-overlapping extensions, use the following instructions to configure your interdomain 

dialing. (DIDs, by nature, are non-overlapping.) 

1. For each domain, create a dial plan entry with Try Loopback in the Trunk field. 

2. Enter * for both pattern and replacement. 

Note: Be sure to place the loopback dial plan entry at the top of the dial plan; however, 911, 411, 

611, etc., can be the topmost entries. 

Domains with Overlapping Extensions 

If the extensions between the different domains are overlapping, users will need to dial a prefix before the 

actual extension number is dialed. For this configuration, you will need to designate a prefix and provide 

proper pattern and replacement strings. 

Note: Prefixing simply makes extension numbers non-overlapping; however, prefix-based dialing can be used 

for both overlapping and non-overlapping extensions. 

1. From the first domain, create a gateway trunk that will be used to dial to the other domain(s). 

2. Create a dial plan entry that uses the gateway trunk just created. 

3. Choose a prefix. Be sure to choose a prefix that doesn’t conflict with your other extensions (i.e., if 

the domain has extensions beginning with 5, you must use a number other than 5). 

4. For the Pattern field, enter the prefix you’ve chosen followed by a string that corresponds with the 

number of digits your extensions have. The example shown below is useful for three-digit extensions 

and will require users to dial a 6 before dialing the extension number (e.g., if the user dials 6400, 

extension 400 will be called). 

5. For the Replacement field, enter sip:\1@\r;user=phone. (It is not necessary to specify the domain 

name here. The PBX will use whatever is configured on the trunk.) 

6. Repeat these steps from the other domain(s). 


100 



EXTENSIONS 

Chapter 

6 

Extensions are at the foundation of the IP telephone system and are essentially the phones that are registered 

to the system for making and receiving calls. Without extensions, the system would be of little use since 

there would be no one to call. 

This chapter shows you how to create and configure extensions. Extensions have a wide selection of settings, 

and many of them can be set by the user. This chapter addresses only the settings that are necessary to get 

the extension ready for the user. These settings include the initial settings that are set when the extension is 

created, the registration settings, and the permission settings. Registration settings allow the administrator 

to define the maximum number of lines and concurrent registrations for the extension, and permission settings 

are used to grant access to advanced system functionality, such as Call Barge and Listen In. A handful 

of administrator settings can also be set from the user account, and they are discussed here. For information 

on the extension settings that can be configured by the user, see Chapter 22. 

Figure 6-1. Extensions on the snom ONE Telephone System 

Preparing the User Account 

When preparing extensions for your users, you have a few options. You can use the default extensions that 

come with the system, or you can create new ones. This section guides you through both. 

Working with the Default Extensions 

The snom ONE telephone system comes with ten preconfigured, two-digit extensions. To modify these 

extensions, use the following procedure: 


102 


1. Click the Domains tab. 

2. Select the domain and click Accounts. The accounts list will be displayed as shown below: 

Figure 6-2. Default Accounts 

3. Replace the First name and Last name fields with the user’s name. 

4. If you want to switch from 2-digit extensions to 3- or 4-digit extensions, you can do that here (e.g., 

change 41 to 4100). Otherwise, go to the next step. 


Chapter 6: Extensions 103 

Note: After changing the number of digits of the extensions, the PnP Dialplan Scheme setting will 

need to be reset so that the phone will allow the user to enter the necessary digits before it begins 

processing the call (this setting is detailed on page 62). 

5. Set the general user settings detailed on page 104. 


7. Before you register the account, configure the administrator, registration, and permission settings. 

The descriptions for these settings begin on page 106. 

Importing Multiple Extensions (Bulk) 

When a large number of extensions are involved, CSV files will expedite the process (see Appendix A for 

instructions on creating CSV files). 

1. Reset the system to factory defaults by clicking Admin > Settings > Configuration. 

2. Click Factory Reset as shown below: 

This will clear the system of all default accounts and will allow you to start fresh. 

3. Create the extensions using CSV files as detailed in Appendix A. 

4. Once the accounts have been created, the administrator, registration, and permission settings need to 

be configured. The descriptions for these settings begin on page 106. 

Creating a Few New Extensions 

If you want to create new extensions and have only a few to create, you can create them manually. To create 

a few extensions, use to procedure below: 

1. From the Accounts tab, click Create. 


104 


2. Select Extensions from the dropdown. 

3. Populate the fields using the information shown below: 

■ Number: This field takes an extension number and/or alias, i.e., a DID number. DID numbers 

allow users to be reached directly and are useful for phones that do not support 2, 3, or 4-digit 

dialing. An unlimited number of DIDs can be directly tied to an extension (DIDs can also be 

used without an extension account). When entering DIDs as aliases to an extension number, use 

the following syntax: 

505/9781234567/9781234567 

Note: If you use a space instead of a slash (505 9781234567), you will create two separate accounts 

that will be unrelated to each other. 

Alias accounts can give the impression that your system has more accounts than it actually has and can 

clutter it up. If this occurs, you can hide aliased accounts using the Accounts Page Display settings (see 

page 58). 

To create multiple accounts at the same time, use a space between them: 

505 506 507 

Note: Extension numbers should range from 2 to 4 digits. The number of digits must correspond to the 

PnP Dialplan Scheme setting so that the phone will allow the user to enter the required number of 

digits before it begins processing the call (see page 62 for more information on this setting). 

■ First name: The First name and Last name fields will be displayed for all internal and external 

calls (international characters are permitted). This is the case when calls are made from that 

extension or when the system sends email to its users. 

■ Last name: User’s last name. 

■ SIP Password: This password influences the connection between a user's VoIP phone and the 

system and influences a user's ability to access other registered SIP devices. When this password 

is set, the system will challenge SIP user agents on requests. 

■ Web password: The user will need this password when logging into the web interface and will 

be required to use a password that is based on the Password Strength setting set by the administrator. 

■ PIN: The PIN is used in several areas of the system (e.g., when a user accesses the voicemail 

system from another extension or from an outside line, when calling into the system from the 

cell phone, when using the calling card feature, and when hot desking). When you create extensions, 

you can leave the PIN field empty and opt to have users set up their own PINs. Once a 

PIN has been set up, no one other than a user who registers with the extension credentials will 

be able to go to the mailbox. For optimum security, we recommend at least five-digit PINs (see 

Voicemail PIN Digits on page 57. 




5. To open the account, click List then click the account or use the search box at the top of the screen 

The following screen will be displayed: 

Figure 6-3. General User Settings (as seen by the admin) 

Once the account has been created, the administrator, registration, and permission settings will need 

to be configured. These settings are outlined in the next section. 


106 


Before Registering the Extension 

Administrator-Only Settings 

■ Dial plan: Dial plans determine which trunks are used for outbound calls. Dial plans that are 

available on the domain will show up in this dial plan dropdown (see Chapter 5 for information on 

dial plans). 

■ ANI: ANI (Automatic Number Identification) is a service that tells the recipient of a telephone 

call which telephone number made the call. In most cases, the ANI is used in the From field in 

the SIP packets or the caller-ID. Each domain can be configured with a default ANI (page 54) 

or you can specify a unique ANI here. 

■ ANI for emergency calls: The system will send this ANI when the emergency number has been 

dialed. 

■ Send daily CDR report to: Enter the extension number(s) that should receive the CDR report 

for this extension. Use semicolons between multiple email addresses, since email addresses may 

contain spaces (e.g., Fred Feuerstein ; Carl Clements ). For 

detailed information about CDRs, see Chapter 21. 

■ Show following ACD queues: This setting is directly related to the Queue Manager setting that 

is part of the agent group account (page 147). A queue manager is a user who monitors agent 

groups. 

■ Show Web Attendant Console on user page: The Web Attendant Console (WAC) allows users 

to monitor extensions, join conferences, and do many system-related tasks. 

Registration Settings 

The registration page is used to control the registrations for the account. You may register one or more devices 

for one extension. When an extension is called, the system will call all registered extensions in parallel. 

The first extension that picks the call up will get the call, and the other extensions will receive a CANCEL 

message. 

Figure 6-4. Registration Settings 



Use the following information when configuring the resistration settings shown in Figure 6-4: 

■ Trusted IP Addresses: This setting dictates which IP address or addresses can be registered to this 

extension. The system will accept registrations only from the specified IP addresses. For example, if 

the extension has been set to 192.168.1.2 and 192.168.1.5 tries to register, it will be denied 

with a 403 forbidden message. Use a space as a separator when entering multiple IP addresses. To 

indicate subnets, use a slash and the number of significant bits behind it (e.g., 192.168.2.0/24). 

■ Bind to MAC Address: This field is used to bind the registration to a specific user agent. To find 

the MAC address for a phone, press the “?” on the telephone keypad. The MAC address will appear 

in the telephone display. 

■ Parameters 1 through 3: Parameters 1 through 3 can be used in various places as generic, extension-related 

parameters. All parameters can be used to TAG extra fields onto the REPLACEMENT 

part of the dial plan, a way of passing information out of the system (in SIP) to another server. This 

is normally used for billing, e.g., sip:*@\r;id1=\x;id2=\y;id3=\z. 

— Parameter 1: If set, it will be used in as CLI if set on an outbound call. 

— Parameter 2: This field is a tagging field. Whatever is placed in this field will appear in the 

SOAP lookup mechanism. 

— Parameter 3: 

■ EPID: EndPoint Identifier is a random number created by the system. The EPID is sent on all 

911 calls, enabling the PSAP to call the extension back directly if the phone is disconnected or the 

person hangs up. The EPID is essentially a DID for that extension and ensures that a returned 

call will reach the extension from which the call originated even if there is an auto attendant in the 

middle. This works to enable 911’s E911 service. 

■ Lines: The lines parameter is used during provisioning to tell the phone how many line keys it 

should allocate (whenever applicable). The line feature is also used to restrict the number of calls 

that an extension can make. For example, if 1 is used in this field, then calls cannot be made 

simultaneously from both the office phone and the home office phone. During an active call, the 

system will neither page the extension nor perform intercom calls to that extension. 

■ Max. number of concurrent registrations per extensions: This setting is used to restricted the 

number of phones that can be registered to an extension. This is useful for service providers or 

enterprises that want to ensure only one phone is registered per extension. 

■ Credit for outbound calls: This setting allows you to place a limit on outbound calling for the 

extension only. You can also place a limit on outbound calling at the domain level (see Credit for 

outbound calls on page 56). 

■ Override Codec Preference: This setting allows you to specify the codec preference for the extension. 

This setting is useful for a remote phone with limited bandwidth, as you can force a low bit 

rate codec like GSM or G.729 to be used on the extension. The format for this setting is similar to 

the Port Setup section of the system. 

■ Emergency Destination Number: This setting is used when you are running the extension in a remote 

location where a different police department should be called. You will need to find the Public 

Safety Answering Point (PSAP) for this number; however, it is not a guarantee that a call will be 

routed there, so ensure that this is tested or approved if your carrier does not have 911 service. 

■ Permissions to monitor this account: Enter the extensions that are permitted to monitor this 

extension. 

■ Log Registration Changes: This setting allows the domain administrator to specify a log level for 

registration log messages specifically for the extension. This setting makes tracking down registra- 


108 


tion problems easier. You can also decide who should receive an email when the registration status 

does change—the administrator, the user, or both. 

Current Registrations 

Displayed at the bottom of the Registrations page are the current registrations (if any exist) for the account. 

Important information includes the type of registration, the registered contact address (as indicated in the 

registration message), and the length of time the contact has been registered. “REGISTER” in the Type 

field indicates a standard registration for receiving calls. Other registrations address specific event types, like 

message waiting indications. 

Clearing registrations: To clear all registrations, use the link at the bottom the page. Be sure to refresh the 

registration from the user agent so that you are able to call it. 

Sending a reboot request: You may also send a reboot request to the phone by clicking the check sync link. 

This way, you don’t have to use the web interface of the device or even go to the phone to reboot it (for example, 

to read new configuration data). 



Manual registrations: You can also add a registration manually. Manual registrations are useful for devices 

like Microsoft speech server that do not support registrations. 

Static registrations: Static registrations can be used to call any other number. 

Permission Settings 

Permission settings allow administrators to define which features the extension can use. The permissions to 

jump into calls and listen to conversations may be legally restricted in certain countries, so use extra caution 

here. The permission to override DND is useful for assistant extensions that are responsible for screening 

calls. 

Note: When entering extensions in this section, use a space as a separator if specifying multiple extensions. 

You may use wildcards, for example, an asterisk (*), to match all extensions of the domain. 

Administration Permissions 

■ Domain Administrator: The permission to manage the domain should be given only to those 

who are allowed to change passwords, create or delete accounts, change dial plans and rates, and so 

forth. 

Call Permissions 

■ Barge into calls of the following extensions: Call Barge-in allows a third party to barge into an 

in-process call. Typically, this third person is either a secretary contacting the boss or possibly a 

co-worker in the next cube who needs help on a call. The permissions to jump into calls and listen 

to conversations may be legally restricted in certain countries; please use extra caution here. Once 

permission has been set, Call Barge-in can be activated using the *81 star code (see page 290). 

■ Teaching (whisper mode) for the following extensions: Call Teach mode allows a third party on 

a line with only one of the parties knowing. This is typically useful in a call center when a trainer 

wants to offer tips to a new agent without the customer knowing. Once permission has been set, 

Call Teach can be activated using the *82 star code (see page 290). 


110 


■ Listen in on calls of the following extensions: In Listen-In mode, a third party can listen in 

without being detected. The two parties speaking to each other are not aware of the listen-in. Users 

with permission to use Listen-in can activate it using the *83 star code (see page 291). 

■ Call the following extensions even if DND is set: If this flag is set, the account will be permitted 

to call the extension indicated in this field, even if that extension is currently on DND. This applies 

only to DND which has been set on the system. The permission to override DND is useful for 

extensions being used by assistants who are responsible for screening calls. 

■ Intercom to the following extensions: This feature is used for communicating with one other 

person. The two parties must each have their own extension number (Intercom will not work between 

two phones that are registered to the same extension). Users with permission to intercom can 

activate it using the *90 star code (see page 281). 

■ Domain accounts that can be put into night mode: This feature enables users to forward domain 

accounts (e.g., auto attendant, hunt group, etc.) to their own extension. Domain accounts that are 

entered into this field can be put into night mode by this user. Star code *80 is used to activate this 

feature (see page 289). 

■ May clean up the following extensions: This feature enables an extension to clean up other extensions 

and is useful in a hotel implementation where voicemail must be cleaned out and greetings 

reset. Star code *84 is used to activate this feature (see page 292). 

■ Only the following extensions may call this extension: This feature can be used to limit the users 

who can call another user’s extension. 


SERVICE FLAGS 

Chapter 

7 

Service flags give companies professional methods for handling callers who call after normal business hours. 

These options include (1) night service, which redirects callers to the extension or phone number of the 

person covering the business during closed hours, (2) voicemail, which allows the customer to leave a message 

and the company to respond as necessary, and (3) a recording of important information (e.g., hours of 

operation, directions, contact information). 

Night 

Service 

Account 

NIGHT 

MESSAGE 

DAY 

MESSAGE 

HOLIDAY 

MESSAGE 

A single service flag can be used for several accounts simultaneously, such as an auto attendant, an agent 

group, and a hunt group. Service flags can be controlled both automatically and manually and can be configured 

so that an LED on a user’s phone will indicate whether the service flag is active or not. Other service 

flag features are listed in Table 7-1 

Table 7-1. Service Flag Features 

Feature Functionality Benefit 

Automatic or 

manual 

Hunt group and 

agent group 

control 

Controlled access 

to flags 

Service flags can be operated either automatically 

or manually. When service flags 

are used manually, users must press a button 

or dial the service flag account number to 

activate or deactivate it. 

Several hunt groups can be used for different 

situations (normal operation, extra busy day, 

shorthanded staff day), and their activation 

can be controlled with manual service flags. 

Automatic mode allows businesses 

with consistent closing hours to have 

the service flag activate on its own. 

Manual mode is useful for businesses 

with irregular closing times and for 

those who want to control it at a 

moment’s notice. 

Quickly shift between plans to 

reallocate calls, services, and staff. 

Advanced planning allows you to 

prepare for the call load. 

Users can switch the service flag on and off. Service flag control can be limited 

to managers at any on-site or remote 

location. 


112 



Auto attendant 

control 

Simultaneous 

multi-account 

activation 

Auto attendants are used to answer incoming 

calls and direct them to other destinations. 

Service flags let you control the 

greeting that is used and where the calls are 

directed. 

The same service flag can be used for multiple 

accounts (e.g., hunt group, agent group, 

and auto attendant). All accounts using the 

flag will switch on or off simultaneously. 

Visual indication When a service flag has been turned on or 

off, the system sends a message to selected 

phones. When the service flag has been 

activated, a designated LED on phones that 

are configured to receive this indication will 

light. 

Cell phone 

control 

Cell phones can be linked to service flags. If 

an extension has subscribed to a service flag, 

then any calls made to that extension will 

be sent to the user’s cell phone when that 

service flag is on. 

Setting up a Service Flag Account 

This section provides instructions for creating and configuring a service flag account: 

1. From the Accounts tab, click Create and select Service Flag from the dropdown: 

Use several auto attendants to reduce 

the risk of losing business. Create 

one for day, lunch, and night (or 

prepare one in advance for unexpected 

closures) and switch as needed. 

Turn them off/on from a remote location 

by phone or via the Internet. 

Prepare several complete business 

communication scenarios, and 

switch to any one of them quickly. 

Viewing the service flag activity can 

be limited to managers at any on-site 

or remote location. 

After-hours calls can be sent to a 

manager’s cell phone automatically. 

By using multiple after-hours service 

flags, calls can be sent to the appropriate 

manager on different days of 

the week. 

2. In the Account Name field, enter an extension number and/or alias, i.e., a DID number. When 

entering an alias in addition to an extension number, use the following syntax: 

505/9781234567 

Alias accounts can give the impression that your system has more accounts than it actually has and 

can clutter it up. If this happens, you can hide aliased accounts under accounts page at the domain 

level (Domain > Settings). 

To create multiple accounts at the same time, use a space between the numbers: 

505 506 507 


3. Click Create. When the page reloads, click List and open the account. 

4. Use the following information to complete the service flag configuration: 

Chapter 7: Service Flags 113 

■ Account Number: This field takes an extension number and/or DID number(s). The number 

of DIDs is unlimited. 

■ Mode: The service flag can be controlled either manually or automatically. Choose a mode: 

— Manual: A manual service flag must be activated/deactivated manually (see “Setting/Clearing 

a Service Flag” on page 117). 

— Day/Night: A day/night service flag is controlled automatically according to a schedule you 

create. Following is a sample schedule: 

When creating a schedule for a day/night service flag, refer to Table 7-2. 

■ Display Name: This setting allows you to enter a small description of the service flag to distinguish 

it from other service flags: 


114 


■ Current State: This setting is reserved for the manual mode only. Choose Set to activate the 

service flag and Clear to deactivate it. 

■ Confirmation Tone: Choose the confirmation sound you would like to hear upon turning the 

service flag on or off. 

— Beep tone: The system will play a short beep to indicate that the service flag has been set or 

cleared. 

— Service active/inactive 

— Day/night 

■ Extensions that may change status: Enter the extensions that are permitted to change the service 

flag. An asterisk indicates that all users are permitted to change the status. 

■ Permissions to monitor this account: Enter the extensions that are permitted to monitor the 

service flag. 

Table 7-2. Day/Night Service Flag Requirements 

Topic Description 

Format You can use either of the following formats; however, both formats require start and 

end times (e.g., HH:MM-HH:MM): 

■ 24-hour format (e.g., 21:00 for 9:00 p.m.) 

■ English AM/PM style (e.g., 9:00P for 9:00 p.m.) 

Multiple time 

segments 

To specify more than one time segment, use a space between the two segments: 

■ 9:00-12:00 1:00P-7:00P 

Holidays To specify the days on which you do not want the schedule to be activated, such as a 

holiday or any other day, use the holiday field: 

■ To specify one day, use the MM/DD format (e.g., 11/24 for November 24). 

■ To specify two days, use the MM/DD format with a space between the days 

(e.g., 12/24 12/25 for December 24 and 25). 

Noon time To specify time during the noon hour (between 12:00 and 1:00 p.m.), you can use 

12:00, 12:00P, or 12:00PM. All are treated as day time 12 o’clock. 

Midnight To specify midnight, use either 00:00 or 24:00, depending on the start and end times. 

■ To start a time period at midnight, use 00:00 (e.g., 00:00-08:00). 

■ To end the time at midnight, use 24:00 (16:00-24:00). 

Important: The day/night service flag will be active during the time outside the hours that are set in the 

service flag. For example, if the service flag is set to 09:00A-5:00P, the status of the service flag will show Set 

for all times after 5:00 p.m. and before 9:00 a.m. If night service is being used, calls will be routed to the 

indicated extension. 

Linking a Service Flag to an Account 

Once you have created a service flag, it must be linked to an account before the system will use it. This can 

be done using two methods: night service or message only (which can also be configured to allow the caller 

to leave a voicemail). 


Method 1: Night Service 


The Night Service setting allows you to provide service to callers during after-hours. Callers will be forwarded 

to the new number based on the service flag schedule. 

In our example, when service flag 600 is active, auto attendant 500 (an auto attendant with a day message) 

will forward calls to extension 501. 

1. Create a day Auto Attendant (see Chapter 8). 

2. Create a day greeting for the day auto attendant: 

5* 

9 

8 

5 0 0 

3. Open the auto attendant account. 

4. In the Service Flag Account field, specify the name of the service flag that will be used (see below). 

5. In the Night Service Number field, specify the account where calls will be directed (see below). 


When the service flag is active, all calls will be forwarded to extension 501. 

Multiple Night Service Numbers 

You can use service flags to forward callers to multiple destinations depending on the time of day (i.e., a different 

destinations for different shifts): 

1. Create a day auto attendant. 

2. Create the necessary service flags. 

3. Create the night service accounts (these can be auto attendants or extensions). 

4. Create a greeting for each account being used as night service: 

5* 

5* 

5* 

9 

9 

9 

8 

8 

8 

5 0 1 

5 0 2 

5 0 3 


116 


5. From the day auto attendant account, link the service flags to the other accounts. 

Method 2: Message-Only and Voicemail Options 

If you don’t want to provide service to callers during after-hours, use the IVR link from the auto attendant 

account. From this link, you can provide callers with a single message (e.g., “The office is closed, etc.”) or 

provide them with numerous options: the option of dialing another extension, the option of leaving a voicemail, 

or the option of remaining on the line for general assistance. 

1. Create a day auto attendant. 

2. Create a day greeting by entering *98 and the account number (e.g.,*98599). 

3. Create a night service flag and a schedule for the system to follow. 

Important: When listing the time schedule in a night service flag, be sure to specify the hours that 

your business is opened, as opposed to closed. Service flags become activated during the hours outside 

your listed schedule, so if the service flag is set to 09:00A-5:00P, the service flag will become activated 

anytime after 5:00 p.m. and before 9:00 a.m. 

4. Open the auto attendant account. 

5. Click IVR, as shown below. 

6. Enter the service flag into the Service flag fields, as shown below. 

7. From the auto attendant, record a greeting for the night service flag. (In the example shown below, 

the auto attendant account is 599.) 

5* 

5* 

9 

9 

8 

8 

5 9 9 

5 9 9 

5* 

5* 

1 

2 


The audio files will be listed under Current File: 

8. To give the caller additional choices, choose from the options below: 


(For the last two options, see “Dial by Name” on page 128 and “Timeout Handling” on page 

127.) 


Note: If you need to change the message multiple times throughout the day (e.g., during lunch and afterhours), 

create a service flag for each period. 

Setting/Clearing a Service Flag 

The Set/Clear status of a service flag can be seen from the Accounts tab: 

Automatic service flags will become activated according to a schedule that you create. Manual service flags, 

on the other hand, need to be manually activated using one of the following methods. 


118 


Setting a Manual Service Flag from the Phone 

Call the service flag extension each time you want to change its state. The confirmation tone that is played is 

based on the Confirmation tone setting: 

Setting a Manual Service Flag from the Web Interface 

A manual service flag can also be set by using the Current State setting in the service flag account. This setting 

can be toggled between Set and Clear. 

Service Flags and Buttons 

Using a Button to Activate a Service Flag 

To activate/deactivate a service flag using a button, add a service flag button to the button profile of the 

extension that will be using it. (The instructions below assume you are adding the button from Domain > 

Settings > Button; however, it can be done from the extension itself.) 

1. Open a button profile (or create one). 

2. In the Name field, name the button (use a numerical value). 

3. From the Type dropdown, choose Service flag. 

4. Enter the service flag account number into the Parameter field. 

5. Save the profile. 

6. Open the extension that will be in charge of activating/deactivating the service flag. 



7. Click Buttons. 

8. From the Configuration Profile dropdown, assign the profile to the extension. 

9. From the phone, press the button configured as Service flag. It will toggle between Set and Clear. 

The values assigned to the buttons correspond with the locations shown below: 

1 

2 

3 

4 

5 

6 

7 

8 

9 

10 

11 

12 


120 



AUTO ATTENDANT 

Chapter 

8 

The auto attendant functions like a virtual receptionist, connecting incoming calls to extensions and other 

devices that have been configured to the system. The auto attendant often announces a company’s name, 

followed by a selection of dialing options. A typical welcome message might be “Thank you for calling 

Company ABC, “for Sales, press 1,” “for Accounts, press 2,” and “for the dial-by-name directory, press 4.” 

Callers press the number that corresponds with their selection and are directed accordingly. 

“If you know 

your party’s 

extension, you 

may dial it now.” 

“Press 1 

for Sales.” 

Figure 8-1. Auto Attendant Welcome Message 

“Welcome to 

CompanyABC” 

“Press 2 

for Support.” 

“Press 3 

for Accounting.” 

“For our 

dial-by-name 

directory, 

press 4.” 

Tasks performed by the auto attendant include searching for names, entering extension numbers, protecting 

certain extensions (e.g., phones that are on DND), and redirecting calls to external numbers. The auto attendant 

also performs other less visible tasks like calling back when an extension becomes available or redirecting 

calls into the mailbox. 

The auto attendant can be configured to play time-dependent messages so that callers can be informed about 

a company’s hours, and it can be configured to redirect calls to other extensions after normal business hours. 

This is known as night service and is configured through service flags. 

Table 8-1. Auto Attendant Features 


Dial-by-name 

service 

Non- 

reachable extensions 

The dial-by-name feature helps callers reach 

the correct person when they don’t know the 

extension of the party they are trying to reach. 

Callers can enter the first or last name of person, 

and the auto attendant will play back the 

names that match the entry. Once the caller 

has made a choice, the auto attendant will connect 

the call. 

The auto attendant can be programmed with a 

list of extensions that callers will be unable to 

reach. 

The auto attendant can send a 

caller to an extension, a conference 

room, or a cell phone. This 

improves the caller’s experience. 

Call flow can be controlled for 

executives and others needing this 

service. 


122 



Transfer to cell 

phone 

By setting an extension’s mailbox escape account 

to an auto attendant, callers will be able 

to access that extension’s Personal Virtual Assistant, 

which can be configured to provide the 

caller with the option of being transferred to 

that user’s cell phone. 

Selectable greetings The auto attendant can be configured to play 

different greetings at various times of day. This 

feature allows you to have a different morning, 

lunch, after-hours, weekend, and holiday 

greetings. 

Nested auto attendants 

Camp On and 

callback 

Cell phone recognition 

Blacklist 

control 

Direct or WAV 

greetings 

An auto attendant can be created within another 

auto attendant. 

Callers with a valid caller-ID can “camp on” 

a busy extension and receive a callback once 

the extension becomes available. (The system 

tracks a person’s availability by monitoring keypad 

activity.) Offer Camp On must be enabled 

from the domain settings (see page 58). 

Callers who call into the auto attendant from 

a cell phone that has been configured to their 

extension are recognized by the system and can 

access the Personal Virtual Assistant. 

Callers on an extension’s blacklist are required 

to identify themselves before being transferred 

by the auto attendant. 

Auto attendant greetings can be recorded 

directly into the system from selected phones. 

Greetings can also be recorded by a professional 

recording studio and uploaded into the 

system via the Internet. 

Fax detection The auto attendant can be configured to listen 

for a fax calling tone (the CNG tone). Upon 

detecting the tone, the system will transfer the 

call to its destination. 

How an Auto Attendant Works 

Callers can be provided with additional 

options, such as directions 

and latest news. 

A company can present a better 

image. 

Nested auto attendants allow callers 

to get closer to their desired 

destination. 

The caller is given a choice when 

an extension they are trying to 

reach is busy, and communication 

is expedited. 

Users can access star codes when 

away from their extension phone. 

Workers are not disturbed by 

unwanted callers. 

Professional greetings give a company 

a positive image. 

You can publish your main telephone 

number as your fax number 

(a separate telephone line is 

not necessary). 

This section details the events that occur from the time the caller first reaches the auto attendant until the 

caller is connected. 

Welcome Message 

When someone calls into the auto attendant, the auto attendant plays a welcome message. The welcome 

message can be a standard message or one that varies according to time of day or week (see “Multiple Messages—Time-Based 

Configuration” on page 124). The caller can press the star key to skip the welcome mes- 


Chapter 8: Auto Attendant 123 

sage and go directly to the menu where the auto attendant asks for the destination, or the caller can listen to 

the available options (see Figure 8-1). 

Processing User Input 

Once the user has made a selection, the system must process the selection. Depending on whether the user 

entered numbers or letters, the process varies. 

Numerical Input 

In the simplest case, the system collects the digits for an extension and calls that extension, but the system 

also checks for direct destinations that might be associated with the input. Direct destinations are pre-configured 

destination fields that can be used to direct inbound calls to popular destinations like sales, support, 

and accounting (direct destinations are discussed on page 129). When an extension overlaps with a direct 

destination, the system calls the direct destination unless the auto attendant has been configured with a 

timeout. A timeout gives the system enough time to assess the user’s complete input before connecting the 

call. For example, if a user enters 155 for an extension, but the system is using a “1” as a direct destination 

to Sales, the caller will be connected to Sales unless a timeout is in place. Figure 8-2 shows how user input is 

processed (based on an auto attendant similar to that shown in Figure 8-1). 

Input 

1 

4 

2 

5 5 

5 5 

Processing Result 

Caller is connected to Sales 

(if no timeout is in place). 

Caller is connected to extension 455. 

Caller is connected to Support. 

Figure 8-2. Processing Numerical and Dial-by-Name Input 

Dial-by-Name Input 

When a caller uses the dial-by-name directory and has entered enough digits, the auto attendant searches 

through a name-searching directory. The digits correspond to the letters that are printed on the keypad button 

(2 = ABC, 3 = DEF, etc.). Key “0” maps only to the symbol “0,” and key “1” maps to all other characters. 

In cases when the result is not unique, the auto attendant recites a list of matching names. When the 

user does not press anything, the auto attendant redirects the call to another destination. 

Ringing an Extension 

Once the auto attendant has processed the user’s input, the auto attendant calls the extension. If the extension 

is not available and the user has only one line (see user’s Registrations tab), the caller will be sent directly 

to the extension’s voicemail. If more than one line has been designated to the extension, the caller will 

hear ringback and the user will hear a beeping noise (call waiting). If the extension does not pick up within 

the specified timeout, the caller will be sent to voicemail. 

If the extension has specified a cell phone number and the time of day permits the inclusion of the cell 

phone, the auto attendant will include the cell phone in its list of ringing devices. This step may be delayed 

for a few seconds, allowing a user who is near the desktop phone the chance to pick up the call from that 

phone first. The ringback tone from the cell phone is ignored, preventing the caller from knowing where the 


124 


call is picked up. While the extension is ringing, the caller can press the star key to cancel the call and enter 

a new number. 

When the Extension Does Not Pick Up 

If a user has programmed a redirection from an extension and does not pick up the call from that phone, the 

auto attendant will handle the call in one of two ways: 

■ If the call comes from an outside user, the caller will be sent to the extension’s voicemail after the 

mailbox timeout has lapsed. 

■ If the call comes from either another extension or from a cell phone that is configured to an extension, 

the user can either camp on the extension or leave a mailbox message. 

If the user has configured the extension to receive emails for missed calls, the auto attendant will also send an 

email about the missed call. 

Blacklisted Callers and Anonymous Calls 

Once the auto attendant intercepts a call from a blacklisted contact or from a caller who does not have a 

caller-ID, the caller is required to leave a name. The auto attendant then calls the extension and informs the 

user of the call. The user can then accept or reject the call. If the call is rejected, it is sent to the mailbox or 

the busy-extension rules apply. (The caller will hear music while waiting.) 

Multiple Messages—Time-Based Configuration 

A single auto attendant can be configured to play up to five different messages at different times of day. This 

is done through the use of service flags (see Chapter 7, “Service Flags”), which must be configured to correspond 

with the message that is linked to it. 

Setting up an Auto Attendant 

This section provides instructions for creating and configuring an auto attendant account. 

1. From the Accounts tab, click Create and select Auto Attendant from the dropdown. 



505/9781234567 




can clutter it up. If this occurs, you can hide aliased accounts under accounts page at the domain 

level (see page 58). 




505 506 507 


4. Once the page reloads, click List, open the account, and configure it as described in the following 

sections. 

Auto Attendant Behavior 

The identity and behavior of the auto attendant (Figure 8-3) are discussed in this section. Identity settings allow 

you to determine how the auto attendant can be reached by outside callers (e.g., extension number, DID 

number, or a combination of both). Behavior settings allow you to specify a wide range of settings, e.g., 

dial-by-name settings, recording settings, and monitoring settings. 

Figure 8-3. Auto Attendant Behavior 

Identity 

■ Account Number(s): This field takes extension number(s) and/or DID number(s). The number of 

DIDs that can be entered into this field is unlimited. 

■ Name: This field allows you to create an alias so that you can quickly identify the auto attendant 

among other auto attendants. For example, you might use the names Day Auto Attendant and 

Night Auto Attendant to distinguish them from one another. 


126 

Behavior 


■ Extension Input: This setting allows you to determine when the auto attendant will begin the 

search for an extension that matches the user’s input. The available options are detailed below: 

— When Extension Matches: The auto attendant will wait until the caller’s digit sequence 

matches an existing account. Once the auto attendant finds a match, it will call that extension. 

This mechanism is useful when accounts of varying name length are used; however, it 

might be annoying to callers who enter a nonexisting number since the auto attendant will 

never begin the search. 

— After 1/2/3/4/5 Digit Input: The auto attendant will wait until the correct number of digits 

has been entered before it will begin looking for an account that matches. If the account does 

not exist, the system will play an announcement indicating that the extension does not exist. 

— User Must Hit Pound: The auto attendant will wait until the user hits the # sign before 

searching for an extension. This mode is useful in variable-length scenarios. 

■ Say Name: When this setting is enabled, the system announces the name that was recorded for 

the account being called. If the user did not record a name, the system will play back the extension 

number. 

■ Accounts that may record a message: Enter the extensions that are allowed to record auto attendant 

messages. 

Warning: Leaving this field blank can be risky, as all users will be permitted to record messages, and 

there is nothing preventing the creation of bogus recordings. 

■ Dial Plan for outbound calls: Choose the dial plan that you would like the system to use when 

routing calls to an outside provider. (For more information about dial plans, see Chapter 5, “Dial 

Plans.”) 

— Domain default: (This is the dial plan that is on the settings page of the domain.) 

— Outgoing: (You could make it dial plan restrictive.) 

■ ANI: Automatic Number Identification (ANI) is a mechanism that allows phone companies to 

determine which account should be charged for a call. The ANI is automatically sent to wherever 

the call is made, and although ANI seems similar to caller-ID, the two are quite different. Caller- 

ID often reflects the main number of a business, rather than the actual number that made the call. 

In addition, the ANI shows the class of service of the phone number, while caller-ID shows only 

the name and number. 

■ Send daily CDR report to: The call data record (CDR) lists all calls that come into the auto attendant. 

The report is sent nightly at midnight to the email address listed in this field (e.g., voicemail@ 

snom.com). Only one email address is allowed in this field. 

■ Set Language: The auto attendant supports multiple-language environments. You may explicitly 

specify the language that should be used as the primary language. This setting may differ from the 

default language in the domain. If you do not select a language, calls that come in from another 

auto attendant will use its chosen language if that auto attendant’s language has been set. 

■ Second Language: If you are operating the system in a dual-language environment (for example, 

Germany and France), you may want to offer callers a second language. To set this up, use direct 

destinations, as shown in Figure 8-4. The destination that you send the caller to should have the 

announced language. If you do not enter a destination, the system will switch the language but 

continue waiting for input. When a hash sign is placed after a direct destination, the system will 

wait a few seconds to prevent a conflict with extensions beginning with the same number (in this 

case, a “1,” as shown in the example below). 


Figure 8-4. Auto Attendant©s Second Language Feature 


Note: Although the system’s language default is limited to two languages, additional languages can 

be offered by creating multiple auto attendants in different languages. For example, you could set 

up the first auto attendant to offer a menu of language choices (e.g., “for English, press 1,” “for 

French, press 2,” or “for German, press 3”). The first auto attendant would be the English auto 

attendant, the second would be French, and the third would be German. In a scenario such as this, 

each auto attendant would be set to its respective language. See “Audio Prompts” in Chapter 17 for 

more information on audio prompts and languages that can be downloaded. 

■ Permissions to monitor this account: By default, the snom ONE telephone system allows 

any phone within the domain to monitor an auto attendant account. The account is monitored 

through the busy lamp field (BLF). To permit only certain extensions to monitor the account, enter 

those extensions here. 

■ Wait before answering the call: This setting determines the number of seconds that will lapse between 

the time the system picks up the call and the time the welcome message begins playing. Use 

this setting if the initial part of the greeting is cut off. 

■ Treat star (*) as special key: This setting allows you to configure the system so that the star (*) key 

will receive no special treatment when entered by the user, e.g., it cannot be used to redirect the 

user to the operator. If this feature is deactivated, the star will be treated like any other key (0-9) 

when the auto attendant is playing a message. 

Timeout Handling 

The timeout functionality allows you to provide callers with a way to exit the auto attendant if they are calling 

from phones that have no dual-tone multi-frequency (DTMF) signaling (i.e., Touch-Tone). Without 

DTMF signaling, callers are not able to press the auto attendant options and would otherwise be stuck. The 

timeout functionality allows you to specify how long the auto attendant should wait before it redirects the 

call or hangs up. 


128 


■ Redirect Number: This setting allows to you to tell the auto attendant where to direct a call when 

a caller does not enter the required information within the set timeout period. Enter an extension 

number into this field. 

■ Timeout(s): The auto attendant will redirect a caller to the redirect number after a specified number 

of seconds. 

■ Hangup Timeout: The auto attendant will terminate a call after a specified number of seconds if 

the user does not enter anything. This feature call help clear a call when the PSTN gateway is not 

able to detect that the caller has already hung up. 

■ Number of times to repeat the welcome message: This setting determines the number of times 

the welcome message is played. 

Night Service 

Night service is used to redirect calls to other extensions or phone numbers during certain times of the day/ 

night or other events that are outside normal business hours. Before night service can be used, a Service Flag 

account must be created for each night service number (see Chapter 7, “Service Flags”). 

■ Service Flag Account: Once you have created the service flag account(s), enter the number of the 

account(s) as shown below. 

■ Night Service Number: Once a service flag is in place, the auto attendant can redirect calls directly 

to the indicated night service number(s). Internal extensions or external number phone numbers 

may be used in this field. 

Note: You may specify more than one night service flag (separated by a space). The first service flag 

account will correspond to the first night service number, and the second service flag account will 

correspond to the second night service number, etc. 

Note: You can also use a special character pattern (#L) to activate the service flag when all agents 

are logged out, even if the logout period is within normal working hours. All calls will automatically 

be redirected to the specified night service number(s). 

Dial-by-Name 

The Dial-by-Name feature allows callers to enter numbers from their telephone keypad. The system will 

search for corresponding names: 



■ Input that triggers name search: Specify the number of digits that a user may enter. 

■ Start search: This setting determines the number of digits that will be required before the system 

begins searching for corresponding names. When there are several matches, the system will list the 

available matches in a menu. The caller can always cancel the search by pressing the star (*) key. 

Direct Destinations 

The Direct Destinations feature is somewhat like a built-in version of the IVR system. To direct inbound 

calls to specified extensions, you can use the pre-configured destination fields and link them to pre-recorded 

announcements and user input options. Using the sample shown below, the auto attendant’s welcome message 

will be as follows: “For Sales, press 1. For Support, press 2. For Accounting, press 3. For all other inquiries, 

press 0.” (The user input options are linked to extensions 555, 518, 511, and 570.) 

When configuring straightforward, uncomplicated auto attendants, direct destinations are a great solution. 

However, when configuring auto attendants that require advanced IVR development and functionality, the 

IVR node is recommended (see Chapter 13). 

Once the direct destination links have been established, the system will call the destination number whenever 

a caller enters the number that is associated with it. In the sample shown above, when the caller presses 

1, the call will be connected to extension 555. 

By placing a pound sign after the direct destination (e.g., “1#”), the system will wait 3 seconds before dialing 

the direct destination. This is useful if you have extension numbers in the 100 range (101, 102, etc.). The 

3-second delay ensures that the caller’s complete input (e.g., 101) will be processed rather than just the first 

digit. 

■ Input number: This number can be one or multiple digits; however, the system dials direct destinations 

immediately after a user has provided keypad input, so overlapping between a direct 

destination and an extension number can be a problem. For example, extensions starting with 

“1” would conflict with a direct destination of “1” because the system would be unable to dial the 

extension number. The best way to avoid this situation is to choose extension numbers that do not 

overlap with either direct destinations or mailbox and outbound call prefixes. The extension range 

4xx through 7xx meets these criteria. 

If circumstances render it difficult to change the extension assignments (e.g., business cards with 

extension numbers are already in circulation), a timeout mechanism can be used. By placing a 

pound sign after the direct destination (e.g., “1#”), the system will wait for 3 seconds before it dials 

the direct destination. 


130 


To redirect fax messages to a specific destination, you can use the direct destination “F”. The CNG 

tone that announces a fax tone is recognized by the system and is translated into the “F” key. 

■ Destination: This number can be either an internal number (e.g., an extension or conference 

room) or an external number. 

When more than ten direct dial options are needed, auto attendants can be nested within one another 

as shown in the next section, “Nesting Auto Attendants.” 

■ Gap time: This setting determines the number of seconds that will lapse between the audio: e.g., 

“For sales, press 1” (3 seconds), “For accounting, press 2” (3 seconds), etc. 

Nesting Auto Attendants 

Nesting auto attendants within other auto attendant allows for a sophisticated hierarchy of information. Figure 

8-5 depicts three auto attendants nested within the main auto attendant. This model could be developed 

even further to include nine additional auto attendants, one for each of the boxes shown to the right (i.e., an 

auto attendant could be created for product A that would include prompts that are specific to that product). 

Auto 

Attendant 

For Sales, press 1. 

For Accounting, press 2. 

For Support, press 3. 

Figure 8-5. Nested Auto Attendants 

1 

2 

3 

For product A, press 1. 

For product B, press 2. 

For product C, press 3. 

To make a payment, press 1. 

To hear your balance, press 2. 

To speak to an agent, press 3. 

To reset password, press 1. 

For help with xxxx, press 2. 


To create a nested auto attendant, individual auto attendant accounts must be set up (see “Setting up an 

Auto Attendant Account” on page 124). Using the model shown in Figure 8-5, four auto attendant accounts 

are needed. Extension 554 can be the main auto attendant, and extensions 555, 556, and 557 can be 

nested within it (Figure 8-6). 


1 

2 

3 

1 

2 

3 

1 

2 

3

554 

For Sales, press 1. 

For Accounting, press 2. 

For Support, press 3. 

555 

556 

557 

Figure 8-6. Nested Auto Attendants and Direct Destinations 


For product A, press 1. 

For product B, press 2. 

For product C, press 3. 

To make a payment, press 1. 

To hear your balance, press 2. 


To reset password, press 1. 

For help with xxxx, press 2. 


To “nest” the three auto attendants within the main auto attendant, they must be listed as direct destinations 

of the main auto attendant. 

Figure 8-7. Direct Destination Settings from Web Interface 

Welcome Greetings 

Overview 

From the auto attendant’s IVR settings (Figure 8-8), you can configure the account so that you can control 

which announcement plays when callers reach the auto attendant. A single auto attendant to play up to five 

different messages at different times of day. For each greeting, a service flag will be needed (see Chapter 7, 

“Service Flags,” for instructions on creating service flags), and a corresponding greeting will need to be either 

uploaded as a WAV file or recorded using the Record Greeting (*98) star code. 

To set up the auto attendant’s messaging system, follow these steps: 

1. Open the auto attendant account and click IVR: 


1 

2 

3 

1 

2 

3 

1 

2 

3

132 


2. Enter your greeting into the system either by recording the message using the star code (*98) or by 

uploading it as a WAV file. See the next section, “Recording the Greeting.” 

Recording the Greeting 

Before an extension can record a greeting for an auto attendant account, the extension must have permission 

to do so. The permission setting can be found in the auto attendant account and is called Accounts that 

may record a message. 

Once an extension has permission, greetings can be recorded using the Record Greetings star code (*98). 

They can also be uploaded as a WAV file by anyone with permission to access the account. 

To record greetings using *98: 

1. From the telephone keypad, enter *98. 

2. Enter the auto attendant account number. 

3. Enter the greeting number following by the star key (e.g., 1*). 

4. Press the confirm button. 

5. The greeting will automatically be assigned to that auto attendant. 

5* 

9 

8 5 0 0 5* 1 

6. Repeat these steps for each greeting, recording a new greeting each time (e.g., *2, *3, etc.). 

The greetings will appear in the Current File field as shown below: 



Note: When a single greeting will be used for all calls, you do not need to specify a greeting number when 

using the *98 star code. This greeting is considered the default greeting and will not appear in the Current 

File field. 

5* 

9 

8 

5 0 0 

To record greetings by uploading WAV files: 

1. Using a recording device, record the greeting (use an 8 kHz mono, 16 bit file). 

2. Go to the auto attendant’s IVR tab and click Browse. 

3. Select the recorded file. 

4. Click Save. The file will be placed into the snom/snomONE/recordings directory. 

Filing System and Naming Conventions 

All greetings are automatically placed into the recordings directory whether they have been recorded 

using *98 or uploaded into the system through the web interface. Unlike uploaded greetings, which can be 

named according to a user’s preference, greetings that have been created with *98 are named according to the 

system’s syntax rules. Its naming system is based on the greeting number that was recorded and the account 

type (personal, auto attendant, agent group, etc.) from which it was recorded. However, the actual account 

number is never reflected in the file name of the greeting. For example, if a user creates a personal greeting 

using *98*1, then the file could be named personal10-1.wav, personal23-1.wav, personal24-1.wav, 

etc. (the numbers 10, 23, and 24 are internal user IDs and could actually be any number). 

Table 8-2. Automatic Naming Conventions for *98 Greetings 

Account Type Input Internal Naming System File Name 

Personal *98*1 personal-1.wav personal10-1.wav 

Personal *98*2 personal-2.wav personal10-2.wav 

Auto 

attendant 

*9870*1 att1_.wav att1_11.wav 

Auto 

attendant 

*9870*2 att2_.wav att2_11.wav 

Agent 

group 

*9873*1 att2_.wav acd1_14.wav 

Agent group *9873*4 att2_.wav acd4_14.wav 

Note 1: Internal user IDs are generated by the system. 


134 


Note 2: Numbers in blue indicate account numbers. 

Greetings can be viewed from either the Windows environment (Figure 8-8) or the command prompt (Figure 

8-9). 

Figure 8-8. Viewing Greetings from the Windows Environment 

To view greetings from a command prompt, open a command prompt and navigate to the following path: 

C:\snomONE\snom\recordings. When navigating, you can use the following steps: 

1. cd .. (this will step you back a directory; you may need to repeat this command if you are not at 

the C prompt yet) 

2. cd Program Files 

3. cd snom 

4. cd snomONE 

5. cd recordings 

6. dir (this will show the list of WAV files) 

Figure 8-9. Viewing Greetings from the Command Prompt 


HUNT GROUPS 

Chapter 

9 

A hunt group is a set of extension numbers that are embedded inside a single virtual extension. Incoming 

calls are rotated through the pool of extensions until someone answers and the caller is connected. (Cell 

phones can also be configured to be included in calls made to a hunt group.) 

500 

Stage 2 

Figure 9-1. Hunt Group Stages 

Stage 1 

508 

Stage 3 

509 

508 

510 

502 

508 

520 

523 

Final stage 

hunt down cell phone ext 

The extension numbers that are part of a hunt group can be arranged into groups known as stages. Each stage 

is configured to ring for a specified period before the system begins ringing the next stage. A last-resort or 

final stage number (often an auto attendant or a cell phone) can also be used when none of the extensions 

pick up the call. A common use case for the hunt group is to use the hunt group as the main number for 

incoming calls, then put the receptionist, secretaries, and assistants on the different stages. 

Table 9-1 lists the hunt group features. 

Table 9-1. Hunt Group Features 


Custom Each hunt group can be configured with Employees can determine which hunt 

ringing different ring tones. 

group is calling. 

BLF indicator The system can be configured so that the Hunt group activity is visible through the 

busy lamp field indicator lights up when 

the hunt group is in use. 

BLF indicator. 

Group name Hunt groups can have a unique name and Helps management decide who belongs in 

caller-ID can be configured to display the name of each named group. 

the group, the caller’s caller-ID, or both 

when a caller enters a hunt group. 

Employees know the nature of the call 

before answering it. 

Three-stage Hunt groups can be configured to ring 1–3 Managers can direct calls to selected 

extension different groups of extensions. The system groups of extensions in a specified order. 

ringing will ring each group in succession until the 

call is answered. 


136 



Final stage to 

cell phone 

Ring stage 

duration 

Recordable 

calls 

After calling the extensions listed in stages 1 

through 3, the system calls the “final stage” 

number, which can be an auto attendant, 

an extension, or an external number. 

Hunt group stages can be configured to 

ring for different durations before the system 

rings the next group of extensions. 

Calls made to hunt groups can be recorded 

and saved on the system. 

How the Hunt Group Works 

Callers are sent to a manager as a final 

means of assistance. This feature can also 

be used to gauge whether the group of 

extensions is too small or whether the 

employees are working properly. 

Call flow control can be established. Intervals 

that keep customers waiting a minimum 

amount of time can be used. 

Recorded calls can be reviewed by management 

to fine-tune employee performance. 

Disputes can be easily resolved by emailing 

the recorded conversation. 

Once a caller has reached a hunt group, the system begins searching for an extension that will answer the 

call. Hunt groups can be configured with one to three stages of extensions, and each stage can consist of any 

number of extensions. When looking for an extension, the system begins with stage 1. Once the ring stage 

duration has been met (page 138), the system stops ringing that group of extensions and begins ringing the 

next stage. This scenario continues until all extensions in all stages have been called. 

As a last-case scenario, the system rings the “final stage” number, which in this case is a cell phone (Figure 

9-2). The final stage number is often an auto attendant or a manager’s cell phone number. 

Incoming 

call 

1. Try 

511, 512, 513 

10 seconds 

2. Keep ringing 

511, 512, and 513, 

and add 802 and 803 

10 seconds 

3. Final stage 

Figure 9-2. Hunt Group Call Flow 

511 512 513 

511 512 513 

804 

802 803 

Note: If an extension belongs to more than one stage, the extension will continue to ring through each stage. 


Setting up a Hunt Group 

Chapter 9: Hunt Groups 137 

This section provides instructions for creating and configuring a hunt group account on the snom ONE 

telephone system. 

1. From the Accounts tab, click Create and select Hunt Group from the dropdown: 



505/9781234567 




can clutter it up. If this happens, you can hide aliased accounts under accounts page at the domain 



505 506 507 


4. Once the account has been created, click List to open the account and configure it as detailed in the 

following sections. 

Identity 

The identity settings allow you to identify the hunt group to hunt group members who are receiving the calls 

and to managers who are monitoring the account. 

■ Account Number: This field takes extension number(s) and/or DID number(s). The number of 

DIDs that can be entered into this field is unlimited. This information can be changed at any time. 

■ Name: This field allows you to name the hunt group. The name allows hunt group members to 

differentiate between calls coming into the hunt group and calls not associated with it. Choose a 

short name if you plan to use the Group name (calling party) option in the From header to ensure 

that it will fit into the display of the phone. This field also distinguishes one hunt group from 

another hunt group on the Accounts page. 


138 

Ring Stages 


The hunt group supports three stages and a final stage. Each stage can be configured to ring for a different 

number of seconds, and the final stage can be either an internal or external phone number. 

ext 500 

Stage 1 

call 

Stage 2 

ext 501 

Stage 3 

ext 530 

Figure 9-3. Hunt Group Search Activity 

ext 504 

ext 502 

ext 520 

Final 

Stage 

If an extension has configured a cell phone and has enabled the When calling the extension in a hunt 

group setting (page 307), the hunt group will also ring that person’s cell phone. 

For each stage: 

1. List the extensions that should be part of that stage. When entering multiple extensions, use a space 

as a separator (only one number is permitted for the final stage). 

2. For duration, specify the length of time that the system should ring each stage before ringing the 

next. The duration of the stages must be specified in seconds. 

3. For unwanted stages, leave the fields empty. 

If all extensions of a stage are unavailable, the system will immediately move to the next stage. 

Note: The number of extensions permitted for the first three stages is limited by hardware only. The system 

must send INVITEs to all extensions in each stage, which requires more CPU. 

Behavior 

Behavior settings allow you to control numerous account behaviors. You can choose a ring melody, a display 

header, and music on hold source. You can also enter a safety net phone number that will be used if no one 

picks up a call. Other settings are here as well. 


cell

Chapter 9: Hunt Groups 139 

■ Ring Melody: This setting allows you to set the ring melody so that hunt group members will be 

able to distinguish regular calls from hunt group calls. 

— Custom 1 through 4: Custom ringtones allow users in a hunt group to distinguish hunt 

group calls from other calls. Before customized ringtones can be used, you will need to provide 

your own ringtones, modify the ringtones.xml file, then place it into the html directory. 

When the phones receives the alert-info header from the system, it will respond 

with a different ring tone based on the ringtone that was selected from the web interface. For 

instructions on configuring custom ringtones, see “Customized Ringtones” in Chapter 17. 

— External Call: The ring tone will be long. 

— Internal Call: This ring tone will be short. 

■ From Header: This setting allows you to set the display name that hunt group members will see 

when someone is calling. Four options are available: 

— Calling party: This mode displays the calling party. 

— Group name: This mode displays the name of the hunt group that was called: 

— Group name (calling party): This mode displays both the group name and the calling party. 

— Calling party (CMC): The calling party’s client matter code will appear in the display, if one 

is available. The CMC identifies the customer and is used to expedite billing in offices that bill 

their clients for phone calls. 

■ Dial plan for outbound calls: Select the dial plan that will be used for this account. 

■ ANI: The hunt group uses an ANI (automatic number identification) when sending a caller to an 

external number. This setting is necessary when the user input handling destinations take callers to 

external numbers rather than extension numbers within the domain. 

■ Send daily CDR report to: The call data record (CDR) lists all calls that come into the hunt 

group. The report is sent at midnight each night to the email address listed in this field (e.g., voicemail@snom.com). 

Only one email address is allowed in this field. 


hunt group account. 


140 


■ Music on hold source: From the dropdown list, choose the music that callers will hear while waiting. 

■ Record incoming calls to hunt groups: When this setting is enabled, the system will record all 

calls that come into the hunt group. The recordings can later be retrieved and reviewed. 

Night Service 

Night service allows you to redirect calls to other extensions during off-hours, meetings, etc. Before you can 

use night service, you need to create a Service Flag account (see Chapter 7). 

■ Service Flag Account: Once you have created the account, enter the number of the account as 

shown below. 

■ Night Service Number: With a service flag in place, the hunt group can redirect the calls directly 

to the night service number(s) shown in this field. Internal extensions and external numbers are 

acceptable. 






be redirected to the specified night service number(s). 


AGENT GROUPS 

Chapter 

10 

An agent group, also known as automatic call distribution, is a system that routes incoming calls to a specific 

group of agents. Before connecting to an agent, callers are placed inside a queue, allowing agents to deal with 

incoming calls without losing other callers to busy signals or unanswered phones. While waiting, callers hear 

music and pre-recorded announcements. 

Agent groups are often found in offices that handle high-volume customer calls from callers who do not need 

to speak with a specific person but who require assistance from any of multiple persons, like sales representatives 

and airline reservations attendants. 

Agents must log into the queue to begin receiving calls. When agents have finished for the day or would like 

to stop receiving calls from the queue, they must log out of the queue. 

The following features are available with the agent group: 

Table 10-1. Agent Group Features 

Feature Functionality 

Display name Each agent group can have a distinct name that will appear on the telephone display 

when calls come in. This allows calls to be properly answered and tracked. 

Ring melody Each agent group can have a unique ring melody so that agents can distinguish 

which agent group is ringing. 

Jump in and out Agents may jump in and out of any group to which they are assigned. This allows 

managers to dynamically adjust to a changing workload. 

Minimum num- Managers can establish a minimum number of agents that must remain in a group. 

ber of agents This ensures that an agent will always be available and that calls will be answered. 

Number of agents You can create a comfortable balance of waiting and connecting within agent groups 

per stage by setting the number of agents who can receive calls at the same time. 

Record agent Calls that come into agent groups can be recorded, and recordings can be reviewed 

group calls and emailed later on. 

Call 

Calls can be distributed to agents randomly, according to most idle, or according 

distribution to skill level. Experienced agents can be used to answer calls first. Agents with fewer 

methods skills can be held in reserve. 

Recovery time Recovery time gives each agent a period of time to prepare for the next incoming 

call. 

Auto add callers Callers can be automatically added to the address book, making it easier to return 

to address book phone calls. 

Caller 

For repeated callers, the system rings the agent who provided assistance on the last 

priority call. If the agent is not available, the caller will be sent to the next available agent. 

Caller priority helps callers maintain continuity. 

Music on hold To help set the mood for a particular agent group, distinct music can be provided for 

each agent group. Callers can hear music that relates to their call (e.g., classical music 

for callers looking to buy classical music, jazz for those looking to buy jazz, etc.). 


142 



Multiple Messages can be delivered to callers while waiting inside an agent group, allowing 

announcements you to present additional services or products. 

Gaps 

The amount of time between agent group announcements can be controlled. Dur- 

between ing this gap time, the caller will hear music. This allows you to set up attention- 

announcements getting sequences. 

Agent group An agent group manager can be designated for each agent group. Queue managers 

queue manager can monitor agent group queues. 

Send daily reports Nightly activity reports are delivered to selected email addresses. This allows management 

to review the previous day’s information. 

How the Agent Group Works 

A caller reaches an agent group in one of two ways: either by calling the agent group directly or by calling a 

company’s auto attendant. When agent groups branch off the auto attendant, callers must press the number 

that corresponds with their choice. In Figure 10-1, the caller has reached an auto attendant offering three 

different choices: Sales, Accounting, and Support. The caller hears these options in the form of pre-recorded 

messages. Each option will connect the caller to an agent group, and the caller will be placed inside a queue 

until an agent becomes available. 

waiting 

Ringback 

BUSY 

BUSY BUSY 

BUSY 

Figure 10-1. Agent Group Queueing 

BUSY 

Ringback 

SALES 

AGENT 

GROUP 

1 

2 

886 

3 

waiting 

waitiiiinng 

BUSY 

ACCOUNTING 

AGENT 

GROUP 

887 

BUSY 

AGENT GROUP AGENT GROUP 

SUPPORT 

AGENT 

GROUP 

888 

BUSY 

BUSY 

BUSY 

BUSY 

AGENT GROUP 

waiting 

Ringback 


The Message-Music Cycle 

Chapter 10: Agent Groups 143 

While the caller is waiting, the system delivers a combination of pre-recorded messages and music. The messages 

can be used to provide additional information about your company, or they can be used to provide the 

caller with additional options and/or a way to exit the queue. Each agent group can be configured to play up 

to nine recorded messages. 

Between the messages are “gaps” of music which can be configured to span 5 to 180 seconds. This messagemusic 

cycle continues for as long as the caller is in the queue. Message 0 is considered the entry point message 

(which will never be replayed), and message 1 is the first message. Figure 10-2 shows an agent group 

using six different messages for its message-music cycle. Once message 6 has been played, the system returns 

to message 1 and repeats the cycle. 

5 

Message 

*98888*5 

START 

6 

Message 

*98888*6 

4 

Message 

*98888*4 

Figure 10-2. Messages Alternating with Music 

0 

Message 

*98888 

Agent 

Group 

888 

3 

Message 

*98888*3 

essag 

Me 

1 

Message 

*98888*1 

2 

Message 

*98888*2 

As previously mentioned, this sequencing of messages and music will continue until an agent becomes 

available. Once an agent is available, the system automatically takes the next call out of the queue, rings the 

agents, and puts the call into ringback state. Within any one agent group, only one call at a time can be in 

ringback state, regardless of the number of agents that are available. All other calls remain queued until the 

ringing call is connected and at least one agent becomes available. 

While it might seem logical to conclude that a caller’s time in the queue has ended once the call has been 

placed in ringback state, this is not always the case. If an agent leaves the desk without logging out of the 

queue, the ringback state will continue until the system finds another agent to answer the call. To prevent 

nonstop ringing, you can take two precautions: (1) you can configure the agent group to include additional 

agents at specified intervals in the ringback cycle and (2) you can configure the agent group to play music 

during the ringback state. This will prevent the caller from hearing long periods of ringing in the event there 

are episodes of extended ringback. 


144 


Setting up an Agent Group Account 

This section provides instructions for creating and configuring an agent group account. 

1. From the Accounts tab, click Create and select Agent Group from the dropdown: 



775/9781234567 







775 776 777 



sections. 

Identity 

The identity settings allow you to identify the agent group to both agent group members who are receiving 

the calls and managers who are monitoring the account. 

■ Account Number: This field takes extension number(s) and/or DID number(s). The number of 


■ Display Name: This field allows you to name the agent group. The name allows agent group 

members to differentiate between calls coming into the agent group and calls not associated with 

it. Choose a short name if you plan to use the Group name (calling party) option in the From 

header to ensure that it will fit into the display of the phone. This field also distinguishes one agent 

group from another agent group on the Accounts page. 


Behavior 


Behavior settings allow you to control numerous account behaviors. You can set a ring melody, music on 

hold, and a display header. You can also specify who will monitor the account, manage the queue, and 

receive the daily Call Detail Record (CDR). Gaps between announcements, agent recovery time, and the 

number of agents allowed to jump in and out of the agent group can also be set here. 

■ Agents: Enter the extension numbers of the agents who will comprise the agent group. Extension 

numbers should be entered in the order in which the system should call them. This field can also 

be left empty; however, callers waiting in queue will need to be pulled out manually using the Call 

Pickup feature. 

■ Extensions that may jump in or out (* for all): This setting controls which agents can dynamically 

join or leave the agent group. It can also be used to add non-agents to the agent group, which 

can be useful during especially busy periods when additional people are needed for the queue. If 

a star is used in this field, extensions that use *64 will automatically be listed in this field, i.e., the 

list will grow and decrease as people log in and out. It is recommended that actual extensions be 

entered into this field. 

■ Minimum number of agents for jumping out: This setting specifies the number of agents that 

must remain in the agent group before agents are permitted to jump out. If this setting is too high, 

agents will be unable to log out of the agent group. For small agent groups, a value of 1 is recommended. 

■ Ring Melody: It is important to set the ring melody so that the phone knows which one to play 

and so that agents will be able to distinguish regular calls from agent group calls. 

— Custom 1 through 4: Custom ringtones allow users in an agent group to distinguish agent 

group calls from other calls. Before customized ringtones can be used, you will need to provide 

your own ringtones, modify the ringtones.xml file, then place it into the html directory. 

When the phone receives the alert-info header from the system, it will respond 

with a different ring tone based on the ringtone that was selected from the web interface. For 

instructions on configuring custom ringtones, see “Customized Ringtones” in Chapter 17. 

— External Call: The ring tone will be long. 

— Internal Call: This ring tone will be short. 

■ From-Header: This setting allows agents to see who the call is coming from. By default, the name 

of the calling party is displayed, but the system can be configured to display the name of the queue 

(see Display Name in this chapter), either by itself or with the calling party. When both are shown, 

the system puts the caller-ID in brackets after the display name. This mode requires that the phone 

have enough space to display a long number. When using this mode, choose a short group name so 

that it requires less space. 

— Calling party: The calling party is shown in the display. 

— Group name: The name of the agent group queue (which is set in the Display Name field) is 

shown in the display. 

— Group name (calling party): Both the calling party and the agent group queue are shown. 

(The caller-ID appears in brackets after the agent group name.) When using this mode, 

choose a short group name to ensure that it will fit in the phone display. 

— Calling party (CMC): The calling party’s client matter code will appear in the display, if one 

is available. The CMC identifies the customer and is used to expedite billing in offices that 

bill their clients for phone calls. 


146 


■ Add to the address book and keep it for . . . : Automatically adding a caller’s phone number into 

the address book allows the caller, when calling again, to be routed to the same agent. 

■ Automatically add CMC: When this feature is turned on, the caller’s CMC will appear in every 

CDR that is written on the disk. The CMC is used for automatic billing. 

■ Ringback tone: 

— Regular ringback tone when agent rings: The caller will hear the agent ringing. 

— No ringback tone, continue playing music: Instead of hearing a ringback tone, the caller will 

hear music. If an agent is no longer at his desk but has not logged out of the agent group, the 

caller will hear music on hold instead of ringback. 

■ Caller priority: This feature allows callers to be connected to agents who have handled their calls 

in the past. 

— Queue entry time: All callers will be placed in queue and will remain there until an agent 

becomes available. 

— Try to pull out callers when their last agent becomes available: Callers will be pulled from 

the queue if their last agent becomes available. 



■ Music on Hold source: From the dropdown list, choose the music that the caller will hear while 

waiting for an agent. 


agent group account. 

■ Accounts that may record a message: Specify which accounts may record announcements. When 

this setting is left empty, all extensions will have permission to record announcements. 

■ Gap between announcement(s): From the dropdown list, select the number of seconds that the 

system should wait between each announcement. 

■ Agent recovery time(s): Use this setting to ensure that an agent has adequate time to recover from 

the previous call before taking another call. In busy call centers, 2 seconds might be an appropriate 

setting. However, agents who are required to record notes about the call will need a longer value. 

In these cases, 60 seconds might be more appropriate. The recovery time also allows an agent to log 

out of the agent group. 

■ Queue position announcement: When Announce queue position is selected, an announcement 

will let callers know their position in queue. 

■ Dial plan for outbound calls: Select the dial plan that will be used for this account. For more 

information on dial plans, see Chapter 5, “Dial Plans.” 

■ ANI: The agent group uses an ANI (automatic number identification) when sending a caller to an 

external number. This setting is necessary when the user input handling destinations take callers to 

external numbers rather than extension numbers within the domain. 

■ Send daily CDR report to: The call data record (CDR) lists all calls that came into the agent 

group on the previous day. The report is sent at midnight each night to the email address listed in 

this field (e.g., Fred Feuerstein ). Only one email address is allowed. 

It you want to send the CDR email immediately, click the Try button, rather than waiting until 

midnight. Keep in mind that this is just a link, and you may have to save your list before clicking 

the link. For sample CDR reports, see Figure 10-4 and Figure 10-5. 

■ Send IM to these extensions: The agent group queue can be monitored from SIP phones that 

support the SIMPLE standard for instant messaging and presence. Multiple extensions are permitted 

in this field (use a comma as a space separator). Phone models with larger screens will be able to 

display the full message. 

■ Queue Manager: This setting allows you to assign an extension as queue manager. Once an extension 

has been designated, the agent group’s identity must be listed in the extension’s Show following 

ACD queues field, as shown below. This configuration is set from the user’s extension account. 

Figure 10-3. Permission to be Queue Manager 

Once access to the status queue has been given, the queue manager will be able to view important 

information about the calls. Information includes whether anyone “escaped” out of the queue to 

connect to other destinations, the length of time each caller spent in the queue, and the length of 


148 


time the caller spent talking. Information about the agents is also shown in the report (e.g., agents 

in the queue and number of calls handled by each agent). The queue manager can access the information 

from the Status tab (user account). The information is refreshed every 10 seconds. 

Figure 10-4. Summary of Agent Group Queue 

The report also shows calls that are currently queued as well as information about each call, including 

source, destination, length of time the call has been in queue, and call state. 

Figure 10-5. List of Calls to the Queue 

■ Record incoming calls to agent groups: When this setting is enabled, the system will record all 

calls that come into the agent group. The recordings can later be retrieved and reviewed. 

■ Automatically log out the agent if they missed the last call: This setting will log out an agent 

who has missed a call. Agents need only log back in to return to the agent group. 

■ Allow multiple ACD calls on agent - even if busy: This setting allows a caller to be sent directly 

to an agent, even if all agents are busy. It prevents the caller from sitting there listening to music 

and gives the agent the opportunity to put their current call on hold and take the new call. This 

setting should generally not be used. 


Connecting Caller to Agent 


This section allows you to choose an algorithm for the system to follow when connecting a caller to an agent. 

When a caller needs to be connected to an agent, the system must determine which agent should be called 

next. There are currently three ways of determining this. 

■ Agent selection algorithm: The agent selection algorithm allows you to establish the method that 

will be used by the system when connecting a caller to an agent. Choose from the following options: 

— Random: This algorithm randomly selects the next agent. This is a reasonable algorithm in 

environments where it is not important as to which agent processes the call. 

— Ring longest idle first: This algorithm is designed to balance active call time amongst all the 

agents. The queue keeps a list of the agents and moves an agent to the end of the list when 

an agent connects a call. This ensures that the agent with the longest idle is at the front of the 

list. 

— Use preference from the Agent’s setting: This algorithm uses the extensions listed in the 

Agent setting, as shown below (see Behavior settings on page 145). The agent listed first 

will be tried first. If that agent is busy, then the system will jump to the second one and so 

on. 

■ Ring stage duration (secs): At regular intervals, the system checks agent availability and determines 

whether the next caller should be pulled out of the waiting state. The ring stage duration 

setting establishes this interval. For larger queues, shorter intervals are recommended, as it is 

quicker to get callers out of the queue and into a ringback state. The default value for this setting is 

5 seconds. 

■ Number of agents added per stage: In every scheduling interval, the system adds only a limited 

number of agents from its pool of available agents to its list of ringing agents. For large agent 

groups, a typical value would be one or two, as this will ensure that not all agents receive calls at 

the same time and so that the system CPU is not overtaxed with simultaneous calls. Figure 10-6 

depicts one agent being added to the list of ringing agents every 10 seconds. 

after 10 

seconds 

one additional 

agent added 

Figure 10-6. Ringing Agents at Fixed Intervals 

after 10 

seconds 

one additional 

agent added 


150 


Preventing Lengthy Periods in Ringback or Queue 

Calls That Approach the Head of the Queue 

This section allows you to establish timeout settings that will determine the length of time a call is kept in 

queue or ringback state in a given situation. In this section, you can instruct the system to add additional 

agents or to redirect a call after the call has been in ringback for a given length of time. You can also instruct 

the system to redirect a call once a caller’s wait time has passed a certain threshold. 

When configuring these settings, use the following table: 

Table 10-2. Preventing Lengthy Periods in Ringback or Queue 

After/If . . . Then . . . How/When to Use 

After hear- include the Use this setting when additional people in the office can help out when 

ing ringback following ad- no one else is picking up. 

for (X) 

seconds, 

ditional agents 

(e.g., 41 42). 

Result: When calls are not picked up within the specified period, the 

system calls these extensions. 

After hear- the system will The destination can be either an internal number (e.g., a hunt group) or 

ing ringback redirect the call an external number (e.g., a cell phone or home number). This option 

for (X) to the destina- is similar to the previous option but instead of ringing both the original 

seconds, tion (e.g., 73). and new agents, the system cancels the original agents and redirects calls 

to the new agent(s) or to the external number. 

Result: After-hour callers are directed to a phone that will be picked up. 

If the caller the system will If a caller has waited a while in the queue, this will immediately redirect 

has already redirect the call the call to another destination. 

waited longer 

than (X) 

seconds, 

to the destination 

(e.g., 73). 

Note: The caller is classified as “waiting” when all agents are busy with 

other customers and the new caller is waiting for a free agent. 

Result: If none of the agents free up within the specified period, the call 

will be redirected to the indicated number. 

Mapping Telephone Keypad Numbers with Extensions 

User Input Handling enables the caller to exit the agent group queue and connect to another destination. 

To give the caller this option, use the User Input Handling fields to map telephone keypad numbers to other 

extensions or phone numbers. 


Figure 10-7. Mapping Keypad Numbers to Queue-Exiting Destinations 


Once extensions or phone numbers have been entered into the “Key” fields shown in Figure 10-7, the caller 

needs to be made aware that these options are available. The system will not do this automatically. The place 

to do this is through the greetings that are delivered by the system (using the *98 star code feature). One 

greeting or several greetings can be used to inform the caller that this additional option is available. In Table 

10-3, greetings 2 and 5 have been recorded to let the user know that keys 1 and 2 can be used to exit the 

agent group queue. (Figure 10-7 corresponds to the table. The user must press 1 to be redirected to 503 and 

2 to go to 504.) 

Table 10-3. Using Greetings for Agent Group Escape Routes 

Greeting Message 

0 Welcome to CompanyABC. Your call is important to us. 

Please hold. 

1 Sorry our lines are still busy . . . Have you looked at the 

support site for answer to your question? 

2 Our newest release offers this and that . . . If you want to 

leave a message, please press 1 now. 

3 Another message . . . 

4 Another message . . . 

5 Another message . . . If you would like to leave a message, 

please press 2. 

Recording Agent Group Greetings 

User Input 

Handling 

The number of available greetings within any agent group is nine. Record only those greetings that should be 

used. If no greetings are needed, even an initial greeting, don’t record any. 

When recording greetings, you need to assign a number to each greeting; for example, *0 is the entry greeting, 

*1 is the first greeting, *2 is the second greeting, etc. The number is entered as the last number in the 

string. 

1. Dial *98 followed by the number of the agent group, then the greeting number (e.g., *1). 

2. Press the confirm button on the telephone keypad. 

5* 

5* 

5* 

5* 

9 

9 

9 

9 

8 

8 

8 

8 

8 

8 

8 

8 

8 

8 

8 

8 

8 

8 

8 

8 

5* 

5* 

5* 

5* 

1 

2 

3 

9 

Greeting 1 

Greeting 2 

Greeting 3 

Greeting 9 


503 

504

152 


Listening to Greetings from the Phone 

To listen to greetings quickly from the phone, change the gap between announcements to 1 and call the 

agent group directly from your phone. The recorded messages will be played quickly. 

Night Service 

Night service is used to redirect calls to other extensions or phone numbers during certain times of the day/ 

night or other events that are outside normal business hours. Before night service can be used, a Service Flag 

account must be created for each night service number (see Chapter 7). 

The first three settings in Night Service allow you to make provision for when no agents are available: 

■ When primary agents are logged out, send calls to 

■ When call agents are logged out, send calls to 

■ When all agents are unregistered, send calls to 

The following two settings are used to configure the service flag(s) to night service numbers: 

■ Service Flag Account: Once you have created the service flag account(s), enter the number of the 

account(s) into this field. 




■ Night Service Number: Once a service flag is in place, calls will be directed to the night service 

number(s) indicated here. Internal extensions or external number phone numbers are permitted in 

this field. 



be redirected to the specified sight service number(s). 

SOAP Interface 

The SOAP interface allows you to use external logic to control the queue behavior. See Chapter 13 for a 

sample SOAP message. 

■ Queue Status URL: http://www.snom.com/soap/pbx/queuestatus 

Logging In and Out 

Agent group members can use one of two methods (buttons or star codes) when logging in and out of agent 

groups. 



Important: Before agents can use either method, agents must be included in the Extensions that may 

jump in or out setting. Another setting that must be considered is the Minimum number of agents for 

jumping out setting (shown below). The latter setting is used to prevent the agent group from becoming 

too small; however, if this setting is too high, agents will be unable to log out of the agent group. 

Method 1: Buttons 

This method allows users to use a button on their phone to log in and out of agent groups. When configured 

properly, the button will have a toggle effect. When the button is pressed, users will be logged in to all 

agent groups, and the associated LED(s) on their phone will light. When the button is pressed again, users 

will be logged out of all agent groups and the associated LED(s) will no longer be lit. If users need to log out 

of one agent group before logging into another, multiple Agent login/logout buttons can be configured. See 

page 210 for instructions on creating the button(s). 

Important: Before you can use this button, the star codes for Agent logged in/Agent logged out must match 

(e.g., *64/*64). 

Note: Agent group calls can be forwarded to an agent’s cell phone. See When calling the extension in an 

agent group on page 307. 

Method 2: Star Codes 

When using star codes to log in and out of agent groups, users have the flexibility of logging in and out of all 

agent groups to which they belong or just a specific group. The standard Agent logged in/Agent logged out 

(*64/*65) star codes (without any argument) will log the agent in to or out of all agent groups: 

* 

* 

6 4 

6 5 

Log in 

Log out 

Agents who are members of numerous agent groups and who need to log out of one agent group before logging 

into another can do so by entering the agent group number after the star code. Using the sample shown 

below, the agent will be logged out of agent group 666 and will retain logged-in status for all other agent 

groups. 

* 

6 5 

6 6 6 

Log out of 666 


154 


Note: Agent group calls can be forwarded to an agent’s cell phone. See When calling the extension in an 

agent group on page 307. 

Monitoring Agent Groups 

Agent groups can be monitored from the web interface through the buttons on the phone and from the 

WAC (Web-based Attendant Console). The method you use will be based on the type of monitoring necessary 

and the size of the domain and agent group. The different methods are detailed in the following sections. 

Monitoring from the Web Interface 

An agent group can be monitored from the user account or from the agent group itself. 

To monitor from a user account: 

1. Enter the agent group into the user’s Show following ACD queues field. 

2. In the agent group itself, enter the extension into the Queue Manager field: 

3. Log in as user and click the Status tab. 

4. Click the agent group shown beneath the tabs. 

5. Click Calls to see the active calls. 

6. Click Agents to see which agents are on calls. 

To monitor from the agent group account: 

1. Open the agent group account. 

2. Click Calls to view calls that are either ringing, in queue, or with an agent. 


3. Click Agents to view agent-related information. 

4. Click Call Log to view the call history of the agent group. 

5. Click Email to send an email to a connected agent. 



156 


Monitoring Using Buttons on Phone 

An agent group can be monitored through LEDs on the phone. To do this, create a button profile with 

Agent Group as the button type and the agent group number as the parameter. To monitor numerous agent 

groups from the phone, assign numerous buttons as Agent Group and enter the appropriate agent group 

into the corresponding parameter fields. When a caller is waiting in the group, the LED will blink fast. By 

pressing the button, you can pick up the call. A solid LED indicates that all calls have been picked up. 

Monitoring from the WAC (Web-Based Console) 

The WAC can be used to view the activities of the entire domain, including the status of agent group members. 

Agents are denoted by “AG” and a check mark. If your domain is too large for all the extensions to fit 

on a single screen, the WAC may not display all the agents. 

Users can access the WAC by logging into the web interface and going to Status > WAC. 

This WAC link will be available only if the administrator has enabled the setting shown below (located in 

the user’s account). 

Users can also access the WAC by choosing Console Login from the Login Type dropdown when logging 

into the system. 


PAGING 

Chapter 

11 

Advanced IP phone systems allow even small offices to have a fully functioning paging system without using 

a standalone system. Paging can be thought of as a one-way audio session, as opposed to intercom’s two-way 

audio. Paging allows a single extension to communicate with a few people (unicast paging) or a potentially 

large group of people (multicast paging). Users can page other users by dialing the paging account and 

speaking into the phone. Recipients will hear the message from the speaker on their phones. Several paging 

groups can be configured on the system so that different audiences can be addressed. 

Joe, call 

on line 2 

Joe, call 

on line 2 Joe, call 

on line 2 Joe, call 

on line 2 

Types of Paging 

Unicast Paging 

Unicast paging is a one-to-one call to each participant in the paging group. The system establishes a regular 

call to each extension (Figure 11-1) through standard SIP calling. While this is usually fine for a small group, 

it can be burdensome for larger paging groups since it ties up a lot of resources in the system. A paging 

group with 20 extensions is essentially a 20-person conference call. Due to the increased demand on CPU 

performance, this unicast paging is appropriate for paging groups consisting of no more than ten extensions. 

Figure 11-1. Unicast Paging 

Unicast Paging 

192.168.0.5 192.168.0.2 192.168.5.68 

588 589 590 


158 

Multicast Paging 


A multicast page is essentially a predefined broadcast address that phones or SIP user agents are programmed 

to listen to. Each phone can be configured to listen to as many as 10 different multicast IP addresses. 

The advantage of a multicast page is that only one SIP session is established between the originator and the 

system (Figure 11-2). Subsequently, the system sends only one RTP stream to the multicast group on which 

the phones are listening. If 100 phones are listening on a multicast page, it is equivalent to a single phone 

call, instead of a 100-party conference call. This makes multicast paging suitable for groups that number in 

the thousands. 

589 

Figure 11-2. Multicast Paging 

Multicast Paging 

588 

192.168.0.2 

192.168.0.32 

224.0.0.1:5000 

192.168.0.5 590 590 

591 593 

192.168.0.33 

Setting up a Paging Account 

596 

192.168.0.37 

192.168.0.34 192.168.0.35 

594 

192.168.0.36 

This section provides instructions for creating and configuring a paging account on the snom ONE. 

1. From the Accounts tab, click Create and select Paging from the dropdown: 


Chapter 11: Paging 159 



515/9781234567 







515 516 517 


4. Once the account has been created, click List and open the account: 

Configure the account as detailed below: 



■ Streaming Mode: This setting allows you to determine how your paging group will be set up: 

— Unicast (SIP): When using unicast mode, you’ll need to provide a list of destination numbers 

for your paging group in the next setting (Destination). With unicast paging, the system 

initiates a call to all extensions listed, so be sure to limit your paging group to no more than 

10 extensions. Unicast paging is not limited to the local area network, so all extensions that 

are connected to the system will be paged regardless of location. 

— Multicast: Multicast mode allows you to designate a single IP address as your paging group 

and create paging groups that encompass entire organizations without taxing bandwidth (the 

end systems are unaware who the members of the multicast group are and do not send SIP 

signaling to the individual paging devices). When you page the group using this method, 


160 


all devices configured to listen on the same multicast IP address will go into paging mode as 

soon as they receive RTP traffic on that port. 

Not all phones support multicast paging. Phones that do have this support must be configured 

through the phone’s web interface before they can be part of a multicast paging group 

(see “Assigning Multicast IP Addresses to IP Phones” on the next page). 

■ Destination: When using unicast mode, enter a group of extension numbers. When using multicast 

mode, enter an IP address. 

Note: Multicast addresses range from 224.0.0.0 to 239.255.255.255. 

■ Source: List the extensions that are allowed to page this group. Enter a star (*) to give access to all 

extensions. 

■ Display Name: The display name will be used to identify the source of the call. SIP phones will 

display this text in the display area. The Accounts page will list the display name next to the account 

number: 

■ Codec: The system comes with G.711 and is free. Other codecs, such as G.729, require a fee due 

to the royalties associated with them. 

■ Record message: The Record message feature can operate in two different modes: 

— Live playback: This mode allows you to send a page in real time. With this mode, you pick 

up the phone, dial the paging account number, and start speaking/paging. All interested parties 

will hear the page. 

— Play recorded message: This mode has a store-and-forward mechanism and requires that you 

record the message before sending it to the interested parties. With this mode, you pick up 

the phone, dial the paging account number, and start speaking. The system will record the 

message once you hang up and will send it to the interested parties. 

■ Permissions to monitor this account: Enter the extensions that are permitted to monitor this account. 


Assigning Multicast IP Addresses to IP Phones 

Chapter 11: Paging 161 

Before a phone can be part of a multicast paging group, a phone must have a multicast IP address that 

matches the snom ONE multicast IP address. This requires configuring the multicast settings for each 

phone. 

1. Go into the phone’s web GUI by entering the phone’s IP address into a web browser. 

The following screen will appear: 

2. Click Advanced at the left side of the screen. 

3. Select SIP/RTP as shown below: 

4. Scroll down to Multicast at the bottom of the page. 

5. Enable Multicast Support. 

6. Enter the IP address that was entered as the destination when configuring the multicast paging account 

in the web interface. 


The phone can now pick up the RTP stream and begin playing it. 


162 



CONFERENCING 

Chapter 

12 

A conference room or conference server is normally an expensive add-on with traditional phone systems. 

With the snom ONE telephone system, you can have your own conference rooms within your own system. 

Several types of conferencing can be used on the system: scheduled and ad hoc. The scheduled conference 

allows users to plan in advance for the conference and the ad hoc conference, as its name suggests, allows users 

to pull together a conference at a moment’s notice. 

Extensions 

586 

587 

588 

PBX 

Conference Room 

Ext. 500 

Remote Users 

589 590 

Figure 12-1. Conference Account Flow 

9781234567 

9787654321 

PSTN 

Internet 

ITSP 

Moderator settings can be configured from the web interface, and conferencing star codes can be used during 

the conference. Several conferences can be going on simultaneously in various conference rooms. All types 

of users (remote, snom ONE, PSTN) can call into both types of conferences. 

Table 12-1 provides an overview of the conference features. 

Table 12-1. Conference Account Features 


Email The system sends an email notification contain- The conference organizer can notify 

notification ing the time, date, conference room number, participants easily, and users are noti- 

and purpose of the conference. 

fied in advance. 

Outlook 

integration 

Ad hoc 

conferences 

Since email notification contains a calendar 

event, it can be read by Microsoft Outlook. 

Meetings can be established at a moment’s notice 

by anyone inside or outside the office. 

Users are unlikely to miss the conference. 

A manager can place a page announcing 

the details of a meeting, and everyone 

can call in. 


164 



PIN codes PIN codes restrict admission to the conference. 

They are generated by the system and are 

included in the notification email that is sent to 

each invitee. 

Moderator 

control 

Conference 

recording 

Backdoor 

access 

Multiple 

conference 

rooms 

The conference leader receives an exclusive PIN 

code. 

Conference leaders can record conferences. 

When participants enter the conference, they 

are informed that it will be recorded. 

Rather than using valuable telephone lines for 

external conference attendees, inexpensive SIP 

trunks can be used. 

The system allows as many conference rooms as 

a license will permit. Each conference room has 

a unique number. 

Scheduled Conferences 

PIN codes add security. Only invited 

people can attend. 

Moderator control allows the conference 

leader to receive the list of participants 

via email, execute star codes 

during the conference, and clear the 

conference when its over. 

The conference leader can review the 

recording and email it. Everyone can 

have a personal copy of the conference. 

SIP trunks significantly reduce the load 

on the main telephone lines and keep 

them open for customer use. 

Conference rooms can be accessed 

from inside or outside the office. Different 

departments can conduct conferences 

at the same time. 

The scheduled conference allows the conference leader to plan for a conference in advance. Users are informed 

via email of the time, date, and access code. The access code ensures that no unauthorized users can 

join the conference. Before users can schedule conferences, the administrator must set up at least one conference 

account. After the account has been created, users can schedule conferences by going to Lists > Conferences 

(available only from the user login). This section shows the administrator how to create the account 

itself. 

Creating a Scheduled Conference Account 

To create the conference account: 

1. Click Accounts > Create. 

2. Choose Conference from the dropdown. 



667/9781234567 




can clutter it up. If this occurs, you can hide aliased accounts under the accounts page at the domain 



667 668 669 



Chapter 12: Conferencing 165 

5. When the page reloads, click List to view the accounts, then click the edit icon to open the account. 

Most of the settings are self-explanatory, but keep the following in mind when populating the fields: 

■ Conference Name: The name that is assigned to the conference will appear in the Room dropdown 

that is displayed on the page when the user goes to organize a conference. Choose a name 

that will help users differentiate one conference room from the next: 

The conference name will also be displayed next to the account number on the Accounts page: 

■ Mode: Be sure to choose Scheduled conference. 

■ Before entering conference: The setting determines what occurs when the participant enters 

and exits the conference. 


166 


Scheduling a New Conference 

Once a conference account has been created, users can create conferences from their own accounts: 

1. Click Conferences from the Lists tab, as shown below: 

2. Populate the form using the following information: 

■ Name: This field allows participants to know the subject of the conference. 

■ Date: Use MM/DD/YYYY (month/day/year) or MM/DD (month/day) format. Leaving this field 

blank will automatically schedule your conference for the day you populate the form. 

■ Start and end times: Use HH:MM (hours/minutes) format. 

Note: The system will use the selected time zone of the extension that is logged in. The start and 

end times must be in the same format. 

■ Participants: Enter extension numbers and/or email addresses. (The user’s domain must be set 

up for sending out emails.) When listing multiple email addresses, use a semicolons as a space 

separator. 

The email contains a calendar event, which can be read by most scheduling tools. The tool will 

pop up a conference reminder with the relevant conference information. 

■ Record the conference: The recording will be placed into the user’s mailbox on the web interface 

(Lists > Mailbox). 

■ Room: Choose a conference room. 

■ Moderator options: These settings are discussed on page 167. 


The new conference will be displayed under Conferences organized, listing the moderator and participant 

PINs. Participants will receive an email that includes conference details, such as date, time, 

and PIN number. 


Modifying Scheduled Conferences 


Once a conference has been created, it cannot be modified; it can only be deleted (click the delete icon in 

the conference list). Conferences are automatically deleted 24 hours after the conference end time; however, 

participants who want to continue talking before the conference is deleted can do so as long as the moderator 

specified Continue the conference as an exiting preference. 

Moderating Scheduled Conferences 

Web interface settings and star codes can be used to control a scheduled conference. The web interface settings 

are located in Moderator Options in the conference account: 

These settings are described below: 

■ Entering conference 

— Start the conference without moderator: Participants will be able to communicate with each 

other upon entering the conference. 

— Mute all participants until moderator enters the conference: Participants won’t be able to 

communicate with each other until the moderator calls into the conference. 

— Mute all participants until moderator unmutes the conference: Participants won’t be able to 

communicate with each other until the moderator unmutes the conference (*8). 

■ Exiting conference 

— Continue the conference: Participants will be able to communicate with one another after 

the moderator has exited the conference. (The conference will not be deleted until after 24 

hours.) 

— Mute all participants on exit: Once the moderator exits, participants’ phones will be muted. 

— Hang up the conference on exit: Once the moderator exits, the system will disconnect all 

participants from the conference call. 


168 


Table 12-2. Conference Star Codes 

Star Code Description 

*0 Send participant list 

*1 Mute your device 

*2 Unmute your device 

*7 Mute other participants 1 

*8 Unmute other participants 1 

*9 Hang up the conference 1 

1. Requires moderator privilege. 

Note: As with the other conference star codes, *0 (send participant list) must be executed during the conference. 

A participants list will immediately be sent to the moderator via email. 

Ad Hoc Conferences 

The ad hoc conference account allows users to hold conferences at a moment’s notice. No planning is required, 

and no emails are sent out. This method of conferencing is handled through the system and produces 

a high-quality three-way conference. Although this method offers an especially simple way to hold a conference, 

unauthorized users will be able to access the conference room if an access code requirement has not 

been set on the account. 

Creating an Ad Hoc Conference Account 

1. From the Accounts tab, click Create and select Conference from the dropdown. 

2. Enter the extension number and a name that will be associated with the account, then click Create. 

3. Once the account has been created, click List to open the account and configure it as detailed below: 



■ Account Number: You can change the name of your conference or the extension number of 

your account at any time. 

■ ANI: The conference account uses an ANI (automatic number identification) when sending 

calls to an external number. 

■ Conference Name: It is recommended that you choose an understandable name for the conference, 

e.g., Company A conference mixer (2106171234 Ext 654), as this text will 

be used in conference invitation emails. 

■ Mode: Select Ad-hoc conference. 

■ IVR Language: From the dropdown list, choose the language that will be used for the voice 

responses. 

■ Moderator Access Code: This field can be left empty; however, the conference cannot be moderated 

without an access code (see “Moderating an Ad Hoc Conference” below). 

■ Participant Access Code: When no moderator access code has been set, this field can be left 

empty; otherwise, a code is required (users will be unable to access the conference without a 

code if the moderator has one). 

■ Permissions to monitor this account: Enter the extension numbers of the individuals who are 

permitted to monitor this account. 


Calling into an Ad Hoc Conference 

To hold an ad hoc conference, users need to know the extension number that has been reserved for ad hoc 

conferencing. If the designated extension is 665, then all parties who enter 665 from their extension phones 

will be joined to the same call. Participants will be required to enter an access code if one has been set; otherwise, 

they will be joined directly into the conference room. Any number of participants can join the call. 

(Participants can also be blind-transferred into the conference.) 

Moderating an Ad Hoc Conference 

Moderators can use the following star codes to dynamically control an ad hoc conference. If no moderator 

access code has been set, these star codes will not function. 

Table 12-3. Conference Star Codes 

Star Code Description 

*0 Send the participants list 

*1 Mute moderator 

*2 Unmute moderator 

*7 Mute all 

*8 Unmute all 

*9 Hang up the conference 

Note: As with other conference star codes, *0 (send participant list) must be executed during the conference. 

An email listing the participants will immediately be sent to the moderator. 


170 



IVR NODE 

Chapter 

13 

At the most basic level, the interactive voice response (IVR) system allows callers to reach their destination 

by making touch-tone entries through the phone keypad, rather than going through the operator. An 

IVR node can also be configured for complex tasks so that callers can enter account numbers and pay bills 

through an external application server. 

Note: The IVR node and auto attendant are two distinct accounts. Auto attendants with basic functionality 

(routing callers from point A to B) do not need an IVR node and can use the built-in IVR that is available 

through the auto attendant account (see Chapter 8 for more information). Table 13-1 lists the IVR node 

features: 

Table 13-1. IVR Node Features 


Caller Callers are required to enter their Caller verification expedites service to legitimate 

verification account number when reaching an callers and prevents undesired callers from wast- 

IVR. 

ing employees' time. 

Password Callers are required to enter their Password call-routing provides an automated 

call routing password before getting routed to means of controlling the flow of information to 

the correct destination. 

the appropriate contacts. 

Auto fax Upon detecting a calling signal A dedicated fax line/number is no longer needed 

detection from a fax, the system answers the 

call and sends it to your fax machine. 

to receive faxes. 

Response-based Response-based routing uses caller By developing a series of questions to which the 

routing input to route a call to the person caller can respond, the system can effectively 

or information best suited to assist. route the caller. Response-based routing saves 

The system asks the caller a question, 

and the caller responds by 

entering a digit. 

time and the expense of an interviewer. 

Expedited Incoming calls that match a num- Employees working outside the office can be 

caller-ID ber currently on an IVR list are au- connected to the correct department by dialing 

routing tomatically routed to a pre-defined the main number. Preferred customers can be 

destination. 

routed based on their caller-ID. Unwanted callers 

can be routed to “dead ends” or busy signals 

to discourage them from calling. 


172 



Announcement 

before ringing 

The system greets every caller with 

a uniform message prior to connecting 

the call: 

Example: “Thanks for calling XYZ 

Sales. Please have your account 

code ready.” 

“Thanks for calling Pizza King. Be 

sure to ask about today’s special.” 

How the IVR Node Works 

Information can be delivered before the caller is 

connected. This prepares the caller, expedites call 

flow, and makes your initial greeting count. 

When a caller reaches an IVR node, the system plays back the prompt that has been set for that IVR node. 

Once the caller answers the system by pressing the required digit(s), the system will continue processing the 

call. This first IVR node is typically used as an entry door into the system, and the nodes that follow are 

used to decide how to proceed next with the caller. For example, the IVR node can ask the customer for account 

information then ask an external application server where the customer should go from there. Based 

on the answer from the external application server, the customer is routed to the appropriate waiting queue. 

Incoming Call 

DTMFs are 

captured. 

6512345 

IVR Node 

Technical 

Support 

External Server 

SOAP 

processes 

the input. 

Is contract valid? 

Yes 

Call goes to 

Support ACD 

Figure 13-1. IVR Node—User Enters Contract Information 

Play Greeting 

“Please enter 

contract number.” 

No, expired 

Call goes to 

Sales ACD 


Chapter 13: IVR Node 173 

An IVR node can also be used to define dialog that will be used to process calls. In this case, each node plays 

one prompt and asks a specific question to the caller. The first prompt may ask the caller what language is 

preferred and then dispatch the caller into two different node systems for two different languages. This IVR 

node mechanism is very flexible and allows you to either process the input internally in the system or use an 

external application server to decide where to go next. The IVR node collects the user input according to the 

list of ERE expressions that was entered in the node’s match list. In the simplest case, a static routing is sufficient 

without consulting the external server, but it depends on the user input. 

Setting up an IVR Node Account 

This section provides instructions for creating and configuring an IVR node on the snom ONE. 

1. From the Accounts tab, click Create and select IVR Node from the dropdown: 



411/9781234567 




can clutter it up. If this occurs, you can hide aliased accounts under the accounts page at the domain 



411 412 413 


4. Once the page reloads, click List. 

5. Click the account to open it. 

6. Configure the account as described in the following sections: 


174 

Identity 


■ Account Number(s): This field takes an extension number(s) and/or DID number(s). The number 

of DIDs that can be entered into this field is unlimited. 

■ Name: This field allows you to create an alias so that you can quickly distinguish the IVR nodes 

from one another. 

Settings 

■ WAV File: An external recorded announcement can be uploaded in wave format (8-kHz Mono, 

16-bit files). Announcements can also be recorded by dialing *98888, where 888 is the IVR node 

account. 

■ Dial Plan for outbound calls: Choose the dial plan that you would like the system to use when 

routing calls to an outside provider. 

■ ANI: The IVR node uses an ANI (automatic number identification) when sending calls to an 

external number. 

■ Send daily CDR report to: The call data record (CDR) lists all calls that come into the IVR node. 

The report is sent at midnight each night to the email address specified here. 

■ DTMF Match List: This field allows you to enter a list of match patterns that will be used by the 

system to direct a call once a user has pressed a number from the telephone keypad. You can also 

use pre-coded IVR nodes (see page 129 for information on direct destinations). 

When entering DTMF match patterns, keep the following criteria in mind: 

— A space is required between list elements. 

— Each pattern must contain two fields separated by a character that does not occur anywhere 

else in the string, for example “!”. 

— The first field must contain the extended regular expression, and the second field must contain 

the destination. 



Example: To specify that the caller should be directed to extension 500 after pressing 0, enter 

!0!500! into the DTMF field. To specify that the system should wait until a user has entered 

three digits before attempting to redirect the call, enter !^([0-9]{3})$!\1!. 

■ From-based routing match list: This field requires syntax similar to the DTMF field. 

■ To-based routing match list: This field is used when the snom ONE opens the IVR node. Use the 

same format as in DTMF Match List above. If there is a match with the From or the To header 

of the call, then the IVR node immediately switches the destination without playing the WAV file. 

This allows you to implement flexible routing schemes. 

■ SOAP URL: If no SOAP URL is specified in the account, the system will assume that the matching 

pattern is the name of the account to switch to. If the SOAP URL is available, it will pass the 

destination to the application to further determine what to do with it. For more information on 

SOAP, see page 176). 

■ Accounts that may record a message: Enter the extensions that are allowed to record welcome 

messages. Use a star (*) to indicate all users. 

■ Timeout (secs): The timeout specified in this field starts either when the system sends the call to 

the IVR node or when the user enters a digit. 


The destination may be any dialable number. If the number requires a dial plan, the default 

dial plan of the domain will be used. If the destination field contains a single dash (“-”) and the 

pattern matches, the system will disconnect the call. 

Recording Messages 

Messages can be recorded using the Record Message (*98) star code, or they can be uploaded through the 

web interface. Both methods are detailed in the following sections. Users must have permission to record 

greetings before they can record greetings for an IVR node using *98. The permission is given in the IVR 

account itself. 

Record the Message Directly 

To record the message: 

1. From the telephone keypad, enter *98. 

2. Enter the IVR Node account number. 

3. Enter *1 (“1” is the greeting number). 


5* 

9 

8 1 2 6 * 1 

In the example shown above, the IVR node account is “126,” and the greeting that is being recorded 

is greeting 1. You may record up to nine greetings. The standard greeting is recognized by the system 

as greeting 0 and can be recorded by entering *98. Use the standard greeting if you want to use the 

same greeting for all hours of the day and night. 


176 

Upload a WAV File 


WAV files allow you to record and edit your message before loading it into the system. They also allow you 

to use professional IVR dialogs. From the IVR Node settings: 

1. Click the Browse button and upload the file (WAV files should be 8 kHz Mono, 16 bit). 


Linking an External Application Server to an IVR Node 

SOAP (Simple Object Access Protocol) is an XML message-based protocol specification used to allow applications 

running on a decentralized, distributed environment to exchange information. It relies on Extensible 

Markup Language (XML) as its message format, and usually relies on other application layer protocols, such 

as RPT and HTTP, for message negotiation and transmission. 

Sample SOAP Message 

Following is a sample data interchange using the HTTP/SOAP protocol. (The indentation shown is for illustration 

purposes only.) 

Request 

POST /ivr.xml HTTP/1.1 

Host: pbx.com 

SOAPAction: IvrInput 

Content-Type: application/xml 

Content-Length: 123 


xmlns:sns=” http://www.snomone.com/soap/pbx”> 

 

 

3525234@snomone.com 

123 

Fred Feuerstein <sip:ff@test.com> 

Tom Test <sip:tt@test.com> 

 

 

 

In the snomone namespace, the record CDR indicates that a CDR shall be transmitted. The CDR may have 

the following attributes: 

■ CallID: This attribute contains the call-ID of the call and makes it possible to put the IVR input 

into a session context. It is also used as an identifier in the response to the request. 

■ Output: The digits that the user has entered into the IVR node. 

■ The To and From fields indicate the To and From headers of the call. 

Response 

The response on the file tells what should be done with the input. An example return could look like the following: 

HTTP/1.1 200 Ok 

Content-Type: application/xml 


 

 

 

3525234@snomone.com 

123 

 

 

 

The response has the following attributes: 

■ CallID: This attribute contains the caller-ID of the call. It is used to identify the affected call. 

■ Destination: This string indicates which account to switch to. 

Example PHP File 

This example shows how to process the CDR on an Apache web server using the PHP extension. This example 

just extracts the data and writes it into a plain file. 

178 


function start_element($parser, $name, $attrs) { 

global $elem; 

$elem=$name; 

} 

function end_element($parser, $name) { 

} 

function xml_data($parser, $data) { 

global $elem, $from_user, $to_user, $from, $to, $duration, $start, 

$callid, $output; 

if($elem==”CALLID”) $callid .= $data; 

else if($elem==”OUTPUT”) $output .= $data; 

} 

$content = $HTTP_RAW_POST_DATA; 

$xml_parser = xml_parser_create(); 

xml_set_element_handler($xml_parser, “start_element”, “end_element”); 

xml_set_character_data_handler($xml_parser, “xml_data”); 

if(!xml_parse($xml_parser, $content, true)) { 

die(sprintf(“Get XML error parsing %s: %s at line %d”, 

htmlspecialchars($content), 

xml_error_string(xml_get_error_code($xml_parser)), 

xml_get_current_line_number($xml_parser))); 

} 

xml_parser_free($xml_parser); 

// Ok here some bogus logic what to do with the input: 

$result = “123”; 

$xml = ‘’; 

$xml .= ‘’; 

$xml .= $callid; 

$xml .= ‘’; 

$xml .= $result; 

$xml .= “\ 

r\n”; 

echo $xml; 

?> 


CALLING CARD ACCOUNT 

Chapter 

14 

The calling card account allows callers to place outbound calls from the system without being logged into 

their extension. This is especially advantageous to the user who is traveling. To place an international call, 

users simply call into the system and enter their extension number and PIN code. The call will reflect the office’s 

caller-ID and will appear in the user’s call log. 

ext 231 

snom ONE 

The calling card account can be used in three different modes: 

■ DISA —The user calls into the account and authenticates himself by entering his extension number 

and the associated PIN code. If the user’s company has an 0800 number, then this is a simple away 

to reduce costs for international calls. 

■ Callback—The user calls into the system and requests a callback. Although this mode requires more 

steps to establish the call, it may reduce telephone costs even further, as the call to the system does 

not get connected and the caller is not charged for the call. When callback is initiated from the 

calling card account, rather than the Personal Virtual Assistant, the user will not be charged for the 

seconds/minutes that accumulate between the time the callback is initiated and the caller is connected 

to the destination. 

■ Pre-pay—The user calls into the auto attendant from an outside phone and enters his extension 

number and PIN. Once the system completes the authorization, the user enters the destination 

number. The minutes accumulate against the user’s Credit for outbound calls setting (see page 

107). 

The calling card account can also be used with an external database. Customers who buy a prepaid calling 

call can dial into a free-of-charge number that goes into the calling card account. The system uses the SOAP 

interface to communicate with an external application server which manages the calling cards and the remaining 

money. Once the call has ended, the system reports the call duration to the server. 


180 


Setting up a Calling Card Account 

This section provides instructions for creating and configuring a calling card account: 

1. From the Accounts tab, click Create and select Calling Card: 



103/9781234567 




can clutter it up. If this occurs, you can hide the aliased accounts under the accounts page at the 

domain level (see page 58). 


103 104 105 



sections. 


Identity 

Chapter 14: Calling Card Account 181 



Behavior 

■ Dial Plan: Choose the dial plan that you would 

like the system to use when routing calls to an 

outside provider. If the system uses the SOAP 

interface for placing outbound calls, it needs to 

know which dial plan to use. If the system uses 

the internal database, it uses the dial plan of the 

selected extension. 

■ ANI: When a dial plan is used, the system also checks the presence of an ANI (automatic number 

identification) number and, if set, the system will use it. 

■ Send daily CDR report to: The call data record (CDR) lists all calls originating from the calling 

card account. The report is sent at midnight each night to the email address specified here. 

■ Callback: This feature allows the account to be used in callback mode. After dialing into the system, 

the user is prompted to enter an extension number and PIN code. The caller may leave a callback 

number; otherwise, the system will use the caller-ID. Once the system calls back, a destination 

number will be requested. The callback feature allows users to make free outbound calls from a cell 

phone as long as the cellular provider offers free incoming calls and the user has an unlimited call 

plan with one of the system’s trunks. 

Important: Before users can use the calling card in pre-paid mode, Callback must be set to off 

(refer to Chapter 15 for information on the pre-pay feature). 

■ Caller-ID: Choose whether to show or block the caller-ID. If you choose to show the caller-ID 

and a call is started on behalf of a known extension, the system will use the caller-ID associated 

with that extension. If the SOAP interface is being used, the system will display the caller-ID of the 

calling card account. 

■ Allow calling accounts on the PBX: When this setting is enabled, the system will also consider 

calling local accounts. This is often useful in corporate environment, where callers can use 0800 

numbers to call internal destinations. 

■ Entering telephone numbers: The system needs to know how to interpret a user’s input. Choose 

from the options shown in the dropdown (a description of each option is listed below): 

— User must press pound: The user will be required to press the pound key after entering the 

telephone number. 

— System dials automatically (NANPA): The system will interpret telephone numbers according 

to the NANPA scheme (North American Numbering Plan Administration), which 

means that international numbers will start with 011 and numbers that have 10 digits will be 

written in the (xxx)xxx-xxxx scheme (see Table 3-2 for more information on NANPA). 


182 


— System dials automatically (ROW): The system will interpret numbers according to a “rest of 

world” scheme, which means that international numbers start with 00 and national numbers 

start with 0 (see Table 3-2 for more information on ROW). 

■ Read out balance when lower than (e.g., “1.00”): The system will let the user know when the 

card balance is lower than the amount that is indicated in this field. 



PRE-PAY FEATURE 

Chapter 

15 

The pre-pay feature allows the administrator to create customized rate tables for multiple pricing strategies. 

Calls are calculated based on call duration, destination, and the rates provided in the tables. The pre-pay feature 

can be configured so that users can access it using the extension account, the Personal Virtual Assistant 

(available from the cell phone), or the calling card account (PIN access). 

586 

Personal 

Virtual 

Assistant 

Calling Card 

Account 

Greece 

Germany 

Cuba 

USA 

RATES 

Users do not need a physical phone connection in order to take advantage of the prepay feature. This feature 

can also be accessed through a calling card account (see Chapter 14). 

Setting up the Pre-Pay 

The two important components of the pre-pay are the rates table and an assigned dollar amount on the user’s 

extension: 

■ Rates table—A rates table, CSV (comma separated value) formatted data, is an essential component 

to the pre-paid feature. It can include any number of entries and must be entered into the trunk. 

■ Dollar amount—A dollar amount must be assigned to the user’s extension account. 

Creating the Rates Table 

1. Create a trunk that will be used specifically for the pre-pay feature. 


184 


2. Click List to view a list of the trunks 

3. Click the Edit icon for the trunk you just created: 

4. Click Rates. 

A sample rates table is displayed: 

The rates table follows standard CSV format. All fields are separated by commas: 

Greece,01130,0.10 

Germany,01149,0.15 

Cuba,011527,0.45,55,1.00 

USA,1,0.05 

Rest of World,*,0.05 

Following is a tabular representation of this information. As the table shows, different telephone area 

codes and exchanges will be charged at different rates. 

Name of Entry Prefix** Cost per Unit 

Duration of Unit 

(seconds)* 

Connection 

Fee 

Greece 01130 0.10 

Germany 01149 0.15 

Cuba 011527 45 55 1.00 

USA 1 0.05 

Rest of World * 0.05 

* The Duration of Unit default is 60 seconds. 

** When establishing the prefix, the following settings must be considered: Country Code and Area 

Code settings (page 53) and Rewrite global numbers (page 85). 


5. Enter your own calling rates in the space under the sample rates table. 

Chapter 15: Pre-Pay Feature 185 

Important: In the event a blank field sits between two other fields (e.g., you’re requiring a connection 

fee but will not be changing the Duration of Unit default, as shown in the table below), you will 

need to make provision for this empty space in your CSV syntax. CSV syntax in this case would be 

Greece,01130,0.10,,1.00 (notice the double commas). 

Cost per Duration of Unit Connection 

Name of Entry Prefix Unit (seconds)* Fee 

Greece 01130 0.10 1.00 

All other trunk settings are detailed in Chapter 4. 

Assigning a Dollar Amount to an Extension 

Before a user can begin using the pre-pay calling card account, a dollar amount must be specified in the 

extension account: 

1. Open the user’s extension, and click the Registrations tab. 

2. In the Credit for outbound calls setting, enter a dollar amount. 

Important: If this field is left blank, the user will be allowed unlimited calling. 

Note: Users can execute the Show Account Balance (*61) star code to determine the number of minutes 

that are left on the pre-paid account. 

Methods of Access 

The pre-pay feature can be accessed via the extension, the Personal Virtual Assistant, or the calling card. This 

section details each method. 

Extension 

This method does not require a calling card. Once the rates table has been added to the trunk and the extension 

has been assigned a dollar amount, all calls made from the extension will be calculated according to the 

rates table assigned to the trunk. To determine the number of minutes remaining on the account, the user 

can execute the Show Account Balance (*61) feature. 


186 


Personal Virtual Assistant 

With this method, users can call into the Personal Virtual Assistant from their cell phone to access their prepaid 

account (the cell phone must be configured to the user’s account). Instructions are as follows: 

1. Call into the main number from the cell phone. Because the cell phone is recognized by the system, 

the user is presented with the Personal Virtual Assistant. 

2. From the Personal Virtual Assistant, press 1 (to make an outbound call). 

3. Enter the destination number. 

As with the previous method, the call will be calculated according to the rates table assigned to the 

trunk. To determine the number of minutes remaining on the account, the user can execute the 

Show Account Balance (*61) feature. 

The Personal Virtual Assistant is discussed page 311. 

Calling Card Account (PIN Access) 

This method requires a calling card account. (Once you have created the account, be sure to deactivate the 

Callback setting and specify an amount for the Read out balance setting.) 

This method will require that the user enter a PIN number for each call that is made. 

Important: For security purposes, always require that your users enter a PIN. Do not try to configure the 

calling card account so that a PIN is not required. 

To use the calling card account: 

1. Call into the auto attendant from an outside phone. 

2. Enter the calling card account number. 

3. Enter the extension number. 

4. Enter the PIN of the extension. 


The call will be calculated according to the rates table assigned to the trunk. The system will read 

out the minutes remaining once the Read out balance threshold has been reached on the account. 

To get a readout of remaining minutes before the threshold has been reached, users can execute the 

Show Account Balance (*61) feature. 


EMAIL 

Chapter 

16 

E-mail support is one of the most useful and powerful features of the snom ONE telephone system for both 

the administrator and the user. The system can be configured to notify the administrator (or anyone else in 

charge) when calls get rejected, when the system disconnects a call, or even when someone dials an emergency 

number. Many other critical system events can also be included in this alert. 

Users, too, can be notified of important events on their end. They can receive emails when they miss a call, at 

the end of every call, or whenever a call is rejected because of a blacklisted number. Users can also configure 

their account so that voicemails are sent as email attachments. This configuration enables them to check their 

voicemails at any time from any location in the world using an Internet connection. 

This chapter shows the administrator how to configure these settings, as well as how to configure the SMTP 

server on the snom ONE telephone system. 

Configuring the SMTP Server 

System-Wide SMTP Settings 

The snom ONE telephone system can be configured as an email client to send email messages to the users. 

To do this, the system needs a mail or an SMTP server (via an external SMTP provider) to send the 

messages. If you already have an SMTP server set up, you can configure it to the system using the following 

instructions. If you do not have one set up, you can set one up through Gmail for free or ask your email 

administrator. 

Note: SMTP settings can be configured by either the administrator or the domain administrator. 

1. Navigate to Admin > Email: 


188 


2. Populate the form using the following information: 

■ From Address: This field will be used by the system when it sends emails (it is the From address). 

You can use names like voicemail@your-company.com or pbxautomailer@ 

your-company.com. The name will appear in the From field of the email. 

■ Account: Provide the e-mail account for the SMTP server. Generally, this is the same as the 

From address. 

■ Password: This is the password for the e-mail account. 

■ Password(repeat): Confirmation for the password. 

■ SMTP Server: This is your SMTP outgoing mail server. (The system can send email messages 

by emulating an email client; however, it needs a mail or SMTP server to send the messages.) 

(Gmail’s SMTP is smtp.gmail.com.) 

■ Enable TLS: This setting indicates whether you use TLS to connect to the mail server. Default 

is Yes. 

■ Use same settings for all domains: If you leave this setting set to Yes, it will pass SMTP settings 

onto your domain. 

Domain-Specific SMTP Settings 

Domain administrators can configure the system’s email settings using the following instructions: 

1. Navigate to Domains. 

2. Select the domain, then click Settings. 

3. Scroll down to Email Settings. 

4. Populate the form using the information from page 188. 

Adding Email Accounts to the System 

Note: First create your email accounts using Gmail, Yahoo, or another webmail service provider. Once you 

have created your email accounts, you can add them to your extensions by clicking the Email link in each 

extension account (Figure 16-1) and then entering the email address into the Email Address field. When 

entering multiple email addresses, use a semicolon as a field separator. 

Figure 16-1. User Email Settings 


Chapter 16: Email 189 

Note: Figure 16-1 shows the standard email settings that are available to the user, but administrators can 

prevent the user from seeing the Send a mailbox message and After sending a message settings (see the 

email control settings on page 48). 

■ Send a mailbox message by email: Users have the following three options: 

— Do not send an email: No email will be sent. Only the extension will receive messages. 

— Send emails without attachments: Email notifications will be sent to the user whenever a 

voicemail message is received. This provides somewhat of a visual voicemail box. The email 

will show the name and number of the caller and the date and time that the call arrived. To 

retrieve the voicemail itself, the user will need to call into the voicemail. The benefit of this 

mode is that emails that are relatively short can easily be read using mobile devices that support 

reading emails. 

— Send message as attachment: Actual voicemails will be attached to the email notifications. 

This option is suitable for users who use email throughout the day and are using a personal 

computer for processing email. 

Note: Users can also use the *95/*96 star code from the phone to request that voicemail be 

sent as email attachments. 

* 9 5 

* 9 6 

Send voicemail as email (activate) 

Send voicemail as email (deactivate) 

■ After sending a message: The user must decide what to do with a voicemail message after it has 

been emailed: 

— Keep message as new message: The message will remain in the user mailbox as a new message. 

This option has the potential danger of eventually overfilling the user's voice mailbox. 

— Mark message as read: The message will remain in the user's mailbox, but the system will 

drop the oldest message to make room for a new message if the mailbox becomes full. The 

disadvantage of this mode is that the message waiting indicator will not alert the user of new 

messages. 

— Delete the message: The message will be deleted after the email has been sent. This keeps the 

user's mailbox clean, but the system does not store voicemail messages permanently. The user 

will have to listen to the messages from the email client. 

■ Send email on missed calls: When this setting has been activated, the system sends the user an 

email for each missed call (see below). Calls to a hunt group or agent group are not included. 


190 


■ Send email at the end of every call: When this setting is activated, the system sends the user an 

email for all calls (see below). This feature is useful for keeping track of a group sales extension, a 

telemarketing campaign, or calls made to a new employee’s extension. (When both “missed calls” 

and “all calls” have been activated, the user will receive two emails for each missed call.) 

Note: Users can also use the Request Call Details (*63) star code to request the call details of a particular 

call. Details include the caller-ID, the duration of the call, and the time the call was made. 

This feature is handy and prevents users from having to ask the caller for contact information. Users 

must: 

1. Dial *63 and press the confirm button on the telephone keypad. 

* 

6 3 

2. Retrieve the message in their email account. 

Note: The email sent by the system includes a link to the last number. When users click the 

link, the system will prompt them for a username and password (in the form user@domain). 

If the user’s browser supports saving the login information, the next time the user 

clicks such a link to dial a number, the call will immediately be initiated. 

This feature works only with user agents that support the REFER mechanism outside of 

existing dialogs. On some phones, users must press “Ok” or lift the handset in order to 

acknowledge the dialing of the number. The remote initiation of a call is a security-sensitive 

topic, as it might turn a user’s phone into a microphone. Therefore, user’s must authenticate 

themselves during the initiation of the call and may have to acknowledge the initiation of the 

call. 

■ Send email on status changes: The user will be notified of all status changes (e.g., changes to the 

DND status, the registration status, etc.). 

■ Send email when mailbox is full: The user will be notified when their mailbox is full. 

■ Send email if call was rejected because of blacklisted number: When users execute the *92 star 

code (Add to Blacklist) to block a particular caller (or if the caller is already tagged as a blacklisted 

caller in the user’s address book), an email will be sent to the user’s mailbox when that caller has 

attempted to call. 



Receiving Email Notifications from the System 

Types of Email Notifications 

The system can be configured to notify a list of administrator accounts when an important system event 

occurs. The types of email notifications are presented in Figure 16-2 and are located in Admin > Email > 

Messages. 

Figure 16-2. Types of Email Notifications 

Customizing Email Notifications 

Email notifications can be customized. To customize the messages, a familiarity with HTML is required. 

1. Navigate to Admin > Email > Text. 


192 


2. From the dropdown, make a selection. The HTML will be displayed: 

Embedded between the standard HTML tags (e.g., and ) 

is additional code (ssi htmvar password, lng text4, ssi var link, etc.) that is used 

by the system to translate various parts of the welcome message, e.g., the specific extension that has 

been created, the language that has been set for that extension, etc. If these settings are not important 

and you prefer to send a static email once an extension has been created, you can create a static message 

similar to the following: 

 

 

 

 

This email is sent by snom ONE 

 

.normalText { 

FONT-SIZE: 12px; FONT-FAMILY: Verdana; border-bottom: 10pt 

} 

.header2Text { 

FONT-WEIGHT: bold; FONT-SIZE: 18px; COLOR: #168099; FONT-FAMILY: Verdana 

} 

 

 

 

Hello new user. You have been added to the snom ONE 

telephone system. This email is sent automatically. Do not reply. 

 

 

 

CDRs to Email 

All types of CDRs can be exported to email. Depending upon which type of information you would like to 

receive, you will need to configure the respective setting on the snom ONE. 

Important: Before you can receive any type of CDR, the SMTP server settings must be configured (go to 

Admin > Email). 


CDRs for Trunk Activity 


If you want to receive a nightly email report on trunk activity, add your email address to the Midnight 

Events setting. 

1. Navigate to Domain. 

2. Click the domain. 

3. Click Settings. 

4. Scroll down to Midnight Events and enter your email address into the Send daily CDR report to 

setting. 

Note: When entering multiple email addresses into the field shown above, use semicolons between 

the addresses (e.g., Fred Feuerstein ; Carl Clements ). 

Note: Clicking the Try button will send the CDR immediately. This allows you to see if your settings 

work, rather than waiting until midnight. Keep in mind that this is just a link, and you may 

have to save your list before clicking the link. 

CDRs for Your Extension Only 

If you want to receive emails for your calls only, add your email address to your extension and enable the 

Send email record setting: 

1. Log on as a user. 

2. Click Settings > Email. 

3. Enter your email address into the Email Address field. 

4. Click Yes for Send email record for all calls. 


194 


CDRs for All Extensions on Your Domain 

If you want to receive emails for every call on every extension, add your email address to the CDR URL setting 

at the domain level. 

1. Click Domain, then choose the domain. 

2. Click Settings. 

3. Scroll down to CDR URL and enter your email address. 

Note: You must use the mailto syntax as shown below. 

CDRs for All Extensions on All Domains 

If you want to receive emails for every call on every extension for all domains, add your email address to the 

CDR URL setting on the Admin > Settings page. 

1. Click Admin > Settings. 

2. Scroll down to CDR URL and enter your email address. 

Note: You must use the mailto syntax as shown below. 

The following report shows a running list of the previous day’s calls: 

Figure 16-3. Sample CDR Report 

Call Recording to Email 

Two types of call recording are available on the system: system-initiated and user-initiated (also known as ad 

hoc). Only ad hoc recordings can be sent to email. With ad hoc recording, calls are recorded using either the 

Record button on the phone or by executing the *93/*94 star codes. See Chapter 17 for more information 

on call recording. 

Depending upon which phone model is being used, the Record button may be located in different places on 

the phone. On the snom 360, the Record button has a white circle on it. When the button is pressed, the 



phone display will show a solid black circle to indicate that the call is being recorded. The recorded call will 

be displayed in the extension user’s log under List > Mailbox. The user can click the audio icon to play the 

recording as shown below: 

Blacklist Activity to Email 

System-Initiated 

To configure the system to send an email for domain-level blacklisted calls, configure the domain as follows: 

1. Navigate to Domains. 

2. Select the domain, then click Settings. 

3. Scroll down to Midnight Events. 

4. Enter an email address into the space provided. 

5. Click Yes for the blacklisted option shown below: 

The email will be sent at midnight to the specified CDR email address. For a trial run of the way this feature 

works, click Try. 

An email will be sent to the email address you’ve indicated. 


196 



AUDIO AND GREETINGS 

Greetings 

Chapter 

17 

Greetings are prerecorded, customized messages that can be used with numerous account types. Users can 

create greetings for their personal voicemail account, and administrators can create greetings for auto attendant, 

agent group, and IVR node accounts. Greetings are created with the Record New Greeting (*98) star 

code and are stored in the recordings folder. Users need permission before they can record greetings for 

domain accounts. 

Personal Greetings 

Users can record up to five greetings for their personal extension accounts. Greetings are created using the 

*98 star code followed by the greeting number: 

5* 

5* 

5* 

5* 

5* 

9 

9 

9 

9 

9 

8 5* 1 

8 5* 2 

8 5* 3 

8 5* 4 

8 5* 5 

Naming Conventions 

Greeting 1 

Greeting 2 

Greeting 3 

Greeting 4 

Greeting 5 

Recorded greetings will appear in the recordings folder and are named according to the system’s syntax 

rules. Its naming system is based on the greeting number that was recorded and the account type for which 

the greeting was recorded (personal, auto attendant, agent group, etc.). The actual account number is never 

reflected in the file name of the greeting, as shown below (the numbers 23 and 24 are internal ID numbers 

assigned to two different extensions). 


198 


Activating a Personal Greeting 

Administrators: Once greetings have been recorded for an extension, administrators have the option to upload 

them from the Mailbox tab of the user’s account. 

Users: Once greetings have been recorded, a user can activate a greeting by using the following steps: 

1. Enter extension number to access voice mailbox. 

2. Press 9 (“select mailbox greeting”), or press * to skip to the main menu. 

3. Once you hear the greeting you would like to use, press the corresponding greeting number when 

the greeting has finished playing. 

To delete personal greetings, navigate to the location where the greetings are stored on the system, and delete 

them. 

Auto Attendant Greetings 

Users must have permission to record auto attendant greetings. This setting is located within the auto attendant 

account. The administrator must enter specific extension numbers into this field. An asterisk can be 

used to indicate that all users are allowed to record messages. 

Auto attendant greetings can be created using the Recording New Greeting (*98) star code or by creating 

them yourself. Up to five greetings can be recorded. 

Star code method: Any extension within the network that has permission to record auto attendant greetings 

can use this method for recording. The account number of the auto attendant must precede the greeting 

number. In the following example, 70 is the auto attendant. The greetings are automatically placed into the 

recordings directory. 

5* 

5* 

5* 

5* 

5* 

9 

9 

9 

9 

9 

8 

8 

8 

8 

8 

7 

7 

7 

7 

7 

0 

0 

0 

0 

0 

5* 

5* 

5* 

5* 

5* 

1 

2 

3 

4 

5 

Greeting 1 

Greeting 2 

Greeting 3 

Greeting 4 

Greeting 5 

Manual method: Only domain administrators can create greetings using the manual method, as access to the 

auto attendant account is required. This method requires that domain administrators create their own recordings 

and upload them through the IVR tab of the auto attendant account. Uploaded recordings should 

be 8 kHz mono, 16 bit files. 



Chapter 17: Audio and Greetings 199 

Auto attendant greetings (when created with *98) are named according to the system’s syntax rules. Its naming 

system is based on the greeting number that was recorded and the account type (personal, auto attendant, 

agent group, etc.) for which the greeting was recorded. The actual account number is never reflected in 

the file name of the greeting, as shown below (the number 11 is an internal ID number). 

Table 17-1. Naming Convention for Auto Attendant Greetings 


Auto Attendant *9870*1 att1_.wav att1_81.wav 





Note 1: Internal user IDs are generated by the system. 

Note 2: The number “70” is the account number of the auto attendant. 

The recordings will be stored in the recordings directory: 

Activating Auto Attendant Greetings 

Once auto attendant greetings have been recorded, administrators can activate them by creating service flags 

and uploading corresponding greetings. 


200 


To delete auto attendant greetings, navigate to the location where the greetings are stored on the system, and 

delete them. 

Agent Group Greetings 

Agent group greetings can be used to provide callers with information about a company or to provide additional 

options for the caller (see “Mapping Telephone Keypad Numbers with Extensions” on page 150). 

Users must have permission to record agent group greetings. This setting is located within the agent group 

account. The administrator must enter specific extension numbers into this field. An asterisk can be used to 

indicate that all users are allowed to record messages. 

Agent group greetings are created using the Recording New Greeting (*98) star code and are automatically 

placed into the recordings directory. Agent group greetings can be recorded from any extension within 

the network that has permission to record greetings. Ten greetings can be recorded (0 through 9). The account 

number of the agent group must precede the greeting number. In the example shown below, 73 is the 

number of the agent group. To listen to the greetings, dial the agent group number (e.g., 73). 

5* 

5* 

5* 

5* 

9 

9 

9 

9 

8 

8 

8 

8 

7 3 

7 3 

7 3 

7 

3 

5* 

5* 

5* 

5* 

0 

1 

2 

9 

Greeting 0 

Greeting 1 

Greeting 2 

Greeting 9 


Agent group greetings (when created with *98) are named according to the system’s syntax rules. Its naming 

system is based on the greeting number that was recorded and the account type (personal, auto attendant, 

agent group, etc.) for which the greeting was recorded. The actual account number is never reflected in the 

file name of the greeting (in the example shown below, the number 14 is an internal ID number). 

Table 17-2. Automatic Naming Conventions for *98 Greetings 


Agent Group *9873*0 acd0_.wav acd0_14.wav 











The recorded greetings are stored in the recordings directory: 

Activating Agent Group Greetings 


Once greetings have been recorded for the agent group, they will automatically play in sequence when a 

caller is waiting in the agent group queue. Once the messages have finished, they will restart. The gaps 

between the messages can be configured to span 5 to 180 seconds (gaps between announcements), during 

which music will be played. See Chapter 10 for more information on agent groups. 

To delete agent group greetings, navigate to the location where the greetings are stored on the system, and 

delete them. 

Recorded Phone Calls 

One minute of call recording consumes approximately 100K of disk space. Using this estimate, Table 17-3 

details the disk space requirements for a 10-person and 100-person call center. This does not take into account 

the compression feature, which could potentially increase that number by a factor of 8. (The CPU 

processor is compromised during compression and the quality of recordings is reduced, so turn on compression 

only if your disk space is limited and you have a good CPU.) 

Table 17-3. Recordings and Disk Space Requirements 

Recorded Time 

Per Person (Mb) 

Disk Space Requirements 

10-Person 

Call Center (Mb) 

100-Person 

Call Center (Gb) 

15 minutes (0.25 hour) 1.5 15 1.5 

30 minutes (0.5 hour) 3 30 3 

60 minutes (1 hour) 6 60 6 

120 minutes (2 hours) 12 120 12 

240 minutes (4 hours) 24 240 24 

480 minutes (8 hours) 48 480 48 

Two types of recordings can be produced on the system: user-initiated and system-initiated. These are discussed 

below. 

User-Initiated Recordings 

User-initiated recordings are generated when the user either presses the Record button on an IP phone (e.g., 

snom 320 and 360) or dials the Record (*93) star code. The recordings will be placed into the record- 


202 


ings folder. Anyone with access to the computer running snom ONE can listen to, copy, save, and forward 

the conversations. 

The recordings will be sent to the user’s email address. Users can listen to them by clicking the paper clip. 

System-Initiated Recordings 

System-initiated recordings include calls that are automatically recorded by the system. A domain can be 

configured to automatically record the following types of calls: 

This list can be accessed in Domains > Settings (see page 60). The settings that are made there will serve 

as the recording defaults for the indicated accounts across the domain. If you have multiple hunt groups 

and/or agent groups on the system and would like to override the default recording settings for some of 

those accounts, you can use the Record incoming calls to hunt groups and Record incoming calls to 

agent groups settings in the account itself. 

File System 

The file location and file name of system-initiated recordings are determined by the Record Location setting 

(page 27). The Record Location string is a two-part string. The first portion of the string determines the 

location of the recording, and the second determines the name of the recording. 

Storage Location 

File Name 

$r/$o/$a/$d-$t-$i-$n.wav 



Using the string shown above (which is the system default), the directory location would be snomONE/recordings/pbx.company.com/73, 

and the file name could be 20100907-165929-i-50.wav. 

There’s a lot of flexibility with regard to where the system can place recordings and the file names that it assigns. 

Following are the variables that can be used in the Record Location field: 

Table 17-4. Syntax for Recording to File 

Syntax Definition Description 

$r Recordings The location (e.g., the recordings directory) 

$o or $m Domain A subdirectory (recordings/CompanyA) 

$a Account The number of the account (e.g., 40, 70, 73) 

$d Date The date of the call (e.g., 20071220) 

$t Time The time of the call, e.g., 134349 

$i Direction of call The system substitutes an i for incoming calls, 

and an o for outgoing calls. 

$n Number The calling party’s number. Depending on the 

direction of the call, this field could reflect the 

number of either the calling or called party. 

$u Extension The primary name of the extension. Depending 

on the direction of the call, this field could reflect 

the number of either the calling or called party. 

$x Connected agent 

in ACD/hunt 

group calls 

Connected agent in ACD/hunt group calls 

$l Caller-ID 

$c CMC Client matter code 

$$ Produces a single $ symbol 

As mentioned earlier, the default setting is $r/$o/$a/$d-$t-$i-$n.wav. The recording will be placed 

into the recordings// directory as shown below: 

The files inside the directory will reflect the date and time, direction of the call, and the calling party: 


204 


Recordings can also be categorized according to date. The following recordings were generated on September 

7, 2010, using the string $r/$d/$t-$i-$u-$n.wav. 

Notes 

The recordings directory is relative to the working directory of the system and contains other WAV files, 

such as mailbox messages and greetings. If you would like to create a new directory for recordings, you must 

first create the directory before using it. The directory path must always be relative to the working directory 

of the system. 

When $u and $n are used at the domain level (e.g., $r/$d/$t-$i-$u-$n.wav), the recorded files will be 

created under /recordings/ directory, and the file will reflect the time 

the call was made, the direction of the call, and the extension number of both the calling party and the extension 

that was called. However, the filename will be based on whether the call was incoming or outgoing. 

For example, if extension 601 calls extension 604 (and Record incoming calls from extension has been set 

to Yes , as shown below), the resulting file will be 095404-i-604-601.wav ($i is the direction). Note 

that $u output represents the called party and $n the calling party). And if extension 601 calls extension 604 

(and Record outgoing calls to internal numbers has been set to Yes ), the resulting file will be 102019-o- 

601-604.wav. In this case, the $u output represents the calling party and $n the called party. 

Audio Prompts 

The voicemail and IVR systems of the snom ONE rely on audio prompts for their menu prompts. By default, 

these prompts are in U.S.-English and are located in the audio_* directory. 

The prompts are easily identifiable by their 2- and 4-digit prefixes. 


Prefix Description 

aa- Auto attendant 

bi- Built-in numbers 

co- Conference 

code- Feature code related 

ex- Extension-related 

mb- Mailbox 

pb- ACD (agent group) 

wk- Time-related 


Some administrators will need to download additional languages so that non-English-speaking users can 

hear prompts in their own language. When additional languages have been uploaded into the system, those 

languages will be available in the Default IVR Language and Default Tone Language dropdown lists: 

Uploading New Languages 

Audio prompts in many languages can be downloaded from snom ONE’s web site. Following is a list of 

available languages: 

Table 17-5. Country Identifier Codes 

Country Identifer Country Identifer 

Brazilian br Greek gr 

Danish dk Italian it 

Dutch nl Polish pl 

English en Russian ru 

English uk Spanish sp 

French (France) fr Swedish se 

French (Canada) ca Turkish tr 

German 

de 

To download a set of audio prompts: 

1. Go to wiki.snomone.com and click the Downloads link at the left: 


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2. Place the new language into the snom/snomONE directory. The new language will be displayed next 

to the default language: 


The new language will be displayed in the language dropdown lists: 

Note: The Default Tone Language setting controls the ringback and busy tones that a caller hears 

during voicemail prompts and IVR interaction. 

Customized Ringtones 

Agent groups and hunt groups can be configured with distinct ringtones so that users answering calls on 

these groups will be able to determine which type of call is coming in. Once custom ringtones have been 

configured, they can be selected from the Ring Melody dropdown in the hunt group and agent group account. 

To create custom ringtones, you will need to come up with your own ringtones and link them to the snom 

ONE by modifying the ringtones.xml file. The following instructions show you how to configure custom 

ringtones for a snom phone: 

1. Go to wiki.snomone.com and click the Downloads link at the left. 



2. Scroll toward the bottom and click Custom Ringtone Configuration File. 

3. Download the file, then open it in a text editor (e.g., NotePad++). The image shows a number of 

vendor elements that address various phone models (Polycom, Cisco). Phone models not indicated 

will automatically use the ringtone that is linked to the following address: 

<http://127.0.0.1/Bellcore-dr4> 

4. To create a custom ringtone for the snom ONE, create a new vendor element using one of the 

other vendor elements (copy and paste it). 

■ For the ua atttribute, enter snom.* 

■ Enter the URL of the wav file as shown below: 

<http://www.snom.com/download/wav/melody8.wav&gt 

5. Replace custom1 with a tone name that reflects the actual tone, in this case Chimes. 

6. Save the file. 

7. Place the file into the html directory (you may need to create the directory). 


208 



9. The new tone will be listed in the Ring Melody dropdown: 


BUTTONS 

Chapter 

18 

Buttons simplify the workflow in almost any environment and are ideal for all types of users. Once a button 

has been configured to the user’s extension, the user can press the button to execute whatever function has 

been configured to it. Button functionality includes many helpful options, including agent log in/log out 

and queue monitoring. These features allow call centers to provide an easy method to agents logging in and 

out of agent groups all day and for queue managers monitoring the queue. Button functionality simplifies 

almost any dialing sequence. Figure 18-1 shows the available options (each button type is detailed in Table 

18-1 on the following page). 

Buttons are part of a plug and play profile that has been configured to the user’s extension and provisioned to 

the phone. 

Figure 18-1. Button Functions 

Table 18-1 describes each type of button that can be configured on the phone. When defining the buttons, 

you can use phone numbers, extensions, star codes, as well as other numbers, such as the direct dial prefix. 

Table 18-1. Button Types and Parameters 

Button Type Parameter 

Private: Multiple lines can be configured on IP phones. Indicate those exten- 508 502 

sions here. 

Shared Line: Shared lines can best be compared to physical lines between the The name of a CO line 

system and the central office. In the past, PBX users seized a line before they in the domain (e.g., 

placed outbound calls. The shared line mode emulates this mode. When a user co1). 

seizes a shared line, other users also sharing this line see the LED go on. Calls 

can be put on hold by pressing the button, and other users sharing this line can 

pick the call up by pressing the button again. This is sometimes referred to as 

“key system” functionality. 


210 



Speed Dial: When buttons are used as speed dials, the phone will simply dial 

the programmed number (the LED next to the button will not light). Speed 

dials can be used for phone numbers, extensions, star codes, and other numbers. 

For example, if you want to transfer a ringing call to the voicemail of a 

specific extension, the parameter would need to include the following: 

■ The "transfer" star code (*77) 

■ The direct dial prefix (e.g., 8) 

■ The number of the extension (e.g., 40) 

9781234567 

501 (call an extension) 

*92 (blacklist a caller) 

*00501 (speed dial 

501’s cell phone) 

*77840 (transfer a 

call directly to 40’s 

voicemail) 

Example: *77840. 

Cell Phone: This mode allows you to quickly call someone’s cell phone. (After 

pressing the button, you will need to enter that person’s extension number.) 

*00 

Monitor extension (on phone): Use this mode if you want to see the call 

status of another extension. (For snom phones, use the “on the phone” mode.) 

The extended mode is intended for the WAC. The monitoring is not passive: 

When a call is either coming in on that extension (fast blinking) or is on hold 

or parked (indicated by slow blinking), you can pick the call up by pressing the 

button. Calls that are connected have a solid color. 

501 502 

Service Flag: This mode allows you to change the status of a service flag. (This 800 (an example of a 

button type only affects service flags that can be changed manually; it will have 

no impact on automatic, time-dependant service flags.) See “Service Flags and 

Buttons” on page 118 for instructions. 

service flag account) 

Park Orbit: Use this mode if you want to assign a button to a park orbit (a Enter nothing (this 

park orbit is an extension number where calls can be parked). This button can parks the call on the 

be used in a number of ways: 

user’s extension) 

■ If no parameter is indicated, the call will be parked on the user’s 

extension. 

505 (this parks the 

call on the extension 

■ If a park orbit is indicated, the call will be parked on the park orbit number that is being 

specified. 

used as a park orbit) 

Once this button has been configured, calls can be parked and retrieved by 

pressing this button. A parked call is indicated by a blinking light. 

Important: Before you can use this button, the star codes for Call Park and 

Call Park Retrieve must match (e.g., *85/*85). 

Do not disturb: This mode shows that extension’s DND status. The LED will *78 

light when DND is in active mode. 

Agent login/logout: This mode shows an agent’s login/logout status. This If you enter nothing, it 

button can be used in a number of ways: 

will log the agent out 

■ If the agent needs to log in and out of all agent groups at the same time of all agent groups. 

(or if the agent belongs to only one agent group), use a single Agent 

login/logout button and do not specify a parameter (leave it blank). 

If you enter a specific 

agent group (e.g., 

■ If the agent belongs to numerous agent groups and needs to log in and 511), it will log the 

out of a specific agent group before logging into another agent group, use agent out of that agent 

multiple Agent login/logout buttons and in the parameter field, enter 

the agent group number. Do this for each agent group. 

Once this button has been configured, the LED will light when the agent is 

logged in. Important: Before you can use this button, the star codes for Agent 

Log in/Agent Log out must match (e.g., *64/*64). 

group. 


Chapter 18: Buttons 211 


Redirect: This mode allows you to redirect calls to a predefined extension 

or number. To use this mode, enter the redirection target into the parameter 

field. Pressing the button activates redirection and lights the LED. Pressing the 

button deactivates redirection. This mode is useful when an executive assistant 

should take over calls. 

Busy Lamp Field (BLF): This mode shows when the specified extensions are 

in use. When the extension is connected, the LED will be solid; when the 

extension is ringing, the light will blink fast; and when the extension is holding 

a call, the light will blink slowly. This mode is similar to the monitor extension 

mode; however, it does not support the picking up of calls. When the user 

presses the button, the phone always dials the programmed extension. 

Agent Group: This button allows users to monitor the status of an agent 

group. When a caller is waiting in the group, the LED will blink quickly. Users 

can then press the button to pick up the call. 

Message Waiting Indicator (MWI): This button indicates when a message 

has arrived for the account that is entered into this field. The MWI button can 

be especially useful in situations where it is necessary to send the same voicemail 

to more than one extension. In this case, a group mailbox must be created, 

and the phones must subscribe to MWI for the extension of the groupmail 

account. (This button can also be used for older phones that do not have a 

Message button on the telephone keypad.) 

Creating a Button Profile 

503 (quickly redirect 

a caller to 503) 

Extension number(s) 

to be monitored 

511 (account number 

of agent group) 

Extension number of 

a group sales account, 

for example. 

Before the buttons feature can be used, a button profile must be created. Users can create their own profile, 

or domain administrators can create profiles so that users across the domain can select from a list of profiles 

that have been pre-configured. 

Two methods are available for creating profiles: the point-and-select method (which provides a dropdown 

from which you can choose the button type and add values) and the CSV approach (which allows you to 

enter each record directly into the web interface). 

Point-and-Click Method 

1. Navigate to Domains > Settings, and click Buttons. 

2. Enter a name into the Create a new button list field. 


212 


3. Click Create. The new profile will be listed under Name as shown below: 

4. Click the edit icon (the pencil). 

5. Enter a name for the button. Allowable values include 1 through 38 and dnd. Numerically, the 

values cannot exceed the number of buttons that are available on the phone. If you have a 12-button 

phone, the values must range from 1 to 12. If dnd is used for one of the buttons (for a phone that 

does not have a DND button), replace the numerical reference with dnd. 

The values assigned to the buttons correspond with the positioning shown below: 

Figure 18-1. Values to use for Button Profiles 

1 

2 

3 

4 

5 

6 

7 

8 

9 

10 

11 

12 

6. From the dropdown, select the type of button you want to use (see Table 18-1 for a description of 

each button type). 

7. Enter a parameter (see Table 18-1). 




9. Repeat Steps 5 through 8 for each additional button. 

10. To create additional profiles, repeat Steps 2 through 9. New profiles will be listed under Name as 

shown below: 

CSV Method 

1. Complete Steps 1 through 8 from the “Point-and-Click” method (previous pages). The result will be 

one button profile with only one button set. 

2. Click the here link as shown above to switch to a text-based editing window. A text-based editing 

window with one button record will be displayed: 

3. Follow the same format for the other buttons, using the following table for the button types and 

Table 18-1 for the parameters. 

Button Type CSV Syntax 

Private line 

private 

Shared line 

shared 

Speed dial 

speed 

Cell phone 

cell 

Monitor extension (on phone) extension 

Monitor extension (on PAC) long 

Service flag 

flag 


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Button Type CSV Syntax 

Park orbit 

park 

Do not disturb 

dnd 

Agent login/logout 

login 

Redirect 

redirect 

Busy lamp field (BLF) 

blf 

Agent group 

queue 

4. When finished, click Save. 

Assigning Button Profiles to Users 

A button profile can be assigned to a single user or to multiple users. The following procedures outline both 

scenarios. 

Assigning a Profile to a Single User 

1. Navigate to Domains > Accounts. 

2. Open the extension account. 

3. Click Buttons. 

4. From the Configuration Profile dropdown, select the profile you want to use. 




The new profile will be sent to the phone. Rebooting should not be necessary. On rare occasions, it 

may be necessary to reset the phone to force the phone to take on the new button mapping. 

Massive Update of Profiles 

Using the single-user method shown above, you can assign the same button profile to multiple users, but if 

a large number of extensions are involved, the process would be unnecessarily time consuming. A quicker 

approach would be to use a CSV file at the time you create the account/extension (see Appendix A for instructions 

on creating accounts with CSV). If you have already created the extensions, you can still use the 

CSV approach. In this case, you will need to modify the extension information by importing a CSV file that 

reflects the button profile information. Following are instructions: 

1. Create your CSV file using the instructions shown in Appendix A. Keep the following in mind: 

■ The values you assign to type and alias must match the extensions’ existing values for these 

parameters. 

■ The parameter that is used for the button profile is profile. Be sure to assign the correct value 

to this parameter (the value is the name of the button profile). Using the list of profiles shown 

below, the values would be 12 buttons, 2 buttons, and Front Desk. 

2. Navigate to the Accounts tab. 

3. Click Create (top left). 

4. From the Account Type dropdown, select Import CSV (Modify). 


Key System Configuration 

The buttons feature can be used to emulate a key system when combined with a snom phone. Although 

there are no physical cables to connect the snom ONE telephone system with the outside world, as with a 

TDM-based PBX, it is still possible to emulate the behavior of TDM-based PBXs and still have “CO-lines.” 


216 


CO-lines are associated with trunks, and each trunk may have several. CO-lines are listed in the account list 

because they share the same namespace. 

Once configured, the buttons will have the following behavior: 

■ When a corresponding extension receives a call, the button will blink (by pressing the button, the 

call can be picked up). 

■ When the corresponding extension is busy, the button will light. 

■ When the button is pressed while the extension is idle, the button will call the extension. 

Setting up a Key System 

Important: Before beginning this procedure, do plug and play for the phone that is being configured (see 

Chapter 19). 

1. Set CO lines on trunk (this setting is made on the trunk). Enter the name of each CO-line. 

Note: Because users can subscribe to the state of the CO-lines, the names given to CO-lines must be 

unique across the domain. For example, you can set up four CO-lines on Trunk1 with the names 

“co1 co2 co3 co4” and more CO-lines on Trunk2 with different names like “co5 co6 co7 co8” (the 

list of CO-lines must be separated by a space). The system will reject names for CO-lines that are 

already used by accounts or other CO-lines in the same domain. 

2. Create a buttons profile for the phone. Do this by going into Domains > Buttons. Enter a name for 

the buttons lists (the name must be unique across the domain). Then click Create. 

3. In the buttons profile you just created, create a button for each of the co-lines from Step 1. To do 

this, name the button (e.g., 1), choose Shared line from the dropdown, then enter the name of the 

co-line (e.g., co1). Click Save. Repeat this step for each button. 

4. From the user’s extension account, click Buttons. From the dropdown, choose the buttons profile 

you just created, and click Save to assign it to the extension. 



To verify that the phone received the plug and play configuration, you will need to check the function key 

configuration from the snom GUI. This is detailed in the following steps: 

1. Enter the IP address of the phone into a web browser (e.g., http://192.168.1.3). If you don’t 

know the IP address, press the question mark key on the phone. The IP address will show in the 

telephone display. 

2. Click Function Keys at the left of the screen. 

Notes 

Toward the lower half of the screen, the buttons that have been configured for Button will be listed. 

■ Key systems operate on the lines and not on extensions. If a call comes in on line 1, then everyone 

will see line 1 flashing. When the call is answered, the light will go solid on everyone’s phones. If 

the call is put on hold, everyone’s light will flash again. 

■ This method is simple compared to parking and picking up a call on the extension. 

■ Key system emulation is good for small offices but is not practical for large enterprises, as there 

would be too many lines to monitor on a T1. 

Multiple Identities and Button Profiles 

Multiple identities on a single phone (shown below) can be problematic with button profiles because a 

phone can utilize only one button profile at a time. When multiple identities are involved, the tendency is to 

configure a button profile that includes a private line for each user, which is okay to do. The problem occurs 


218 


when a new button profile gets uploaded after the first one and overrides the private line settings. In these 

cases, it may be wiser to manually configure a button profile on the handset instead of using plug and play. 


PLUG AND PLAY 

Chapter 

19 

The goal of plug and play is for the phone to be able to find the snom ONE and download its configuration 

settings with little or no configuration. In many cases, this can be done without actually touching the 

phone. Plug and play gives the phone accessibility to many system settings and features that can be configured 

by the system, making it unnecessary to configure these settings on the phone itself. When plug 

and play is done properly, setting up a group of IP phones is as easy as setting up a series of analog phones, 

removing the need to manually configure each and every phone. 

Finding snom ONE 

So how does the phone find the snom ONE? In most environments, the phone will find its IP settings 

automatically via a DHCP server. Once a DHCP is found, it will learn its IP address, subnet mask, default 

gateway, and DNS server. Once the phone can access the network, it can start looking for the snom ONE 

and begin provisioning its configuration files. How the phone actually finds the snom ONE is governed by 

whether the snom ONE is local or remote. Table 19-1 shows the possible scenarios and the protocols that 

are used to facilitate each. 

Table 19-1. Plug and Play Methods 

IP Address 

Touch- 

Free 

Method Locator LAN WAN Yes No Requirements 

DHCP, option 66 DHCP � � DHCP server must support 

option 66. 

SIP SUBSCRIBE 

(multicast) 

Multicast � � Limited to a single snom ONE. 

Mass deployment HTTP, 

TFTP, or 

HTTPS 

� � � snom must have your server info 

and MAC address. 

Manual setup HTTP � � � All settings must be configured 

manually. 

Each plug and play method shown in Table 19-1 is described below: 

■ DHCP, option 66: This method can be used to provision in both LAN (internal) and WAN 

(remote) environments as long as you can control the local DHCP server and the DHCP server 

supports option 66. Once the extension has been created, all configuration settings are made using 

the GUI of the phone. The phone retrieves its IP address and other configuration information 

through dynamic host configuration protocol (DHCP). While this method can be used to find the 

snom ONE on the WAN, it cannot guarantee that the phone’s MAC address will not be spoofed. 

Because of this, the user will be prompted to enter the password on the phone. (On the LAN, the 


220 


snom ONE looks in its ARP cache and is 100% sure who it is talking to, so users on the LAN will 

not be required to authenticate.) Instructions using this method begin on page 224. 

LAN 

TFTP 

192.168.1.x 

Private IP 

HTTP 

209.170.1.x 

Public IP 

■ SIP SUBSCRIBE (multicast): Multicast plug and play is generally used when the DHCP server 

does not support option 66. Since multicast packets cannot be routed over the public Internet, this 

method cannot be used if the phone is connected remotely. With this method, the phone gets its 

IP address from the DHCP server and multicasts the SUBSCRIBE message with its MAC address, 

vendor information, model information, and the version number. This method requires no 

configuration on the phone, but the Listen to sip.mcast.net setting will need to be turned on and 

the MAC address for the desired extension account will need to be set up. Instructions using this 

method being on page 225. 

■ Mass deployment (“auto provisioning”): This method, known as redirection service, allows for 

touch-free, WAN-based plug and play and can be used for a large deployment of phones. You can 

arrange to have the MAC addresses of your phones provisioned to a setting URL that reflects your 

snom ONE. This way, when the phone boots up, the DHCP server will tell the phone your snom 

ONE’s IP address, as well as other information. The redirection service requires a special arrangement 

with snom (mass deployment is discussed in more detail on page 226). 

■ Manual setup: This method is done by accessing the phone through a web browser. You will need 

to configure the phone’s HTTP client information and the Settings URL setting so that the phone 

knows where the snom ONE is. Once the phone has logged into the system, the snom ONE will 

know which configuration files to send down to the phone (see page 220). 

Plug and Play Checklist 

Many snom ONE settings can be downloaded into the phone when the phone requests the configuration 

files. It is best to configure these settings ahead of time, so use this section as a preparatory checklist when 

preparing a phone for plug and play. 

Configuring the Administrator Settings 

To configure administrator settings for plug and plug, complete the following steps: 

1. Go to Admin > Settings > PnP. 

2. Scroll down until you see the snom settings. 

3. Choose a transport layer from the available choices: UDP, TCP, or TLS. 


WAN

Chapter 19: Plug and Play 221 

4. If you have snom 300 phones and would like to customize them, you have the option to statically 

configure buttons 3 through 6 (see Figure 19-2). 

5. For firmware, select the version you would like to have in the phone. If the firmware in the phone 

is newer than what is shown on the screen, you will need to get a matching version of software from 

snom’s website; otherwise, the phone will be downgraded. You can copy the software to a web server 

of your choice and even drop it into the snomONE/html directory so that the system can serve it. 

Figure 19-2. snom Plug and Play Settings 

Configuring the Domain Settings 

To configure the domain settings for plug and plug, complete the following steps: 

1. Go to the Domains tab, click the domain, then click the Settings tab. 

2. Choose a default IVR and web language, a time zone, and country and area codes. 

■ The IVR language influences the voice prompts of the voicemail system. 


222 


■ The web language is used on the web interface pages. 

■ The time is reflected on the telephone display and in voicemail/email. It is also used in autogenerated 

reports, such as CDRs and status reports. 

■ The country and area codes affect the formatting of address book phone numbers. 

3. Scroll to the bottom of the page to Provisioning Parameters, and select a default dial plan scheme. 

■ The dial plan scheme tells the phone when to begin dialing, and numerous schemes are available 

(see page 62). If you want to customize the dial plan scheme, you can overwrite the 

http://localhost/snom_3xx_dialplan_usa2.xml?model=snom320 file (where 

localhost is the FQDN or IP address of your snom ONE, and snom_3xx is the model of 

your phone, e.g., snom820, snom 870, etc.). Enter the URL into a browser, then download the 

file. Once you have made the necessary changes, place the modified version into the html directory. 

When snom ONE looks for the file, it will look in that directory before looking on the 

server. 

Creating an Authentication Password 

The authentication password setting allows administrators to provision phones and log into a phone’s web 

interface remotely. This setting enables administrators to use identical usernames and passwords for all 

devices in a domain, simplifying the authentication process from the GUI of the phone. The password will 

bypass users’ passwords (the web interface passwords), so users will not be able to use their passwords; hence, 

this setting can be used to keep users out of the phone’s web interface and prevent them from inadvertently 

changing anything. When accessing the user’s extension from the GUI of the phone, the format extension@ 

domain is still required. The default username is admin, and the default password is password. 

To set the Authentication Password setting, complete the following steps: 

1. Go to Domains, click the domain, then click the Settings tab. 

2. Scroll to the bottom of the page until you see Provisioning Parameters. 

3. Enter a user name and password. 


Setting up the Domain Address Book 


Domain address books can be used by all extensions on the domain, and users can upload them into their 

phone (see page 275 for more on address books). Domain address books can be created through CSV files 

(instructions are provided in Appendix A). 

Configuring the Button Profiles 

This section allows you to create button profile(s) and assign them. Not all users will need the same button 

profile, so you may want to set up multiple profiles (see Chapter 18). If configuring a button for the busy 

lamp field (BLF), the Watch the calls of the following extensions setting (in the user’s extension account) 

will need to list the extension(s) that will be monitored, and the Permissions to monitor this account setting 

(for the extension being monitored) will need to be set in order to allow for the monitoring. 

Creating the Extension(s) 

The MAC address is an important setting when preparing a LAN-based extension for plug and play. 

Important: WAN-based provisioning relies on login information rather than on MAC addresses, so if you 

are preparing WAN-based extensions for plug and play, you can skip this section. 

The MAC address is the phone’s unique hardware number and is used to identify the phone to snom ONE. 

It is a 12-digit hexadecimal number located on the back of the phone and is written in the following format: 

MMMMMMSSSSSS, where MMMMMM represents snom’s ID number and SSSSSS represents the serial number 

that snom assigned to the phone. 

00041323024C 

MAC Address 

The snom ONE needs to know how to associate the phone’s MAC address with an extension. What determines 

this association is the Plug and Play setting located in the user’s account. Three different options are 

available: manual, permanent assignment, and temporary assignment. These methods are detailed in Table 

19-2. 


224 


Once you have selected a method, you will need to go to the extension account you want to associate the 

phone with and modify the Bind to MAC Address setting (see page 107). Populate the field with either the 

phone’s MAC address, a star (*), or a question mark (?) (see Table 19-2 for syntax details). To get the MAC 

address of the phone, press the question mark (?) key. It will be displayed on the telephone display. 

Table 19-2. Plug and Play Methods 

Method of 

Plug and Play 

Manual Setting up a few 

phones 

Permanent 

Assignment 

(*) 

Temporary 

Assignment 

(?) 

Use Case 

Scenario How it’s Use 

Deploying phones 

in a specified order 

in a large office 

Setting up phones 

at trade shows, class 

rooms, etc. 

The manual mode allows you to permanently set the MAC address 

in the extension account. This will bind the registration to a specific 

phone (see Bind to MAC Address on page 107). The MAC 

address can be used in multiple extensions, and multiple MAC 

addresses can be assigned to the same extension. This allows for 

users to have multiple phones. (Use a space to separate the MAC 

addresses.) 

This mode allows you to easily assign MAC addresses to a large 

number of extensions. For example, rather than entering 100 different 

MAC addresses into separate user accounts, you can use the 

star symbol (*) to expedite the process. The system will assign a 

permanent extension to each phone once it boots up and requests a 

configuration from the system. The system will remember the MAC 

address of the phone and from that point on, the extension number 

will be used exclusively for that phone. This mode requires that a 

star (*) be used in the Bind to MAC Address field. 

The temporary assignment mode simplifies preparation when setting 

up at trade shows and for classroom training sessions. When 

this mode is used, the system will not remember which phone had 

been assigned to which extension. A phone might receive a new 

extension number every time it boots up. (As long as no other user 

agent is registered to a particular extension, the extension will be 

available for plug and play.) This mode requires that a question 

mark (?) be used in the Bind to MAC Address field. 

If you are creating extensions using CSV files, the options shown in Table 19-2 can still be used. For the 

parameter, use mac and for the value, use the syntax that is indicated in the table. 

Performing Plug and Play 

DHCP, Option 66 

This method can be used only if you have a DHCP server that supports option 66. You do not need to rely 

on multicast. Once you have set option 66 with the IP address of snom ONE, the DHCP server will be able 

to tell the phone where snom ONE is. 

Note: If the phone was configured against a different location/snom ONE, you will need to reset the phone 

(see Resetting the Phone on page 229). 



Preliminary step: If you have multiple snom ONE systems on the network, turn off the Listen to sip.mcast. 

net setting in the snom ONE web interface (Admin > Settings). Changing this setting requires a system 

restart. 

1. Set up the MAC address of the phone for the desired extension account. To get the MAC address of 

the phone, press the question mark (?) key. It will be displayed on the telephone display. (This does 

not work for snom 300 phones; refer to your user guide.) 

2. Connect the Ethernet cable to the phone, and power up the phone. 

When the phone reboots, it will get the IP address from the DHCP server (it may also get the TFTP 

server address). The phone will multicast the SUBSCRIBE message with its MAC address, vendor 

information, model information, and the version number. The system looks up the vendor information, 

model information, and version information of the phone from the message with the PnP 

configuration data (from the built-in pnp.xml file). If there is a match, then the system will send 

the NOTIFY with the “Settings URL” file for the snom phone in the message body. 

Example: https://192.168.1.109:443/provisioning/snom320.htm. This file contains 

a list of files that are needed for the operation of the phone. Once the phone receives this file, it will 

send a request for each of the files that are in the snom320.htm file (see Table 19-3). The snom 

ONE will generate these files and then send them to the phone based on the original SUBSCRIBE 

from the phone. The content of these files is controlled by the templates that are built into the binary 


Table 19-3. Provisioning Files Requested by Phone 

Settings Provisioning Files 

Provisioning file http:///provisioning/snom320.htm 

Firmware 

http:///provisioning/snom_3xx_ 

fw.xml?model=snom320 

Password 

http:///provisioning/snom_3xx_phone. 

xml?model=snom320 

Dial plan scheme http:///provisioning/snom_3xx_dialplan_ 

usa2.xml?model=snom320 

Buttons 

http:///provisioning/snom_3xx_fkeys. 


Web 

http:///provisioning/snom_web_lang. 

language 


Phone language http:///provisioning/snom_gui_lang. 


Custom settings http:///tftp/snom_320_custom.xml 

SIP SUBSCRIBE (Multicast) 

This method is limited to environments with a single snom ONE. It is generally used in LANs, as multicast 

typically does not traverse routers. It requires configuration on the phone and is not supported if the phone 

is connected remotely (over the public Internet). 

Note: If the phone was configured against a different location/system, you will need to reset the phone (see 

Resetting the Phone in this chapter). 


226 


1. From the web interface, click Admin > Settings, then scroll down to SIP Settings. 

2. Turn on the Listen to sip.mcast.net setting. (Note: Changing the Listen to sip.mcast.net setting 

requires a system restart.) 

3. Set up the MAC address of the phone for the desired extension account. To get the MAC address of 

the phone, press the question mark (?) key. It will be displayed on the telephone display. (If you have 

a snom 300, refer to your user guide for instructions on getting the MAC address.) 

4. Connect the Ethernet cable to the phone, and power up the phone. 

When the phone reboots, it will get the IP address from the DHCP server (it may also get the TFTP 

server address). The phone will multicast the SUBSCRIBE message with its MAC address, vendor 

information, model information, and the version number. 

The system looks up the vendor information, model information, and version information of the 

phone from the message with the PnP configuration data (from the built-in pnp.xml file). If there 

is a match, then the system will send the NOTIFY with the “Settings URL” file for the snom phone 

in the message body. 

Example: https://192.168.1.109:443/provisioning/snom320.htm. This file contains 

a list of files that are needed for the operation of the phone. Once the phone receives this file, it will 

send a request for each of the files that are in the snom320.htm file (see Table 19-3). The snom 

ONE will generate these files and then send them to the phone based on the original SUBSCRIBE 

from the phone. The content of these files is controlled by the templates that are built into the binary 


Mass Deployment 

Mass deployment (also known as “redirection service”) allows for touch-free, WAN-based plug and play and 

is ideal for a large deployment of phones. To use this service, you will need to contact snom and have the 

MAC addresses of your phones provisioned to a setting URL that reflects your snom ONE. 

By default, when the phone boots up for the first time, it contacts the snom server for its software updates 

(the setting URL is defaulted to www.snom.com), but once the setting server has been pre-programmed to 

redirect the phone to a different server (e.g., pbx.itsp.com), it will tell the phone the location of your snom 

ONE’s IP address. The configuration files will be created in real time and will be sent to the phone using 

one of three protocols: HTTP, TFTP, or HTTPS. 

The terms and conditions of snom’s redirection service is detailed on the snom wiki (wiki.snom.com). 

Download the document, fill it out and sign it, then scan it and send it back to redirection@snom.com as a 

PDF attachment. You can also contact your nearest snom headquarters and send the contract by fax or mail. 

snom will contact you by e-mail to provide you with the necessary credentials. 

Manual Method (HTTP) 

This method of plug and play can be used for phones in either a LAN (internal) or WAN (remote) environment. 

When used in a LAN environment, you need to set up the MAC address for the extension(s). This is 

done when you first create the extension (see the plug and play checklist in this chapter). When this meth- 



od is used in a WAN environment, you will be able to use login information to authenticate the account, 

and all configurations will be made from the GUI of the phone. 

Preliminary step: If you have multiple snom ONE’s on the system, turn off the Listen to sip.mcast.net 

setting in the web interface (Admin > Settings). Changing the Listen to sip.mcast.net setting requires a 

system restart. You can do this by either restarting the computer, right-clicking My Computer and clicking 

Manage (choose Service Applications from the right pane), or by entering net start snomone from a 

command prompt. 

1. Power up the phone. 

2. Log in to the web interface of the snom phone by entering the phone’s IP address into a web browser. 

3. Click Advanced from the left-hand section of the screen. 

4. Click the QoS/Security tab. 

5. Scroll toward the bottom of the page until you see the HTTP Client information. 

Note: The HTTP Server section contains user name/password information that is needed to log in 

to the phone using a web browser. After PnP, the system will configure this section with information 

from Domain > Settings > Authentication User and Domain > Settings > Authentication Password. 

6. For User, enter the extension number of the phone. (If the phone had been registered to another 

extension and was not reset, you can delete the existing extension and enter a new extension.) 


228 


Note: In multi-domain environments, the username must contain the extension and the domain 

(e.g., extension@domain). 

Note: The snom 870 requires the following format (there is no @ support): 

508;domain=sip.itsp.com 

The domain does not have to be a fully qualified DNS domain. The system will first identify the 

domain and then look for the extension within the domain. If found, it will request the HTTP Authentication 

details. 

7. For Password, enter the extension level web password. The Authentication User/Password setting 

(see also the plug and play checklist) allows the administrator to create identical usernames and passwords 

for all devices in a domain. 


9. Click the Update tab. 

10. Set Update Policy to Update automatically. 

11. Set Settings URL to http:///provisioning/, where 

can be the IP address or the FQDN of the PBX server, and snom820.htm 

should be replaced with the correct phone model (snom360.htm, snom370.htm, etc.). 

Important: Be careful when entering the text for the Settings URL setting. Plug and play will not 

work if the syntax is not entered exactly as indicated above. 

12. Set PnP Config to on if the extension is in a LAN environment. For WAN, turn this setting to off. 

Figure 19-3. Setting URL for the snom 820 


14. Reboot the phone when prompted. 

15. Click Yes to confirm that you want to reboot. 

During the reboot process, the phone will contact the snom ONE and request the snom820.htm 

file with the authentication information that was set during Steps 6 and 7. The system will verify the 

authentication information and, if valid, will generate and send the requested file. Once the phone 

receives the file, it will start sending a request for each of the files that are in the snom320.htm file 

(Figure 19-4). The system will generate these files and send them to the phone based on the request 

from the phone. The content of these files are controlled by the templates that are built into the 

binary of the snom ONE. 


Figure 19-4. Log File for snom 820 

Resetting the Phone 


If you have a phone that has been registered in the past or if are having trouble going into the phone GUI, 

you will need to reset the phone’s configuration. You can do this through either the web interface, the phone 

itself, or from the GUI of the phone. 

Using the Web Interface 

1. Go to Settings > Maintenance > Reset Values. Click the confirm button. 

2. Enter the password (snom’s default password is 0000). 

3. Press the confirm button to reboot the phone. (The phone will update its default configuration.) 

Using the Phone 

From the phone, enter **#*. 

Using the Phone GUI 

1. Enter the IP address of the phone into a web browser. 

2. Click Advanced from the yellow left-hand section of the screen. 

3. Click the Update tab. 


230 

4. Click Reboot. 


Registering Numerous Extensions to One Phone 

Registering multiple identities to a single phone can be useful in cases when the phone is being used to 

answer calls for numerous companies. Each identity has its own extension, and the extension can be configured 

to a system that is either local or remote. 

1. Enter the MAC address of the phone into the Bind to MAC Address field of the new extension, as 

shown below (Registrations tab). 

2. Reboot the phone by entering **#* from the telephone key pad. The phone will reboot and reprovision 

itself as it counts down from 20 to 1. 

3. Enter the IP address of the phone into a web browser. 

4. Click Settings > Identity (at the bottom left of the screen). 


5. Populate the screen using the following information: 

■ Identity Active: Enable this setting 

■ Display name: Enter the name that will be used for the account. 

■ Account: Enter the additional extension number. 

■ Password: Enter the SIP password of the additional extension. 

■ Registrar: Enter the domain name of the system. 


■ Outbound Proxy: Specify the IP or DNS address of the registrar/proxy where you want to 

register this account. After a successful registration, the registrar knows how to reach this specific 

identity and can route requests (e.g., incoming calls) from other registered parties to this phone. 

If the extension is local, enter the IP address of the system; otherwise, enter a domain name. 

■ Failover: This field allows you to enter an extension where calls will be sent if the extension is 

down. 


7. Click Re-Register. 

The telephone display will reflect the new identity: 

Overriding Plug and Play Defaults 

Plug and play default settings can be modified using a few different methods. The first method is to use the 

Admin > PnP tab, which allows you to modify only a small number of settings. To make large-scale custom 

configurations, you will need a customized XML file. 

Changing Default Settings from Admin > Settings > PnP 

This section shows you how to manually change plug and play settings using the Admin > PnP tab. The 

only changes that you can make using this method are the transport layer, the password of the snom phone, 

some of the buttons on the snom 300 phone, and the firmware versions. Once the changes have been made 

and the phone has been rebooted, the phone will request the configuration files from the server and the 

changes will take effect. 


232 


Figure 19-5. Overriding Default Plug and Play Settings (Using Admin > PnP) 

To make any of the custom settings shown in Figure 19-5(A), use the following steps: 

1. Go to Admin > Settings > PnP and scroll down to the snom settings. 

2. Make the necessary configurations. (The button settings are for snom 300 phones only.) 




4. Reboot each phone on the domain by going into the Registrations tab of each phone and clicking 

check-sync. 

The generated files within each user’s account will reflect the changes. 

Changing Default Settings Using a Customized XML File 

snom phones have hundreds of settings that are used to give the phone its functionality. Not all these settings 

are currently part of plug and play. However, through the use of a customized file, you can integrate 

missing settings into plug and play. Before doing any customization, read through these next few pages so 

that you have a thorough understanding of how customized files reach the phone. 

Generated Files and Where They Come From 

When the phone contacts the system for its configuration files during plug and play, the system creates a 

directory for each extension, so that it will have a place to put the files. The directory is called generated 

and the files are shown in Figure 19-6. The full path on Windows is Program Files\snom\snomONE\ 

generated\pbx.company.com, and on the Mac it is usr\Applications\pbx\generated\pbx. 

company.com. 

Figure 19-6. Directory Structure of Extensions on Domain 

The files are generated from templates that exist in the binary of snom ONE and are inaccessible. When the 

phone boots up, it looks at the files in the generated directory and will take on the settings that are represented 

by those files. By default, the only files it sees are the files shown in Figure 19-6. However, by placing 

snom_3xx_custom.xml (xxx must reflect the phone model) into an tftp directory (which you must 

create), the phone will see those settings as well. 


234 


Figure 19-7 provides an abbreviated overview of how plug and play works and the role that generated files 

play. After snom ONE has sent the phone its snomxxx.htm file (Step 4 below), the phone will request its 

configuration files. At this point, snom ONE will process the plug and play settings and compile them into 

generated files, which are received by the phone once it reboots. 

40 

snom_3xx_phone.xml 

40 

snom_3xx_dialplan.xml 

Figure 19-7. Generated Extension Files 

Extensions 

40 43 46 49 

41 44 47 50 

42 45 48 51 

PBX Templates 

snom_3xx_fkey.xml 

snom_3xx_web_lang.xml 

snom_3xx_gui_lang.xml 

Generated Files 

43 46 49 

41 44 47 50 

42 45 48 51 

snom_3xx_custom.xml 

42 44 46 48 50 

41 43 45 47 49 51 

1. Extensions are created 

on the snom ONE. 

2. Plug and play begins. 

3. Authentication process 

takes places (either MAC 

address or login info). 

4. snom ONE gives phone 

its snomxxx.htm file. 

5. Phone request its 

configuration files. 

6. To produce the files, 

built-in templates 

process the extension 

settings. 

7. Generated files are 

produced. 

8. During reboot, each 

extension gets its 

settings from the 

generated files. 



Figure 19-8 shows one of the template files (snom_3xx_phone.xml) that the system uses to create one of 

the generated files. Though you cannot access this file, it is important to notice the structural syntax that is 

used in the file because the customized file will need to follow a similar structure. 

Figure 19-8. Sample of a Template File (snom_3xx_phone.xml) 


236 


Figure 19-9 shows an example of the generated version of snom_3xx_phone.xml. 

Figure 19-9. Sample of a Generated File (snom_3xx_phone.xml) 



Before you can create a customized file, you need to know the XML element tags that have been reserved for 

each setting, as well as the parameters that can be used. You can find this information on the snom web site 

(Figure 19-10). (Once you have HTTP’d into a phone, click Settings at the bottom left part of the screen. 

The settings are listed on the right side of the page.) You may also need the snom administrator guide for an 

explanation of parameter requirements. 

Figure 19-10. snom Phone Settings 


238 


When the settings are viewed in XML, the following document tree is displayed: 

Figure 19-11. Document Tree of snom Phone Settings 

Creating a Customized File 

1. Create a directory named tftp on the system: 

Windows: Program Files\snom\snomONE\tftp 

Mac: usr\Applications\pbx\tftp 



2. Create the customized file using an XML editor. Following are a few lines from a customized file 

(these settings are highlighted in Figure 19-11). The information that will be picked up by the 

phone is shown in bold. 

 

 

 

9781234567 

6031234567 

off 

 

 

3. Save the file as snom_xxx_custom.xml (where xxx reflects the model of the phone; for example, 

snom_320_custom.xml for a snom 320). 

4. Place the file in the TFTP directory you just created. 

Once the phone reboots, it will pick up your customized settings. 

Troubleshooting 

Following are solutions for common plug and play issues: 

Table 19-4. Troubleshooting 

Issue Notes Resolution 

Provisioning countdown issue: The The phone may be sending the Reset the phone. 

phone was registered somewhere else, old authentication information 

and you are trying to move the phone to 

a new extension. 

and the system will reject it. 

Provisioning countdown issue: The This will happen if sip.mcast. Turn on sip.mcast.net on 

phone tries the multicast SUBSCRIBE net is turned off. 

the system. You can also try 

and the system ignores it. 

turning off the Advanced 

> Update: PnP Config on 

the phone and following the 

HTTP-based PnP process. 

Provisioning countdown issue: The This applies to the HTTP PnP Properly match the user- 

proper username and password combina- method. 

name and password on both 

tion have not been set. 

the snom ONE and the 

phone. 

Provisioning countdown issue: Some 

Turn off the SIP awareness 

SIP-aware routers are altering the mes- 

on the routers or replace the 

sages. 

routers. 

Multiple snom ONE instances are run- This could inadvertently affect Verify who is responding 

ning on the same network. 

the multicast SUBSCRIBE- to the phone’s requests by 

based PnP. 

looking at the phone log 

(or some type of network 

sniffer). 

Username and password do not show up. You may have a firewall issue. 


240 


Issue Notes Resolution 

Can’t HTTP into the phone or unable to 

reset the password on the phone. 

1. Copy the phone’s firmware from the wiki.snom.com 

web site to the tftp directory of the snom ONE. 

2. Modify the PnP tab of the snom ONE with the firmware 

version that was downloaded. 

3. Rename it snomxxx.bin. 

4. Reboot the phone. 

5. Hit any key on the phone within 3 seconds, and then 

manually enter the IP address of the phone, the IP 

address of the snom ONE, the default gateway if the 

system is off from the network, and the subnet mask. 

6. Click the checkbox. 

The phone should connect to the snom ONE and download 

the firmware and reset itself. 


MUSIC ON HOLD 

Chapter 

20 

Music on hold (MoH) is recorded music that callers listen to while waiting on hold. It is used to avoid 

silence on the line and is used in several places of the system: when a call is on hold and when a caller is waiting 

in an agent group queue (see Chapter 10 for more on agent groups). The system allows three different 

source types for its music on hold: file, wave input, and RTP stream. These sources serve different purposes 

and are detailed below. The sources can be used in parallel in different locations on the same system. 

■ Files: The system can use one or more files for MoH. These files are read by the system on demand 

and are played in an endless loop. Files must be placed manually into the audio_moh directory of 

the system. Although files are loaded only once, long files will be read into memory and can consume 

a lot memory space. Keep in mind that each minute of the file will require about 1 MB of 

memory space, so a 7-minute file will take up about 6.7 MB (128 kbit/s * 7 * 60 seconds). After it 

gets past the initial hick-up (jitter), subsequent calls to the MoH file will be smooth. 

Format: Files must be in 8-kHz sampling frequency, and they should be in 16 bit per sample 

signed format. The format must be mono WAV. You may also use other formats (u-law and GSM), 

but these formats will have less audio quality and require more CPU performance. 

■ Wave input: In Windows, snom ONE can read MoH from an audio input jack, which is a convenient 

way to connect a CD, MP3 player, or a radio to the system. The disadvantage of this method 

is that only one external music source can be used. 

You can also internally loop the audio output of the local computer back to the audio input of the 

computer. This allows you to use an MP3 player running locally to provide a large number of MP3 

files. However, we recommend keeping an eye on the memory usage of the MP3 player, as some 

players have memory leaks and slowly consume the memory of the computer. 

Note: This feature is currently available only for Microsoft Windows-based operating systems. The 

appliance uses the RTP streaming mode for the audio input jack. 

■ RTP stream: Streaming RTP data is a popular way of providing music from external sources. As 

with a telephone conversation, the system receives the audio data in a standard RTP stream. Several 

external tools are available that are able to generate a compliant RTP stream. Because the system 

can have several RTP streams, you can use this method to generate different music on hold sources 

for the system. 

Format: The RTP stream must use G.711 encoding. No SIP signalling is involved with this method, 

and the system does not send RTP data back. 

Important: Be sure to specify the port on which the system should listen for RTP input (e.g., 

42000). This port must be available on the system. If you change the setting, you might need to 

restart the system service so that the change takes effect. 


242 


Setting up Music on Hold 

MoH sources are available to all domains on the system, so setup is done from the administrator level. 

1. Navigate to Admin > Settings > MoH. 

2. Populate the fields using the following information: 

■ Name: Assign a name to the MoH source (this name will appear in the Available Sources list at 

the end of this exercise). 

■ Type: From the dropdown, choose the type of MoH that you will be using. 

■ Domain: From the Domain dropdown, select your domain. 

■ Filename: This field depends on which type of MoH you are using: 

File: If you are using files, upload the file by clicking Browse. (Once the file has been uploaded, it will 

automatically get filed in the 

audio_moh directory.) 

WAV Input: If you are reading MoH from an audio input jack, the Filename field will disappear. 

RTP Stream: If you are streaming RTP data, enter the port on which the system should listen for RTP 

input (e.g., 42000). This port must be available on the system. If you change the setting, you might 

need to restart the system service so that the change takes effect. 

3. Click Create. The new source will be listed as an available source. 


Chapter 20: Music on Hold 243 

4. To verify that the source has been added on the domain level, navigate to Domain > Settings. You 

should see the newly added source in the Music on Hold Source dropdown: 

Once MoH has been created, it can be selected for use on a domain level within both the agent group and 

hunt group accounts. 

Editing MoH Sources 

Once you have created an MoH source, you can do the following: 

■ Change the name of the source. 

■ Change the domain(s) that will have access to it. 

■ Delete it from the list of available sources. 

To edit a source, click the source link and make the changes in the form below it. Then click Save. 

To delete a source, click the source link and click Delete. 

Creating WAV Files 

Creating a WAV file from a CD is easy and can be done using most music player programs. These instructions 

were done using WavePad (freeware from www.nch.com). 

1. Install WavPad on your computer. 

2. Launch the program (START > All Programs > WavePad Sound Editor). 

3. Put the CD into your CD-ROM drive. 


244 


4. Click File > Load Audio CD Track(s). 

5. Select the track you want to use and click Load. 

The following screen will be displayed while the track is loading: 

When the file has finished loading, you will see the following screen: 

6. Click Save File. 


7. Choose the .WAV format from the list if it’s not selected by default. 


8. Navigate to the audio_moh directory (C:\Program Files\snom\snomone\audio_moh), 

and save the file. 

Configuring Paging/Music on Hold 

The snom ONE Paging/Music on Hold application allows you to use a PC to stream audio from a music on 

hold input and play back audio on a paging output. This section details the installation and basic configuration 

of the application. The following conditions are assumed: 

■ You are running the application on the same system/PC on which you are running the system. 

■ You are running the system on a Windows PC equipped with a sound card. 

snom phones, due to plug-and-play features, will automatically play back the audio stream. The system 

searches for the first MC paging account in the domain and automatically configures the phones to play 

back that audio stream. This is valid for version 7.x only; later versions have multiple MC streams and will 

need to be configured to specify which phones are permitted to listen to a particular stream. 

Download and Installation 

Go to wiki.snomone.com and download and install the Paging and MoH application. 

The installation will create the application as a service. The installer will also copy a pagmoh.xml file to the 

installation folder. This file will contain the default settings that are used by the application. These defaults 

are listed below: 

■ Paging port = 5000 

■ MoH port = 6000 

■ Codec = µ law 

■ Service name = pagmoh 

You can edit this file so that different ports may be used. After changing any values, restart the service. 


246 

Setting up Paging 


Creating a Multicast Paging Account 

This section provides instructions for creating a multicast paging account on the system. 

1. From the Accounts tab, click Create and select Paging from the dropdown. 

2. Enter an account number and name, then click Create. 

3. Once the account has been created, click List to open the account and configure it as detailed below: 

■ Streaming Mode: From the dropdown, choose Multicast. Multicast mode allows you to 

designate an IP address as your paging group and create paging groups that encompass entire 

organizations. All devices that are configured to listen on the same multicast IP address will go 

into paging mode as soon as they receive RTP traffic on that port. 

Note: Not all phones support multicast paging. Phones that do have this support can be configured 

through the phone’s web interface. 

■ Destination: Enter an IP address into this field. The loopback address is especially important 

since the paging application and the system are running on the same system. 

Note: Multicast addresses range between 224.0.0.0 and 239.255.255.255. 

■ Source: List the extensions that are allowed to page this group. Enter a star (*) to give access to 

all extensions. 

■ Display Name: The display name will be used to identify the source of the call. SIP phones will 

display this text in the display area. 


account. 




5. From any registered SIP phone, dial 75 (which is the number assigned to the paging account in this 

example). You should hear the audio from the speakers connected to the audio OUT on the PC. The 

system will send paged audio to the 127.0.0.1:5000 and the paghoh application will read from 

port 5000 and write to the audio OUT device. 

Assigning Multicast IP Addresses to IP Phones 

To be part of a multicast paging group, a phone must have a multicast IP address that matches the snom 

ONE multicast IP address. To do this, you will need to configure the multicast settings for each phone. 

1. Go into the phone’s web GUI by entering the phone’s IP address into a web browser. The following 

screen will appear: 

2. Click Advanced at the left-hand section of the screen. 

3. Select SIP/RTP as shown below: 

4. Scroll to Multicast at the bottom of the page. 

5. Enable Multicast Support. 

6. Enter the IP address that was entered as the destination when configuring the paging account (refer 

to page 246). 



248 


The phone can now pick up the RTP stream and begin playing it. 

Configuring Music on Hold 

This section shows you how to create the new RTP stream-based MoH source. 

1. From the domain, click Admin > Settings > MoH, as shown below. 

2. Use the following settings to configure the Music on Hold source: 

■ Name: Click pagmoh rtp from the available sources. 

■ Type: From the dropdown, choose RTP Stream. 

■ Domain: Choose the domain(s) that will be using the Music on Hold. 

■ Port Number: The default port is 6000. 

3. Define pagmoh rtp as the MoH source within the domain where it will be used. Do this by going 

into the domain’s default settings (Domains > Settings), as shown below. 

4. Plug in the music source to the audio-in (generally the microphone input) of the PC that is running 

the pagmoh and the system. Make sure that the music is really being played on the music source. 

5. Pick up a phone, make a call, and place the call on hold. From the phone that is on hold, you will 

hear the music from the music source. 


XML Configuration File 


The configuration options in the XML file are set in the parameter:value format. Available parameters 

and sample values are shown below: 

Table 20-1. Available Options in pagmoh.xml 

Parameter Function Example 

page Defines the port where to listen for RTP traffic. If specified, 

it binds the socket to the provided address. 

5000 

moh Defines where to send the MoH stream. If the destination 

contains only the port number, the destination is the 

loopback device. 

6000 

log Sets the log level. Valid levels are 0–9. 5 

dev Sets the audio device. 

priority Defines whether the process should run at high priority or True 

not. 

False 

multicast Joins a multicast address. 

ptime Specifies the packet length in ms. Multiples of 10 

dtmf Defines what codec to use for out-of-band DTMF. 

map Sets the codec mapping. The argument is a list that maps l16, ulaw, alaw, 

the codec number to codec names. 

gsm, g722, and 

g726 

codec Specifies the codec that should be used for sending. The 

parameter is the name of the codec. 

ulaw 

ipv4 Use only IPv4. True 

ipv6 Use only IPv6. False 


250 



CALL DETAIL RECORDS (CDRS) 

Chapter 

21 

A call detail record (CDR) is a file that contains information about recent system usage. A CDR can include 

the identities of the originator and the endpoint, the duration of each call, the direction of the call, and 

many other important pieces of information. CDRs can be used for measuring traffic, and they can also be 

exported to third-party vendors, which makes them useful for billing purposes. 

A CDR can include the following information: 

■ Caller-ID 

■ Type of call (auto attendant, hunt group, agent group, etc.) 

■ CMC code 

■ Domain name of call 

■ Trunk name 

■ Language of call 

■ From and To headers 

■ Extension number, either originating or receiving 

■ Direction of call (in or out) 

■ Language of call 

■ Start time of call, as well as connected duration, answered duration, hold duration, and IVR duration 

■ Date and time of call 

■ Start and end times 

Automatically Generated CDRs 

For every call, the system generates three CDRs: a CDR that contains information about the extension 

(cdre), a CDR that contains details about the ivr (cdri), and another that contains information about the 

trunk (cdrt). Once these CDRs have been generated, they are stored in the working directory of the local 

system and are placed into the following directories: 

The length of time these CDRs are kept on the system is determined by the Keep CDR Duration setting 

(see page 26). The storage duration can be expressed in seconds, minutes, hours, or days. The CDR 

listing size setting dictates how many CDRs are permitted on the page (default is 30). This field is used to 

prevent a user from getting too many CDRs displayed at any one time. 


252 

“Extension” CDRs 


CDRs that are placed into the cdre directory contain extension-related information, such as caller-ID, To 

and From headers, and details as to whether the call was a missed call. All possible fields are listed in the 

following table: 

XML Tag Definition 

i Caller-ID of call, as seen in the SIP packet 

f The From header, as seen in SIP packet 

t The To header, as seen in SIP packet 

o Direction of call (i for inbound; o for out) 

u User ID 

d Domain name of the call. 

s Start time in seconds 

c Connection timestamp 

e Disconnect timestamp 

p IP address 

vq VQ report 

m Missed call 

cmc Client matter code 

rec Record location 

XML is the default format for CDRs. Figure 21-1 shows extension-related information wrapped inside 

XML start and end tags. These tags correspond to the XML tags in the above table. 

Figure 21-1. XML Output for CDRs—Extension 

“IVR” CDRs 

CDRs that are placed into the cdri directory contain IVR-related information. These CDRs indicate information 

about the IVR account that picked up the call (auto attendant, hunt group, etc.), the language of 

the call, and how long the caller had to wait before the call was picked up. All possible fields are listed in the 

following table: 


y Type of CDR (attendant, hunt, acd, external, etc.) 

u IVR account that ran the call (e.g., ACD) 



e Extension account that connected (e.g., Agent) 

f Descriptive From header 

t Descriptive To header 

d Domain in which the call took place 

l Language of the call 

s Time when call started 

di Duration that the call waited 

dr Duration of ringing 

dt Duration of two-way audio 

dh Duration of hold period 

r Reason 

Chapter 21: Call Detail Records (CDRs) 253 

Figure 21-2 shows IVR-related information wrapped inside XML start and end tags. These tags correspond 

to the XML tags in the above table. 

Figure 21-2. XML Output for CDRs—IVR 

“Trunk” CDRs 

CDRs that are placed into the cdrt directory contain trunk-related information, such as To and From 

headers, direction of the call, and connection/disconnection timestamps. The range of possible fields are 

listed in the following table: 


f The From header, as seen in SIP packet. 

t The To header, as seen in SIP packet. 

o Direction of call (i for inbound; o for out). 

r The remote party (as provided to the dial plan) 

l The account that was charged (dial plan) 

u The cost (if available) 

d Domain name of the call. 

s The start timestamp 

c The connection timestamp 

e The disconnect timestamp 

p The IP address (for inbound calls) 

vq VQ Report 

cmc Client matter code 

rec Record location 


254 


Figure 21-3 shows trunk-related information wrapped inside XML start and end tags. These tags correspond 

to the XML tags in the above table. 

Figure 21-3. XML Output for CDRs—Trunk 

CDRs for Third-Party Software Vendors 

CDRs can be exported to third-party software vendors for a wide range of purposes. When used for this 

purpose, CDRs can be exported to the following formats: 

■ CSV (comma separated files) 

■ SOAP (XML) 

■ Email (sent as attachments) 

To specify which format you want the system to export, you must use the CDR URL setting. This setting is 

located in Admin > Settings (see Chapter 2). 

In the example shown above, the CDR will be sent to name@company.com once the call is finished, and 

the CDR will be sent as an email attachment. Other options are available when exporting CDRs. These options 

are shown below: 

File Type Syntax 

CSV 

file:disk 

Email 

mailto:name@company.com 

Simple TCP 

cdr: 

SOAP/XML 

http:// 

https:// 

CDRs that are used for this purpose automatically include the following fields: 

Table 21-1. Default Fields for CDRs 

Field Values 

From “00432638316014” 

To “Steve Fuller” 

Call-ID 0080-1008-25C105DA-0@D141DFC50C7AA2248 

Direction I 

Remote +432638316014 

Local 411 


Field Values 

TrunkName PSTNGateway 

TrunkID 1 

Domain company.com 

LocalTime 20100310105708 

Start 20100310095708 

Connect 20100310095710 

DurationHHMMSS 0:00:10 

DurationSec 10 

End 20100310095720 

IPAdr udp:192.168.0.248:5060 

Record_Location Indicates the location of the file 

CDR to CSV 


To generate CDR files on the local disk in CSV format, use file:disk in the CDR URL setting and 

specify a string that will be representative of both the filename and the contents of the CDR. The default 

value is $c/$m/$d (cdr/domain/date). 

Table 21-2 shows two examples in which the CDR URL setting has been set so that CDRs will be saved as 

CSV files. This file represents all calls that occurred on Dec. 10, 2008. The second file represents all calls that 

occurred on Dec.11, 2008 between 10–11 a.m. (The $h is replaced by a number between 1 and 24.) You can 

configure the cdr_file_name setting through HTTP also. 

Table 21-2. Setting the CDR URL Setting for CSV Output 

CDR URL Setting File Created 

cdr_file_name=$c/$m/$d.csv cdr/localhost/20081210.csv 

cdr_file_name=$c/$m/$d/$h.csv cdr/localhost/20081211/10.csv 

Notes: 

It is advisable to always use $c as the first variable. 

Each CDR line will be terminated with \r\n. 

The default field separator for CSV files is a comma (,). You can change this by using the cdr_field_separator 

global setting. 

If you want to save the CDRs to a mapped drive, configure the cdr_file_name setting to z:/cdrs/$d. 

txt. 

Example: http:///reg_status.htm?save=save&cdr_file_name=z:/ 

cdrs/$d.txt,where is your system’s address. 


256 

CDR to Email 


To send CDRs to a third party as email attachments, use mailto in the CDR URL setting, and specify an 

email address: 

CDR to Simple TCP 

Several commercial tools are available for collecting CDR information through simple TCP-based communication. 

Each CDR is sent in one line of ASCII text terminated by a CRLF pair over a TCP-based communication 

link. To differentiate the destination from a SOAP CDR, use cdr in the CDR URL setting in front 

of the IP address and the port (the fields are separated by a colon). 

If you want to change the default settings of CDRs on the system, you can modify cdr_format in the 

global configuration file. This file is located in the working directory of the system and is named pbx.xml 

by default. It is a standard XML file, is encoded in UTF-8, and contains configuration information which 

cannot be assigned to any specific domain. Before making changes to the file, be sure to make a backup. 

And once changes have been made, restart the service. 

When customizing the cdr_format string, you can use the fields shown in Table 21-3. Both the short form 

and the long form of all fields are included. You can use either format or switch between the two formats. 

When specifying lengths, put the length of the string between the dollar sign and the character. For example, 

the expression $10c will insert a 10-digit caller-ID into the string. Following is a sample string that specifies 

lengths. The resulting output is also shown: 

Sample String Sample Output 

$w$5e$12c$5d 20071023123024 123 9781234567 120 

Table 21-3. CDR Formats—Short and Long 

Short 

CDR Syntax 

Long 

Description Value 

$$ $$ A dollar sign $ 

$i $(call_id) Caller-ID of call, as seen in the SIP packet 

$v $(call_type) Type of call. Default is attendant* attendant 

$m $(domain) Domain name of the call domain.com 

$l $(lang) Language of the call en 

$f $(sip_from) The From header, as seen in SIP packet 

$t $(sip_to) The To header, as seen in the SIP packet 

$F $(calling) Extension number of the caller 123 

$T $(called) Extension number of the called party 123 

$x $(orig_trunk) Name of originating trunk, if present Trunk 1 

$y $(dest_trunk) Name of destination trunk, if present Trunk 1 

$R $(account) Account that is charged for redirected call 

$r $(redirect_dest) Destination for a redirected call (used only 

if call is redirected) 

$S $(start_time) Start time in seconds 

$C $(talk_duration) Connected duration of the call 



Short 

CDR Syntax 

Long 

Description Value 

$A $(ring_duration) Answered duration of call (useful if the call is ringing/answered 

through queue) 

$E $(hold_duration) Hold duration for the call 

$W $(wait_duration) IVR duration for the call 

$w $(start_date_ 

time) 

Start time (YYYYMMDDHHMMSS) 20071023123224 

$b $(date) Date (YYYYMMDD) (version 2.1.9) 20071023 

$B $(time) Time (HHMMSS) (version 2.1.9) 

$e $(extension) Extension number of call (originating or 

receiving) 

$o $(direction) Direction of call (i for inbound; o for out) i or o 

$c $(remote_call_id) Caller-ID of remote party 

$d $(duration) Duration of call in seconds (inc. hold time) 

$s $(extn_duration) Speaking duration (does not inc. hold time) 

$M $(cmc) Client matter code (CMC) 781299 

$(record_location) The location of recorded files 

* Other values include attendant, acd, ivrnode, hunt, conference, mailbox, extcall, 

starcode, and srvflag. 

Note: The dollar sign has a special meaning. If you would like to produce an actual dollar sign, use two dollar 

signs ($$) as the value. 

Sample Strings 

Following are sample strings that can be used to change the CDR format: 

■ Example 1: cdr_format=$w$m$F$T$o$d$R,$c$v$(record_ 

location)$e,$(cmc) 

■ Example 2: cdr_format=StartTime:$w,Domain:$m,From:$F,To:$T,Dir:$o,Remote 

CID:$c,Type:$v,Rec:$(record_location),Extn:$e 

■ Example 3: cdr_format=$16w$10m$32F$32T$2o$16d$10R$16c$v$(record_ 

location)$4e 

Although example 1 (the short form) is more concise, example 2 (the long form) is easier to decipher. 

The third example reflects specified lengths. 

Making Changes through HTTP 

If you’d prefer not to make changes to the global configuration file, you can use the web interface to create a 

request that will change the cdr_format setting. For example, if you want to change the CDR format of 

a system running on a localhost, you would enter something like this (replace localhost with your snom 

ONE address): 

http://localhost/reg_status.htm?save=save&cdr_format=$w$5d$25m$2o$20F$20T$ 

20R$20r$15c$20f$15v 

You can also use the web interface to make changes to the cdr_file_name setting: 

http://localhost/reg_status.htm?save=save&cdr_file_name= $c/$m/$d.csv 


258 

CDR to SOAP/XML 


The CDR SOAP mechanism can communicate with an external web server. The system starts a SOAP call 

to a URL specified in the system each time a call is completed. The third-party web script which receives the 

SOAP call will then parse the XML packet and file the data in the required place. The storage of the data is 

outside of the system and is the responsibility of the integration team. 

To send CDRs to an external SOAP server, use http:// or https:// in the CDR URL setting and 

specify the IP address and port number: 


Part III 

The User Interface 




WEB INTERFACE 

Chapter 

22 

The web interface is one of the most important differences when comparing modern-day VoIP systems with 

legacy IP-based systems. The web interface allows users to integrate their cell phone and configure other settings 

related to their account. Before users can access the web interface, they will need the URL of the system 

and a user name and password. This information can be sent to users at the time the extension is created 

in the form of a welcome message (other methods can also be used). The Send welcome email setting is a 

domain administrator setting and is mentioned on page 55. 

Logging in 

When users log in, the default of Automatic can be used for the login type; however, if a user has been given 

administrator or domain administrator privileges and wants to log in as a user for any reason, User Login 

must be selected from the dropdown (see Chapter 1 for more information on logging in). 

Figure 22-1. Web Interface Login Screen 

Once the user has logged in, the web interface shown in Figure 22-2 will be displayed. 

Figure 22-2. Web Interface Welcome Page 


262 

These tabs are detailed below: 


■ The Settings tab allows users to change their password, set up redirection, and control their mailbox. 

Voicemail, buttons, and instant message settings are also located here. 

■ The Lists tab allows users to access important lists: email, missed calls, and personal and domain 

address book contacts. Also from the Lists tab, users can schedule conferences. 

■ The Status tab displays a snapshot of the user’s account settings. Users can quickly see their call 

forwarding, hot desking, and agent status. 

The following pages cover the settings that users can make from their own accounts. Other settings that are 

not shown here can also be configured from the user’s account (permission and registration settings). These 

settings are visible only to administrators who log into the user’s account (these settings are detailed beginning 

on page 106). 

Extension Settings 

This section describes all the configurations that can be made from the Settings tab: 

Once the user clicks the Settings tab, the web interface will present the following new options to the user: 

■ General: This is where users can change their password, upload their picture, and specify pickup 

preferences. 

■ Redirection: From this link, users can activate “Do not disturb,” establish treatment for anonymous 

callers, and set call forwarding. Cell phone configuration can also be done here. 

■ Mailbox: Users can enable their mailbox, set a pickup time, assign a maximum number of messages 

that can be sent to their mailbox. 

■ Email: Users can add an email address to their account and establish email account settings. 

■ Buttons: This is where users can configure a button profile. 

■ Instant-Message: From this link, users can send instant messages from the web interface. 

General Settings 

From the Settings tab, the settings shown in Figure 22-3 will be displayed. Administrators can block some 

of these settings to prevent the user from changing them (see Figure 2-3 for more information). 


Figure 22-3. User©s Settings Tab 

The settings shown in Figure 22-3 are detailed below: 

General 

Chapter 22: Web Interface 263 

■ First name and Last name: These fields will be displayed for all internal and external calls (international 

characters are permitted). 

■ Position: This field is for the user’s position/job function in the company. 

■ Company name: This field is reserved for the company name. 

■ SIP Password: The SIP password influences the connection between the VoIP phone and the system. 

Users should not change this password, as it will interfere with access to other registered SIP 

devices (e.g., soft phone). 

■ Web Password: This password is used to access the snom ONE web interface. 

Note: Use a combination of letters and digits and even a symbol to ensure password security. After 

you’ve entered a new password, the current HTTP session will stay valid until your next login. 

Permanent cookies will be invalidated on the next login. 

■ PIN: The user’s PIN will be used in several areas of the system (e.g., when accessing their voicemail 

from an outside phone, when using the calling card account, and when hot desking). For optimum 

security, user’s should be required to use at least five digits. 


264 


■ Timezone: This setting tells the system the user’s time zone. It will affect the time stamps of the 

mailbox and all other time stamps related to the extension. When using the extension from a time 

zone that differs from the domain’s time zone, users will need to override the domain setting. 

■ IVR Language: This setting controls the language of voice prompts delivered by the system, as 

well as the language that is displayed on the LCD of the user’s phone. (Callers will not hear the 

language when calling into the user’s mailbox.) Administrators can add additional languages onto 

the system (see page 204 for instructions). 

■ Web Language: This setting allows users to control the language of the snom ONE web interface 

and in the emails they receive from the system. Users can override this setting during their initial 

login to the web interface. 

■ Upload picture: The uploaded picture will be displayed whenever the user calls another extension 

within the same domain (as long as the other extension has picture-displaying capability). To facilitate 

easier picture upload during the call setup, users should use a thumbnail image size. The format 

of the picture depends on phone model: 

Monitoring 

— For snom 370 phones: Grayscale BMP image. 

— For models that can display color photos: JPEG image. 

The system will display the picture on the landing page of the snom ONE web interface. 

■ Watch the calls of the following extensions: This feature allows users to watch the presence of an 

extension(s). The system will automatically generate configuration files of the extensions specified. 

Note: Wildcard patterns are permitted in this field. If only a star is used, the system will place all 

extensions of the domain into its call-watching list. 

■ Watch the presence of the following extensions (* for all): This setting is similar to the previous 

setting, but it doesn’t provision calls. Instead, it asks the device to subscribe to the presence state 

of the listed extensions. The system will forward the presence state of the phones, but the phones 

themselves will be responsible for generating the presence information. 

■ Watch the following accounts on PAC or WAC: This setting allows users to list the extensions 

they would like to monitor from the WAC. When using the WAC, this setting enables users to join 

a conference room (the conference room must be included as one of the extensions that will be 

monitored). 

■ Permissions to monitor the account: This setting allows users to specify the extensions that are 

permitted to see the status of their extension, including their active calls. By default, all extensions 

are allowed to see the status of other extensions. A user can either list the extensions (e.g., 511, 518, 

523, 522, 513) or use wildcards to match multiple extensions (e.g., 5*). 

Park and Pickup 

■ Explicitly specify park orbit preference: This setting allows users to specify a list of park orbits 

that can be used for parking calls. When park orbits have been entered into this field and the user 

executes the Call Park (*85) feature, the system chooses the first available orbit. When a star (*) is 

entered into this field, the system will have no preference list to draw from, and the user will be 

required to enter the number of the park orbit after entering *85 (see page 283 for more information 

on Call Park). 

■ Explicitly specify pickup preference: This setting allows users to specify the accounts that will be 

picked up when they execute the Call Pickup (*87) feature. This setting is especially useful when 



extensions have been grouped with certain agent groups and hunt groups, as it will prevent the 

wrong calls from being picked up. If a star (*) is entered into this field, the system will have no 

preference list to draw from and will require a park orbit number after the *87 (see page 283 for 

more information on Call Pickup). 

Address Book 

■ Phone address book preference: This setting allows users to upload contacts from the domain 

and personal address books to their phones. Available options are as follows: 

— Both domain and personal addresses 

— Domain addresses only 

— Personal addresses only 

■ Include local extensions in the phone address book: This setting allows users to add extension 

numbers from the system to their phone. 

Miscellaneous 

■ Block outgoing caller-ID: This setting allows the user to control whether their caller-ID is shown 

when making outbound calls. 

■ Enable Call Waiting: This setting allows the user to turn Call Waiting on and off. 

■ Wakeup time: This field allows users to receive automatic wakeup calls from the system. Though 

this feature is used primarily in hospitality environments, it can also be used on an individual basis. 

A wakeup call can also be activated using star code *62. See “Wakeup Call (*62)”. 

Redirection Settings 

Redirection settings allow users to forward their calls to other extensions or phone numbers. These settings 

are detailed below (cell phone redirection settings are detailed in Chapter 25): 


266 


■ Do not disturb: This setting allows users to tell the system to leave their phone silent for all calls, 

both internal and external, and even calls from the hunt group. Only extensions with permission 

to override DND will be able to call the extension. DND override is an administrator setting (see 

Call the following extensions even if DND is set on page 110). An extension on DND can 

place outbound calls at all times. You can also set DND using the star code. See “Do Not Disturb 

(*78)”. 

■ Incoming anonymous calls: This setting controls the treatment of incoming calls; however, it 

does not affect internal calls or calls from a hunt group or agent group. See also the Black List 

(*92) feature on page 292. 

— No special treatment: All calls are sent directly to the user’s extension. 

— Reject Call: The caller will be informed that the user does not accept anonymous calls and 

will not be put through to the extension. 

— Pretend to be busy: This setting prevents the call from going through but does not tell the 

caller why. 

— Ask for name: The caller is prompted for his or her name and then put on hold while the 

extension is called. After the system reads out the recorded name, the user can decide whether 

to accept the call, send it to the mailbox, or reject it. 

— Ask for name even if the caller-ID is present: This setting requests the caller’s name, even if 

the caller-ID is available. Only white list callers will be put directly through. 

■ Hot Desking at: This setting allows users to specify an extension where they will pick up their 

calls. All calls (including calls from an agent group or hunt group) will be sent to this extension. 

For more information on hot desking, see page 286. 

■ Call forward all calls to: This setting allows users to redirect all calls to another extension. Only 

calls that are headed directly to the user’s extension will be affected by this setting. If a user is part 

of a hunt group, agent group, or paging group, calls originating from these groups will not be redirected. 

■ Call forward calls when busy to: This setting allows users to forward all calls to the extension that 

is specified, only when the extension is busy. This condition is true if either the phone itself signals 

that it is busy or if the lines parameter (page 107) has been set for the extension and this number 

has been reached. 

■ Call forward on no answer to: This setting allows users to redirect all calls after there has been no 

answer for a predefined period. The waiting period is defined in the domain but can be overridden 

by the Call forward no answer timeout setting (next bullet item). (If the mailbox picks up earlier, 

this setting will have no effect; if the call is redirected, the mailbox timeout will be cancelled.) 

■ Call forward no answer timeout: This setting determines how long the system will wait for the 

user to answer the phone before forwarding the call (default is 10 seconds). This field is effective 

only if the Call forward on no answer field has been set. 

■ Call forward when not registered: This setting allows the user to specify where calls will be 

forwarded if the user’s extension is no longer registered for any reason. If the user is part of a hunt 

group, agent group, or paging group, calls originating from these groups will not be redirected. 

Cell phone settings are discussed in Chapter 25 beginning on page 306. 


Mailbox Settings 


Every extension has a voice mailbox on snom ONE; however, the administrator can disable users’ mailboxes 

and use an external mailbox, such as Microsoft Exchange. This section applies only to the voice mailbox of 

the system. Mailbox settings are located in Settings > Mailbox and are detailed below: 

■ Mailbox Enabled: From this setting, users can enable or disable their voice mailbox. If it is disabled, 

callers will not be able to leave messages and the user will be unable to use the mailbox functions. 

If Act as park orbit is chosen, those who call the extension will be parked at the extension 

until their call is picked up. This is obviously not useful for extensions that have phones registered 

and make phone calls. 

■ Time until mailbox picks up: This setting allows users to specify how long the system should wait 

before redirecting calls to the user’s voicemail. If Call forward no answer timeout has been set 

(see Redirection settings on page 265), the shorter timeout will be executed first. If Include the 

cell phone in calls to extension has been set (see Redirection settings), the system must be given 

enough time to call the user’s cell phone (which may take 10 or more seconds) and enough time for 

the user to view the caller-ID. The default is 20 seconds. 

■ Maximum number of messages: This setting allows the user to specify the maximum number of 

voicemail messages that should be stored at any one time. If the maximum has been reached and a 

caller wants to leave a new voicemail message, the system will attempt to delete a saved message. If 

space is still unavailable, the system will inform the caller that the mailbox is full. 

■ Announcement Mode: From this setting, users can choose an announcement that callers will hear 

when reaching the their extension. 

— Anonymous Announcement: The caller will hear the number of the extension. 

— Name Announcement: The caller will hear the recorded name of the extension, if one is 

available. 

— Personal Announcement: The caller will hear the recorded announcement of the extension, 

if one is available. 

■ Allow Access for Extensions: This setting allows users to create a group mailbox. A group mailbox 

can be used for anything from a joint boss/secretary mailbox to a place to collect after-hours calls or 

holiday calls. All parties will be notified of incoming messages by the MWI that is located on their 

individual phone (see MWI below). Users can listen to the messages from either their own voice 

mailbox or by dialing into the shared mailbox from an outside phone or an outside line on their 


268 


own phone (see page 297 for instructions on accessing voicemail from an outside phone). Dialing 

directly into the shared mailbox allows users to hear messages that were called into the shared extension 

only (i.e., the messages will not be mixed with the user’s own voicemail messages). Users will 

need the PIN of the shared mailbox when accessing it directly. 

When entering multiple extensions into this field, use a space as a separator (e.g., 101 102). 

■ Send message waiting indication: When this feature is activated, users will receive a message 

waiting indicator (MWI)—a blinking light on their phone—that notifies them of a new message. 

Once the user has listened to the message, a solid light will replace the blinking light. This enables 

the system to alert the user the next time a new message has arrived. Star code *99 will clear the 

message indicator even if messages have not been read. 

■ Call cell phone when new message arrives: If users have decided against forwarding their calls to 

their cell phone but still want to be alerted when a new message arrives on their extension, they can 

activate this feature. Once a caller has left a message, the system will call the user on the cell phone. 

After the announcement, the user must press 1 to hear the message. 

■ Mailbox Escape Account (overrides domain setting): This setting gives users the option of 

changing the direction that a call will take once the caller presses 0 to speak with the operator. The 

extension that is entered into this field will receive the calls that the user thinks are directed to the 

operator. 

The following section of the Mailbox page is visible only to the administrator. From here, greetings 

for the user’s extension can be uploaded by the administrator. 

Email Settings 

The email settings on this page allow users to receive emails as notification of all incoming calls on their extension. 

Users can also request an audio file, or WAV file, of voicemail that has been left on their extension. 

Email settings are located in Settings > Email, and the descriptions of the settings are listed after Figure 

22-4. 

Figure 22-4. User©s Email Settings 



The following information details the settings that can be set from the user’s Email link: 

■ Email Address (e.g. abc@company.com): This feature allows users to integrate their email account 

with their extension account. Voicemail messages will be sent to the user’s email account 

according to the settings chosen below (a semicolon is required when entering multiple email addresses 

into this field). 

■ Send a mailbox message by email: 

— Do not send an email: No email will be sent. This option should be used when the user 

prefers to use only the phone to receive voicemail messages. 

— Send emails without attachments: Email notifications will be sent to the user for every 

voicemail that is received. This provides somewhat of a visual voice mailbox. The user will see 

the name and number of the caller and the date and time that the call arrived. To retrieve the 

voicemail itself, the user will need to call into the voicemail. The benefit of this mode is that 

emails that are relatively short can easily be read using mobile devices that support reading 

emails. 

— Send message as attachment: The user’s actual voicemails will be attached to the email 

notifications. This option should be selected if email is being used throughout the day and a 

personal computer is being used to process the email. 

■ After sending a message: This setting allows users to decide what to do with a voicemail message 

after it has been emailed to them: 

— Keep message as new message: The message will remain in the user’s mailbox as a new message. 

This option has the potential danger of eventually overfilling the voice mailbox. Administrators 

can use the Voicemail Size setting (page 56) to restrict the number of voicemail 

messages. 

— Mark message as read: The message will remain in the user’s mailbox, but the system will 

drop the oldest message to make room for new messages if the mailbox becomes full. The 

disadvantage of this mode is that the message waiting indicator will not alert the user of new 

messages. 

— Delete the message: The message will be deleted after the email has been sent. This keeps the 

mailbox clean, but because the system does not store voicemail messages permanently, the 

user will need to listen to the messages from the email client. 


270 


■ Send email on missed calls: When this setting has been activated, the system sends the user an 

email for each missed call. Calls to a hunt group or agent group are not included. 

■ Send email record for all calls: When this setting is activated, the system sends the user an email 

for all calls. This feature is useful for keeping track of a group sales extension, a telemarketing campaign, 

or calls made to a new employee’s extension. (When both “missed calls” and “all calls” have 

been activated, the user will receive two emails for each missed call.) 

■ Send email on status changes: The user will be notified of all status changes (e.g., toggling the 

DND status, changing the registration status, etc.). 

Instant Message 

The Instant Message feature allows users to display an instant message (IM) on the screen of snom phones. 

Although snom phones cannot generate IMs, they can display them. This feature comes in handy when users 

need to pass information along to someone during a conference call, for example. This feature is located in 

Settings > Instant-Message: 



1. Enter the extension numbers into the Destination field. 

2. Enter your message into the Message field. 


4. Note: When multiple destinations are used, a space should be used as a separator (e.g., 102 103 

208). The system will send the message to all registered devices for the extensions that have been 

entered. 

Call Lists 

This section covers material that is related to call lists only. 

Mailbox 

The Mailbox link allows users to check their voicemail from the web interface. Call details (e.g., the caller, 

time of call, and length of message) are included. The voicemail list can be accessed by going to Lists > Mailbox. 

■ Time: The time the message was recorded (receiver’s time zone). 

■ Number: The caller-ID and name of the person who left the message, if available. 

■ Duration: Length of message (in minutes:seconds format). 

■ Flags: The message can be flagged with the following icons: 

Missed Calls 

Message delete Message marked private 

Message unread Message marked urgent 

Download message 

The Missed Calls list allows users to view calls that were attempted to their extension but did not connect. If 

display names are available, the system will display them here. The length of the list is set by the administrator 

from CDR listing size (page 26). By default, the system keeps call list records for 7 days; however, this 

setting can be customized by the administrator (see Keep CDR Duration on page 26). 

The Missed Calls list displays the following information: 

■ The time when the call was started. 

■ The caller-ID and the name of the person who called 

■ The extension number the caller was trying to reach. 


272 


The following icons are associated with the Missed Calls page: 

Call Log 

One-click dialing: Click to initiate a callback to the party who called you. 

The call will be charged to the user's extension. 

Address book: Click to save the contact in the personal address book. 

The Call Log allows users to access up-to-the-second information for calls made to and from their extension. 

Each entry in the call log contains the following information: 

■ Time the call was started. 

■ Number the call was made from (display names are included when available). 

■ Number the call was made to (display names are included when available). 

■ Call duration (in minutes and seconds). If the call was not connected, the duration will be empty. 

Notes: 

The Call Log displays records of calls beginning with the oldest. The number of records that can be stored in 

the Call Log is limited by the number of lines that can fit on a user’s computer screen. 

From the Missed Calls page, icons are available that allows the user to initiate a callback to the party who 

called and to save a caller’s information to the personal address book. 

Click-to-dial: Allows the user to initiate a call to the calling party. When the following dialog 

screen is displayed, the user needs to answer the phone. The system will call the requested extension 

or phone number. (The call will be charged to the user’s extension.) 


Contacts 


Address book: Allows users to add contacts to the personal address book: 

Address books allow users to see who’s calling. They can also be used to expedite billing (when a CMC is 

used) and to speed-dial contacts. The system offers two types of address books: a personal address book, 

which is private for the extension, and a domain address book. The latter is visible to the entire domain but 

is not editable by its users. This section covers both the personal and the domain address book. 

Personal Address Book 

Users can access the personal address book by logging into the web interface and going to Lists > Address 

Book as shown below: 

From this page, users can edit and delete existing address book entries and create new entries. 

Create Address Book Entries 

Users can add a contact to their personal address book using two different methods. 

Method 1—Using the Call Log: If the contact is in the user’s Call Log (i.e., contact has been made by 

either the user or the caller), the following instructions apply: 

1. Click Call Log from the Lists tab (shown below). 

2. Click the Address Book icon. 


274 


The contact will be added to the user’s address book: 

3. If you add a contact whose information is incomplete, click the account number and populate the 

necessary fields: 

■ First Name: Enter the first name of the contact if one was not available. 

■ Last Name: Enter the last name of the contact if one was not available. 

■ Number: Enter the telephone number of the contact (this is not a SIP URL). (This field will 

already be populated if this contact was in your Call Log.) 

■ Speed Dial: Enter a 2-digit star code (e.g., *12) to speed-dial this contact. 

Note: Ensure that the star code does not overlap with any of the standard star codes. 

■ CMC: The client matter code identifies the customer and is used to expedite billing in offices that 

bill their clients for phone calls (e.g., a lawyer’s office). 

■ Contact Type: Using the dropdown list, identify the contact type. 

— Regular Contact 

— White List: The white list usually contains contacts that are trusted. If the flag is set to 

White List, the contact becomes part of the white list and gets preferred treatment. Contacts 

on the white list are never intercepted by the auto attendant and will not need to state their 

name. Callers on the white list are allowed to receive a callback when the extension becomes 

available. 

— Black List: The black list usually contains the list of contacts users would prefer to avoid. If 

a caller is on the black list, the user’s anonymous call treatment settings will determine how 

that call is treated (see Redirection). If the caller should be blocked, then the system will 



block that call. Otherwise, the system will ask the person to leave his name before calling 

your extension. Black list contacts are not allowed to camp on an extension. 

4. Enter comments that are relevant to the contact. 


Method 2—Using the Address Book: If the contact is not in the user’s Call Log, users can add entries 

to their address book by going to Lists > Address Book and populating the following form using Steps 3 

through 5 above: 

Note: Contacts can also be added to the address book by using the black list (*91) and white list (*92) star 

codes. If the number does not exist, the system will automatically create an address book entry for the number. 

Users can edit the entry by going to Lists > Address Book, then clicking Edit. 

Edit Address Book Entries 

1. From the Lists tab, click Address Book. 

2. Click the edit icon for the entry you would like to edit. 

3. Make any necessary changes. 


Delete Address Book Entries 

1. From the Lists tab, click Address Book. 

2. Click the delete icon next to the entry you would like to delete. 

Upload Address Book to Phone 

The administrator can enable the setting shown below so that users can upload both the personal and domain 

address books to their phone. 

Domain Address Book 

Domain address book entries are visible to and searchable by all members of the domain. Users can access 

the domain address book by going to Lists > Domain-Addresses: 


276 


The address book can also be used to indicate which DID number has been dialed. If the user adds an address 

book entry that matches a specific DID, the system will add the display name to the caller-ID and the 

phone can then display the text associated with the caller-ID. 

Upload Domain Address Book to Phone 

The administrator can enable the setting shown below so that users can upload both the personal and domain 

address books to their phone. 

Extension Status 

The Status tab shows the current status of the user’s extension. It displays redirection settings, DND status, 

last number dialed, and many other settings. Users can view their status settings by clicking the Status tab: 

The Status tab addresses the following settings: 

■ Do not disturb: This field will be set to true if the extension is on DND mode. 

■ Agent logged in: This field shows whether the extension is logged into an agent group. 



■ Hot Desking at: If the extension has redirected all calls (including calls from hunt groups and 

agent groups) to another location, the destination is shown in this line. 

■ Call forward all destination: If the Call forward all calls setting has been set, the destination is 

shown here. 

■ Call forward busy destination: If the Call forward calls when busy has been set, the destination is 

shown here. 

■ Call forward on no answer: If the Call forward on no answer setting has been changed, the destination 

is shown here. 

■ Timeout for call forward on no answer: If the Call forward no answer timeout has been 

changed, it will be indicated here. 

■ Timeout for mailbox: If the Time until mailbox picks up setting has been changed, the change 

will be indicated here.Call redial: This entry reflects the last number that was dialed from the user’s 

extension. The Redial a Number (*66) star code can be used to redial the number. 

■ Call return: This entry reflects the last number that dialed the user’s extension. The Call Return 

(*69) star code can be used to return the call. 


278 



STAR CODES 

Chapter 

23 

Star codes are two-digit numbers prefixed with a star (*) and are used to instruct the system. Each star code 

maps directly to an action of the system, allowing users to quickly access snom ONE features, such as Do 

Not Disturb (DND), call forwarding, and a host of other features. All users across the domain have access 

to the same set of star codes and can access the list by logging into their account and going to Lists > Star 

Codes. 

This chapter provides instructions on how to use each star code with snom phones. In most cases, once the 

star code has been entered, the confirmation button (the confirm button) must be used to indicate that the 

dialing is complete. 

Administrators may define their own plan for star codes, and for a few situations, the star codes will need to 

be modified. For example, when configuring an Agent Log In/Agent Log Out toggle button, the star codes 


280 


must be identical (e.g., *64/*64). This applies to Call Park and Call Park Retrieve also (e.g., *86/*86). Star 

codes must begin with a star and include only two digits. Codes 01–60 are reserved for speed dialing. 

Figure 23-1 displays the list of star codes that are available on snom ONE. 

Making Calls 

*66 

*69 

*90 

Redial 

Call Return 

Intercom 1 

Transferring Calls 

*77 

*85 

*86 

*87 

Transfer 2 

Call Park 3 

Call Park Retrieve 3,5 

Call Pickup 3,5 

Anonymous Calls 

*67 

*68 

*88 

*89 

Block CID (Activate) 

Block CID (Deactivate) 

Block Anonymous Calls 

(Activate) 


(Deactivate) 

Call Forwarding 

*70 

*71 

*72 

*73 

*74 

*75 

*76 

*78 

*79 

*80 

Hot Desking 

Call Forward All (Activate) 2 

Call Forward All (Deactivate) 

Call Forward Busy (Activate) 2 

Call Forward Busy (Deactivate) 

Call Forward No Answer (Activate) 2 

Call Forward No Answer (Deactivate) 

DND (Activate) 

DND (Deactivate) 

Set Night Mode for 

Domain Accounts 3 

Voicemail 

Send Voicemails as Emails (Activate) 

Send Voicemails (Deactivate) 

Go to Voice Mail 

Record New Greetings 

Clear the Message Indicator 

Call Center 

3, 5 

Agent Log In 

3, 5 

Agent Log Out 

Call Barge In 1 

Teach Mode 1 

Listen In 1 

Cell Phones 

Retrieve Call from Cell Phone 

Move Current Call to Cell Phone 

Call Cell Phone of the Extension 1 

Miscellaneous 

Show Account Balance 

Wakeup Call 

Request Call Details 

Clean up an Extension 

Add to White List 

Add to Black List 

Record ON 4 

Record OFF 4 

1. An extension number is required after these star codes. 

2. A destination number (e.g., an extension or an external number) is required after these star codes. 

3. An argument is optional for these star codes. 

4. These codes are used during a conversation; the tones are audible and can be heard by the connected party. 

5. These star codes must be identical to one another (e.g. *64/*64 and *86/*86) when configurated as a toggle button. 

Figure 23-1. snom ONE Star Codes 

Basic Star Codes 

Redial a Number (*66) 

*95 

*96 

*97 

*98 

*99 

*64 

*65 

*81 

*82 

*83 

*51 

*52 

*00 

*53 

*61 

*62 

*63 

*84 

*91 

*92 

*93 

*94 

Conference 

Redial (*66) allows users to redial the last number that was called from their phone. The Call Log (page 

272) can also be used to dial the last number dialed. 

* 6 6 


Call Return (*69) 

Chapter 23: Star Codes 281 

Call Return (*69) will dial the number of the call that was received last. The number will be available until 

a call has been established to that number—not necessarily connected, which can help prevent users from 

calling back the same person twice. (If the user executes *69 and discontinues the call during its ring state 

though, *69 will not be able to access that number again unless the caller rings again.) 

Notes: 

* 6 9 

If a user reaches an external mailbox, the system will handle the Call Return as if the call had been successfully 

established and will clear the number. If the user reaches a mailbox of another extension on the system, 

the Call Return number will not be cleared and the user will be able to make successive attempts at reaching 

that extension by dialing *69. The system will store a Call Return number only for calls that contain a valid 

caller-ID. 

Telephones that share the same extension number will share the same redial and Call Return number. 

The Call Log (page 272) can also be used to dial the last number received. 

Intercom (*90) 

The Intercom feature mimics the old secretary-boss intercom systems and can be thought of as a one-part 

page with two-way audio. Intercom is used to communicate with one other person. The other party must 

have a separate extension number, as the Intercom feature will not work between two phones that are registered 

to the same extension. The Intercom feature will also not work when multiple registrations are involved 

with the receiving extension. 

To activate: 

1. Set the user’s Call Permission settings for intercom (see page 110). 

2. Dial *90 and the extension you would like to intercom. 


* 

9 0 51 2 3 

You can now begin conversing with the other party. 


Transfer Calls 

Calls can be transferred using the Transfer button on the phone or star codes. The system supports two 

types of call transfers: blind transfer and attended transfer. 


282 

Blind Transfer 


When the user activates a blind transfer, the call is transferred with no knowledge as to whether the receiving 

party will be available to receive the call. A blind transfer can be used for transferring calls to voicemail or the 

cell phone. 

1. Press the Transfer button on your phone, or press Hold and dial *77. 

2. Enter the extension number (or phone number) of the recipient. 

A blind transfer offers no confirmation of call connectivity. If the number is busy or does not exist or no one 

picks up, the user will not be notified. 

Attended Transfer 

When using attended transfer, the user must speak with the party who is receiving the call to ensure that the 

call can be received. 

1. Put the caller on hold (using the phone’s Hold button). The LCD will display the call that is on 

hold. 


3. Once the party on the receiving end is prepared to take the call, press the Transfer button twice (the 

first Transfer connects the call; the second Transfer clears the music so that the parties can speak. (If 

the phone does not have a Transfer button, *77 can be used to transfer the call.) 

Users do not need to press the Hold button to free the call. If the person is not available to take the 

call, the user can press the Hold button to reclaim the call and discuss the available options with the 

caller. 

Hold 5 0 58 Transfer Transfer 

Transfer a Call Directly to Someone’s Voicemail 

The systems administrator must enable a direct dial prefix before users can “blindly” transfer a call to another 

user’s voice mail. The default for direct dial is “8” (see page 57 for details on this setting). 

1. Put the call on hold using the Hold button on your telephone keypad. 

2. Press the Transfer key. (If your phone does not have a Transfer button, dial *77 to transfer the call.) 

3. Enter the mailbox prefix number. 


Hold 

Transfer 

*77 

8 

1 2 53 

The caller will be transferred to the extension’s voicemail. Because the caller will bypass the voicemail 

timeout, the caller will not hear any ringing. This method does not need to involve an outside caller. 

To call a person’s mailbox directly, the user can dial 8123. 

8 

1 2 53 


Call Park (*85) and Call Park Retrieve (*86) 


Call Park allows the user to send a caller to someone whose whereabouts (within paging proximity) are not 

known. The receiving party uses Call Park Retrieve (*86) to retrieve the call. 

Scenario: A call comes in for Diane, but you do not know where she it, so you put the customer on hold, 

dial *85, and press the confirm button. You then page Diane, “Diane, you have a call on extension 509” 

(509 is the extension on which the call was received). Diane executes the Call Park Retrieve (*86) feature, 

dials 509, and retrieves the call. Park orbits can also be used for parking calls (rather than using your extension). 

When using a park orbit, enter the extension number of the park orbit after *85. 

Park a call on your own extension 

1. Put the call on hold (music plays in the background). If your phone does not have a Hold button, 

refer to the user guide that came with your phone. 

2. Press *85 to park the call, then press the confirm button on your telephone keypad. 

Hold * 

8 

5 

An announcement will indicate that the call has been parked. At this point, the call can be retrieved 

by any extension by dialing *86 and the extension number on which the call has been parked. 

* 8 6 1 2 3 

If the call is not picked up within a specified time (default is 1 minute), the user will receive a call 

from the system as a reminder of the waiting call. 

Park a call on a specific extension: 

The system administrator may configure specific park orbits (e.g., 800, 801, 802, and 803) on the system so 

that all users can park calls to the same place. This is useful for monitoring parked calls and buttons, as well 

as for applications that need to park calls. Park orbits also ensure that voicemail does not pick up. 

1. Put the call on hold (music will play in the background). 

2. Press *85 to park the call. 


Hold 

1 52 

5* 

53 

8 

5 

An announcement will indicate that the call has been parked. At this point, the call may be retrieved 

by any extension (which has permission) by dialing *86 and the extension number on which the call 

has been parked. 

5* 

1 52 

Call Pickup (*87) 

8 

53 

6 

Call Pickup allows users to pick up a ringing phone. It can be used to pick up calls ringing into a specific 

extension or for ringing calls in general, including calls ringing into a hunt group or agent group. 


284 


Note: When compared to Call Park Retrieve, Call Pickup is associated with a sense of urgency (i.e., the user 

must pick up the ringing phone before the call goes to voicemail.) With Call Park Retrieve, the caller is 

already in the system, so there is no hurry. 

Before users can use Call Pickup, they must first put their last call on hold so that the system does not interpret 

and execute the star key during calls made to external systems that also use star keys. 

Directed Pickup 

1. Dial *87. 

2. Dial the extension on which the call is ringing. 


* 8 7 

51 2 3 

General Pickup 

Call Pickup can also be used without specifying an extension. To activate, dial *87 and press the confirm 

button on the telephone keypad. 

* 

8 7 

The system searches for calls to pick up in the following order: 

■ Hunt groups: For a call that is headed for a specified hunt group, the system will reroute the call 

to the extension that dials the pickup code. If several calls ring the hunt group, then the system will 

pick up only the first call that was routed to the hunt group. 

■ Extensions: When the account number is an extension, the system searches for calls that go directly 

to the specified extension. These calls can come from an auto attendant or through direct 

extension dialing. Calls that ring the extension due to it being part of a hunt group or agent group 

will not be affected by Call Pickup. 

■ Agent groups: As with the hunt group, the system picks up a call from the specified agent group. 

For calls in the ringing state, the system picks the first call that is in that state. Otherwise, the system 

picks the call that entered the queue first. 

Conferencing 

This section covers conference-related star codes and the three-way conference that users can be set up from 

the phone. For information on others types of conferencing, see Chapter 12. 

Three-Way Conferences 

Users who have Hold and Conference buttons can do a three-way conference using the phone: 

1. Put the initial party on hold by pressing the Hold button on the phone. 

2. Dial the extension or external number of the new party, and press the confirm button on the telephone 

keypad. 

3. When the new party picks up, press the Conference button to merge the calls. 


Conf. 

Hold 

9 

7 8 51 

2 3 4 5 6 57 


Note: This method of three-way conferencing may affect the network usage and call quality since numerous 

calls are being processed and merged together. 

Conference (*53) 

The Conference (*53) feature allows users to quickly bring someone into an in-session conference, even 

though the person is not part of the participant list. The new party can be on a cell phone or regular phone. 

The benefit of this feature is that the new party is required to do nothing but answer the phone (conference 

number and access code are not needed). The party in charge of the conference simply needs to call the new 

party (from a line that is separate from the conference) and use *53 to join the party to the conference. 

Scenario: You’re in the middle of a conference call and you realize you need the input of someone who is not 

present in the conference. The individual happens to be traveling to a meeting, so you call the person on his 

cell phone and then connect him to the conference using the Conference feature. 

To bring the party into the conference: 

1. Call the party from a different phone line. 

(Use the *00 feature if you need to access the individual’s cell phone but do not know what that 

number is.) 

2. Place the call on hold. 

3. Enter *53, then press the confirm button on the telephone keypad. 

Line 2 

9 

7 

* 5 53 


Block Caller-ID (*67) 

8 

1 2 53 

4 

5 6 57 

Block Caller-ID (*67) allows users to keep their phone number and name hidden from the parties they call. 

It also prevents called parties from looking the user up in an address book and initiating a callback. Calls 

from one extension to another extension will not be affected by *67. These calls will always reflect the caller- 

ID. 


* 

6 7 

The user will hear an announcement indicating that the caller-ID will be blocked for all future calls. 

Note: Caller-ID is always presented for internal calls. 

2. Dial *68 to re-enable your caller-ID. 

* 

6 8 


Hold

286 


Reject Anonymous Calls (*88) 

Users can reject anonymous calls by dialing *88 and pressing the confirm button. 

* 

8 8 

To re-allow anonymous calls, users must dial *89 and press the confirm button. 

Note: If a user has set Call Forward to busy condition (page 266), the system will forward anonymous calls 

to the designated destination (e.g., an assistant). This will ensure that the user’s anonymous calls are screened 

before the user takes the call. If the user has not set Call Forward to busy, the system will announce to the 

caller that the call cannot be taken because of the blocked caller-ID. 

Other options are available for handling anonymous calls, such as screening the calls or using the Ask for 

name setting (page 266). 

Forwarding Calls 

Call forwarding allows users to forward their calls to another number. Multiple methods are available for 

doing this. 

Hot Desking (*70) 

Hot desking can minimize the need for office space and is often used as a cost-reduction method. It allows 

multiple users to use the same desk and phone, although at different times. Hot desking can be used on a 

temporary or permanent basis. In either case, once the user activates the hot desking feature, all calls made to 

the user’s extension (or as part of a hunt group or agent group) will be routed to the new extension or destination. 

Outbound calls from the hot desking phone will reflect the user’s caller-ID. 

It is not expected that other significant inbound traffic is routed to the originally registered extension. This 

fact suggests that offices use “virtual” and “real” extension numbers: 

■ Real extension numbers are used for employees with a fixed location (e.g., switch board, management). 

■ Virtual extensions are not registered. They are used simply for routing calls to a specific user. 

Note: If users would like to route their calls to a colleague who is covering for the day, they can use either the 

Redirection star code (*71) or the redirection settings from the web interface (page 265). 

To log into a hot desk: 

Note: Users must set their voicemail PIN before they can use the Hot Desking feature. 

1. Go to the phone where you will be hot desking. 


3. Enter your extension number when prompted. 

4. Enter your voicemail PIN code for the extension number when prompted. 

* 7 0 

1 2 3 

PIN 


The system will acknowledge that the Hot Desking feature is in service. 


Note: The system will automatically log a user out of a hot desk each night. Also, after activating 

the hot desk feature, users will be notified via email that a status change has been initiated on their 

extension. 

To log out of a hot desk: 

To log out of a hot desk, users must dial *70 and their extension number from either the location that is currently 

registered as a hot desk or from a phone that has been configured to their extension. The system will 

acknowledge that hot desking is no longer in service. 

* 7 0 

1 2 3 

Note: Hot desking has limitations. Telephone preferences, such as ring tones, address book programming, 

etc., cannot be moved to another desk. 

Call Forward All Calls (*71/*72) 

The Call Forward All (*71) feature allows users to redirect all calls to another number (e.g., to a colleague 

who is answering the user’s phone). 


2. Enter the call forwarding number once the announcement is complete. 

3. Press the # key. 

59 

* 

7 8 

7 1 

1 2 3 4 5 6 7 # 

The system will repeat the number and then hang up. If you want to change the forwarding number, 

just call *71 again. 

4. To deactivate, press *72, then press the * key when prompted. The announcement will indicate that 

call forwarding has been turned off. 

Users do not need physical access to their phone to modify this setting. This setting can also be activated 

from the web interface (page 266). 

Call Forward on Busy (*73/*74) 

The Call Forward on Busy (*73) feature allows users to forward calls to another number if their extension 

is busy. This feature is typically used for redirecting calls to a team member or colleague to ensure that all 

incoming calls are answered. To activate: 



3. Press the pound key (#). 

59 

* 

7 8 

7 3 

1 2 3 4 5 6 7 # 


288 


The system will repeat the number and then hang up. If you want to change the forwarding number, 

just call *73 again. 

4. To deactivate, dial *74. The announcement will indicate that call forwarding has been turned off. 

Users do not need physical access to their phone to modify this setting. This setting can also be activated 

from the web interface (page 266). 

Call Forward on No Answer (*75/*76) 

The Call Forward on No Answer feature allows users to redirect calls when the extension has not picked up 

after a certain time. This feature is typically used to redirect a user’s incoming calls to an assistant or secondary 

phone. To activate: 



3. Press the pound key (#). 

59 

* 

7 8 

7 5 

1 2 3 4 5 6 7 # 

The system will repeat the number and hang up. If users want to change the forwarding number, 

they must dial *75 again. 

4. To disable this feature, dial *76. The announcement will indicate that call forwarding has been 

turned off. 

The length of time the system will wait before forwarding the call is determined by the Call forward on no 

answer timeout setting on page 58. Keep in mind that the voicemail timeout (see Time until user’s mailbox 

picks up on page 56) works in tandem with the no-answer timeout. If the voicemail timeout kicks in 

before the no-answer timeout, the call will go to voicemail. Both of these settings can be set at the domain, 

but users can override these settings (page 267) if administrators have not blocked user access (Figure 2-6). 

Do Not Disturb (*78) 

Do Not Disturb or DND allows users to silence their phone for all incoming calls, even calls made to a 

hunt group or agent group. When DND is activated, all calls will stop, including those redirected to the cell 

phone. However, the system will still call the user’s cell phone though if a message has been received while 

DND is activated and the user has activated Call cell phone when new message arrives (page 268). 

1. To activate DND, dial *78 and press the confirm button on the telephone keypad. 

* 

7 8 

2. Dial *79 to deactivate. 

Notes: 

If the DND button on the phone is used to activate DND, this will not stop the user’s cell phone from 

ringing. It will affect only that phone. To stop all phones from ringing, the user will need to activate the 

*78 star code so that the system will know to put all phones that have been configured to the extension into 

DND. DND can be overridden by someone who has DND override permission. See Call the following 

extensions even if DND is set on page 110. Typically, this is a secretary who needs access to the boss, 

regardless of DND status. 



DND takes precedence over Call Forwarding, as the latter cannot prevent hunt group calls from reaching 

the user’s extension. 

Set Night Mode for Domain Accounts (*80) 

This feature allows users to forward domain accounts (e.g., auto attendant, hunt group, etc.) to their 

own extension. Before this feature can be used, permission for each account must be activated by the 

administrator. Permission is set from the extension's Permission tab (page 109). 

To forward all auto attendants, hunt groups, and agent groups of the domain, *80 must be used without an 

argument: 

5* 8 0 

5 3 3 # 

To forward only a specific account, the account must be specified after the star code: 

5* 8 0 

5 3 3 # 

7 7 7 

To disable this feature, enter a star in place of the call forward number: 

5* 

5* 

8 

Call Center Features 

0 

Call monitoring features must be used with caution. Please consult corporate and government regulations 

to determine whether these features are permitted in your environment. Illegal listening to phone calls is a 

severe crime, and system administrators must be aware of this. 

Many call center features are available only if extensions have been given permission to use them. These settings 

are controlled by the domain administrator and are located in the user’s Permission tab (page 109). 


290 


Agent Log in/Log out (*64/*65) 

The Agent Log In/Log Out star codes allow agents to log in and out of agent groups. Agents can use these 

codes to log themselves in at the beginning of a shift and then log back out when finished. Before agents can 

activate the log in/log out (*64/*65) features, agents must be given permission by the administrator. Agent 

groups are discussed in detail in Chapter 10. 

To log out of all agent groups: 

* 

* 

6 4 

6 5 

Log in 

Log out 

Agents who are members of numerous agent groups will need to specify which agent group they would like 

to log in or out of. Otherwise, they will be logged out of all agent groups. Using the sample shown below, 

the agent will be logged out of agent group 666 and will retain logged-in status for all other agent groups. 

* 

6 5 

Call Barge (*81) 

6 6 6 

Log out of 666 

Call Barge allows a user to barge into an existing call between two people. Typically, the person barging into 

the call is either a secretary reminding the boss about another appointment or a co-worker in the next cube 

needing help on a call. Both parties can hear the third person come into the call and can hear the third person. 

Users cannot activate Call Barge unless permission has been given by the domain administrator (page 

109). 

To activate Call Barge: 

1. Dial *81 then enter the extension you would like to interrupt (e.g., *81508, where 508 is the extension). 

2. Press start on the telephone keypad. 

* 8 1 1 2 3 

Call Teach Mode (*82) 

Call Teach mode allows a third party to call into an existing call and communicate with one of the participants 

without the other person knowing. This is typically useful in a call center when a trainer wants to offer 

tips to a new agent without the customer knowing. This mode is sometimes referred to as “whisper mode” 

because the agent’s phone must have an especially good echo cancellation so that the customer hears no 

background echo. Users cannot activate Call Teach unless permission has been given by the domain administrator 

(page 109). 

To activate Call Teach mode: 

1. Dial *82, then enter the extension number that should hear your voice. 

2. Press the confirm button from the telephone keypad. 

* 8 2 1 2 3 


Listen In (*83) 


Listen In mode allows a third party to listen in without being detected. The two parties speaking to each 

other are not notified about the listen-in and have no idea that a third party is on the phone. Users cannot 

activate Listen In unless permission has been given by the domain administrator (page 109). 

To activate Listen In: 

1. Dial *83, then enter the extension number that you would like to monitor. 

2. Press the confirm button from the telephone keypad. 

Miscellaneous 

* 8 3 1 2 3 

Show Account Balance (*61) 

This feature allows users to hear the balance of their pre-paid calling card account. To get the balance, users 

simply enter *61 (followed by the confirm button) from the extension the pre-paid card is configured to. 

For more information on the pre-pay feature, refer to Chapter 15. 

Wakeup Call (*62) 

The Wakeup Call (*62) feature allows users to receive automatic wakeup calls from the system. Though this 

feature is used primarily in hospitality environments, it can also be used on an individual basis. 

1. Dial *62. 


3. Enter the wakeup time in 24-hour format (HHMM); for example, 0715 (for 7:15 a.m.) and 1915 

(for 7:15 p.m.). 

* 6 2 

0 7 1 5 

The system will confirm the wakeup call. To cancel a wakeup call after it has already been confirmed, 

users must press the star (*) key. 

Note: Wakeup calls cannot be transferred to other registered devices (i.e., the wakeup call will not 

ring the user’s cell phone). 

Request Call Details (*63) 

The Request Call Details (*63) feature allows users to request the details of a call (caller-ID, call duration, 

and time of call). The information will be sent to the user’s email address. This feature is handy and prevents 

the user from asking for contact information from the caller. 

To activate the Request Call Details feature: 


* 

6 3 

2. Retrieve the message in your email. 


292 


Clean up an Extension (*84) 

This feature allows users to quickly clean up an extension. This makes it ideal in hospitality environments, as 

it can be used to clear an extension of its messages after guests have checked out and before new ones arrive. 

It can also be used to clean up extensions of employees who have left the company. Users cannot activate 

this feature unless permission has been given by the domain administrator (page 109). 

Add to White List (*91) 

This feature allows users to add a contact to the white list of their personal address book. White list contacts 

receive preferred treatment. They are not required to say their name before speaking with the user, and they 

can receive a callback when the user’s extension becomes available. 

1. Dial *91. 


If the number does not already exist in the user’s address book, the system will automatically create an entry 

for the number. 

* 

9 1 

Users can edit the address book entry from the web interface (page 275). 

Add to Black List (*92) 

This feature helps users avoid unwanted contacts (e.g., unwanted callers, fax spammers, etc.). Once a caller is 

on the black list, the user’s Incoming anonymous call setting will determine how that call is treated. If the 

caller should be rejected, then the system will reject that call. Otherwise, the system will carry out whatever 

instruction the user has configured. The Incoming anonymous call setting is detailed on page 266. 

1. Dial *92. 


* 

9 2 

If the number does not already exist in the user’s address book, the system will automatically create an entry 

for this number. Users can edit the address book entry from the web interface (page 275). Black list contacts 

are not allowed to camp on an extension. 

Call Record (*93/*94) 

Call Record allows users to record a conversation between themselves and another party. This feature is 

handy when it is critical that a user remember the contents of a particular call. Users should check state and 

federal laws before recording calls. To activate the record feature, dial *93 on the telephone keypad. To deactivate, 

dial *94. 

* 9 3 

* 9 4 

Record ON 

Record OFF 

The recording can be retrieved from the user’s mailbox from the web interface (Lists > Mailbox), as shown 

below. 




294 



VOICEMAIL 

Following is a brief overview of what users can do with their voicemail account: 

Table 24-1. Voicemail Features 


Multiple 

announcement modes 

Chapter 

24 

Users can choose which type of announcement callers will hear when calling 

their extension. Choices are anonymous announcement, name, or a personal 

greeting 

Users can access their voicemail from their desk phone, cell phone, or any external 

phone. 

Users can specify the number of seconds that their extension should ring before 

voicemail picks up. 

Convenient voicemail 

accessibility 

Voicemail 

timeout 

Shared mailbox access If the extension is being used as a group mailbox, all parties can access the voicemail 

from their own extension. 

Cell phone 

This feature allows users who have chosen not to forward calls to their cell 

notification of new phone to still receive a voicemail notification on the cell phone when a new 

voicemail 

message arrives. 

Email notification of This feature allows you to receive your voicemail as email attachments. 

new voicemail 

Transfer a caller to If the systems administrator has enabled a direct dial number, users will be able 

another extension’s to “blindly” transfer a call to someone else’s voicemail. Because the caller will 

voicemail. 

bypass the voicemail timeout, the caller will not hear any ringing. 

Setting up the Voicemail Account 

Changing the PIN 

Users will need a voicemail PIN in several areas of snom ONE (e.g., when accessing their voicemail from an 

outside phone, when using the calling card feature, and when hot desking). Domain administrators can set a 

requirement for the minimum number of digits that users will be required to use for a PIN (see page 57). At 

least four or five digits is recommended. The PIN can be changed by the user as follows: 

1. Dial extension number (if messages are waiting, press * to skip to the main menu). 

2. Press 2. 

3. Enter new access code. 

4. Press 1 to confirm. 

Note: Users can also set their PIN from the web interface (page 263). 


296 

Record Your Name 


This recording will be used in announcements to all callers. Users can choose not to record their name. The 

system will recite the extension number instead. The user’s name can be recorded as follows: 

1. Dial your extension number (if messages are waiting, press * to skip to the main menu). 

2. Press 3. 

3. Begin recording your name. 

4. Press the # sign. 

From here, users can listen to the recording (press 1), choose to use the recording (press 2), record 

again (press 3), or delete the recording and return to the previous menu (press 4). 

Record a Personal Welcome Message 

1. Dial extension number (if messages are waiting, press * to skip to the main menu). 

2. Press 4. 

3. Record your personal greeting. 


From here, users can listen to the recording (press 1), choose to use the recording (press 2), record 

again (press 3), or delete the recording and return to the previous menu (press 4). 

Accessing the Voicemail System 

The main menu of the voicemail system is shown in Figure 24-1: 

1. Hear Your Messages 

1 

2 

3 

4 

5 

6 

7 

8 

9 

0 

Start reverse playback 

Stop/resume playback 

Start fast-forward playback 

Hear the message again 

Play message with envelop 

information (date, caller-ID). 

Move or copy 

To move message, press 1. 

To copy message, press 2. 

To record a comment, 

press 3. 

Delete 

Call back 

To confirm the callback, 

press 1. 

To cancel the callback, 

press *. 

Skip 

Help 

Figure 24-1. Main Menu 

2. Change Your PIN 

Enter your new access code. 

1 

* 

Use the access code. 

Return to the main menu 

without using the new 

access code. 

3. Record Your Name 

Record your name and press 

the # sign. Then choose from 

the following options: 

1 

2 

3 

4 

Listen to recording 

Use recording 

Record again 

Delete recording 

4. Record a Greeting 

Record your greeting* and 

press the # sign. Then choose 

from the following options: 

1 

2 

3 

4 

Listen to recording 

Use recording 

Record again 

Delete recording 

*You can record up to five 

greetings. 

Star code: *98 

5. Record a Message 

Record your message and 

press the # sign. Then choose 

from the following options: 

1 

2 

Move message 

Copy message 

9. Select a Mailbox 

Greeting 

Choose a greeting from the list 

provided by PBX . You can have 

up to five choices: 

1 

2 

9 

Greeting 1 

Greeting 2 

. 

. 

. 

Greeting 5 


Users can access the voicemail system using any of the following methods: 

Chapter 24: Voicemail 297 

Listen to the messages from your main phone: Dial *97 and press the confirm button on the telephone 

keypad. (Another way to listen to your messages from your phone is by dialing your extension number.) 

Listen to the messages from your cell phone: Dial into the main number and press 3 once you hear the auto 

attendant. The auto attendant will recite your messages and then deliver the prompts shown in Figure 24-1. 

Listen to the messages from an outside phone: 

1. Dial into the main number. 

2. From the auto attendant, enter your extension number or if you know the direct dial prefix (8 is 

default), dial the prefix plus your extension (i.e., if your extension is 511, then enter 8511). 

3. Press the star (*) key; otherwise, you will be prompted to leave a message. 

4. Enter your voicemail PIN. 

Example: Main # 8 5 1 1 * PIN 

When users call their mailbox for the first time, they will be prompted to record their name. The recording 

will be used when someone calls their extension from the auto attendant. Users will also be prompted to 

record a greeting, which they can opt to use in place of their name. Once this has been established, anytime 

users call into their mailbox, the auto attendant will automatically recite their messages before delivering the 

prompts shown in Figure 24-1 (users can press the * key to skip their messages and go straight to the voice 

prompts). 

Forwarding a Message 

To forward a message to another recipient while listening to the message: 

1. Press 6. 

2. Press 1 to move the message to a single mailbox. 


4. Press #. 

To copy a message to another recipient(s) while listening to the message: 

1. Press 6. 

2. Press 2. 

3. Enter the extension number. When copying to multiple extensions, separate them by the # sign. 

4. Press ## when finished. 

Leaving Voicemail Messages 

When users call another user, they have the option of receiving a callback (press 1) or leaving a message 

(press 2). Once they leave a message and press the # key, the options shown in Table 24-1 will be available. 

Table 24-1. Options When Leaving a Voicemail Message 

Key Feature 

1 Delete the message. 

2 Record the message again. 


298 


Key Feature 

3 Mark the message as urgent. 

4 Mark the message as private. 

0 Send the message and go to the operator. 

9 Send the message and hang up. 

Composing a New Message 

Rather than leave a message directly on another user’s voicemail, users have the option of composing a message 

and then moving it to another mailbox or copying it to multiple mailboxes. 

1. Call into your voice mailbox. 

2. If you have messages waiting, press * to skip to the main menu. 

3. Press 5 from the main menu. 

4. Record the message. 


6. Select an action: 

■ Press 1 to move the message to a single mailbox. 

■ Press 2 to copy the message to one or more mailboxes. Use a # sign to separate multiple mailboxes. 

■ Press 3 to mark as urgent. 

■ Press 4 to mark as private. 

7. Press ## when finished. 

Personal Greetings 

Users can record up to five individual greetings. Greetings will remain on the system and can be activated 

whenever users need them. See also Chapter 17 for detailed information on greetings in general. 

Recording/Activating Personal Greetings 

1. Dial *98 followed by the first greeting number (*1). 

2. Press the Confirm button on the telephone keypad. 

5* 

5* 

5* 

5* 

5* 

9 

9 

9 

9 

9 

8 

8 

8 

8 

8 

5* 

5* 

5* 

5* 

5* 

1 

2 

3 

4 

5 

Greeting 1 

Greeting 2 

Greeting 3 

Greeting 4 

Greeting 5 

If users need to re-record a greeting, they can activate the greeting and record again (i.e., dial *98*6 

followed by the confirm button to re-record message 6). 

Greeting 0 is the initial greeting that was activated at the time the user set up the voicemail box. 



3. To activate a greeting, users must dial their extension number, press 9, and select the greeting they 

want to use (* will bring the user to the main menu of the mailbox.) 

Hearing Your Personal Greetings 

1. Call into your voice mailbox. 

2. Press * to skip to the main menu. 

3. Press 9 from the main menu to hear your recorded greetings. 

Advanced Features 

Send a Voicemail Message Without Ringing Extension(s) 

The voicemail system allows users to send voicemail messages to other users without ringing their extension. 

This is done by calling the voicemail system rather than the user’s phone. 

Scenario: Suppose two employees, Mary and Jack, are working on a project, and Mary has some information 

that Jack needs. Rather than phone him, she calls the voicemail system and records a message for Jack. She 

then tells the voicemail system to send it to Jack’s extension. The message is immediately put into Jack’s voice 

mailbox without his phone ever ringing. Jack’s message waiting light immediately alerts him of a new message. 

The message can be sent to any number of additional employees. 

To send a voicemail to multiple extensions without ringing extensions: 

1. Enter your extension into the telephone keypad. 

This should bring you to the main menu of the auto attendant. If you hear your message instead, 

press the * key on the telephone keypad. 

2. Press 5. 

3. Record your message, then press the # sign. 

4. Press 2 to copy the message to an extension(s). 

5. Enter the extension(s). Multiple extensions must be separated by the # key. Press ## when finished. 

1 1 1 # 2 2 2 # # 

Transfer a Call Directly to Someone’s Voicemail 

The system administrator must enable a direct dial number before users can “blindly” transfer a call to someone 

else’s voicemail. The default for direct dial is “8.” 

1. Put the call on hold using the Hold button on your telephone keypad. 

2. Press the Transfer key. (If your phone does not have a Transfer button, dial *77 to transfer the call.) 

3. Enter the mailbox prefix key. 



Hold 

Transfer 

*77 

8 

1 2 53 


300 


The caller will be transferred to the extension’s voicemail. Because the caller will bypass the voicemail timeout, 

the caller will not hear any ringing. 

This method does not need to involve an outside caller. If you want to call a person’s mailbox directly, you 

can dial 8123 followed by the confirm button. 

8 

1 2 53 

Listen to Voicemail on Cell Phone 

Users can configure their account so that the system rings their cell phone whenever someone has left a 

voicemail. Even when DND has been activated, users will still be notified of new messages. For more information 

on this feature, see page 268. 

Voicemail Notification 

Via Email 

Users can receive voicemails in their email account (as WAV files) by enabling the Send message as attachment 

setting (see page 269). Other settings are also available at this page (go to Settings > Email). 

Via Cell Phone 

If users have a cell phone (or another phone) configured to their extension, the system can alert them whenever 

a new message has arrived in their voicemail (see page 268). Even if the user’s phone is on DND (“do 

not disturb), the system will call the second phone and let the user know that a new message is waiting. 

Voicemail Feature Codes 

This section covers the handful of star codes that are associated with the voicemail system (*95, *96, *97, 

*98, and *99). 


Send Voicemail to Email Account (*95/*96) 


Users that have an email account on the system can have their voicemail forwarded to their email account. 

To activate this feature: 


* 

* 

9 5 

9 6 

Activate 

Deactivate 

Users can ask someone to leave a voicemail message to ensure that the forwarding works (the email is 

shown below). If they do not receive an email, they should check their spam folder. 

Important: When using the *95 star code, users will not receive voicemails on their phone. All voicemails 

will be directed to their email account only. Users who prefer to receive voicemail messages on 

both their phone and email account should use the Send a mailbox message by email setting from 

the web interface (page 269). 

Go to Voicemail (*97) 

Users can listen to their voicemail from their phone by entering *97 and pressing the confirm button on the 

telephone keypad. 

* 

9 7 

Users can also listen to their voicemail from an outside phone: 


2. From the auto attendant, enter your extension number or if you know the direct dial prefix (8 is 

default), dial the prefix plus your extension (i.e., if your extension is 511, then enter 8511). 

3. Press the star (*) key; otherwise, you will be prompted to leave a message. 

4. Enter your voicemail PIN. 

Main # 

8 5 1 1 * PIN 

Go to Group Mail (*97) 

Group mailboxes can be used for anything from a joint boss/secretary mailbox to a place to collect calls that 

come in after hours, at lunch, or during a holiday. Users that share a mailbox with other users can access its 

voicemail messages in a way similar to accessing mail from a single user account. 

Note: The messages that are left in the group mailbox can be picked up from either the user’s extension or 

the group mailbox extension. 

To create a group mailbox, the administrator needs to create an extension and enter the extensions of those who 

will be allowed to access the mailbox (see the Allow Access for Extensions setting shown below). 


302 


Users can listen to groupmail messages from their main phone by dialing *97 and pressing the confirm button 

on the telephone keypad. 

* 

9 7 

Users can also listen to groupmail messages from an outside phone: 


2. From the auto attendant, enter the extension number of the shared mailbox or if you know the direct 

dial prefix (8 is default), dial the prefix plus the extension (i.e., if the extension is 511, then enter 

8511). 

3. Press the star key; otherwise, you will be prompted to leave a message. 

4. Enter the voicemail PIN of the shared mailbox. 

Example: Main # 8 5 1 1 * PIN 

Clear Message Waiting Indicator (*99) 

The message waiting indicator (MWI) notifies the user through the message light on the phone when a new 

message has arrived. Depending on the type of phone the user has, the light will be either yellow or red. For 

all phones, the light will blink when a new message has arrived. Once the user has listened to the message, a 

solid light will replace the blinking light. This enables the system to alert the user the next time a new message 

has arrived. 

Users can clear the message indicator without listening to their messages by dialing *99. (This will not delete 

their messages; it will only clear the indicator.) 

Note: Users must activate the Send message waiting indication setting (page 268) to be notified of new 

messages. 


CELL PHONES 

Chapter 

25 

One of the most valuable benefits of cell phone integration is the ability to receive calls regardless of where 

you are without having to worry about unwanted ringing during off hours. Users can define which hours 

of the day or night they want to receive calls. Whether the phones ring simultaneously or in sequence, or 

whether the cell phone rings at all, is also controlled by the user. 

The cell phone is treated like a registered IP phone. When calls come in, the system dials both the cell and 

desk phones. Cell phone integration gives the user numerous advantages, including lower cell phone costs 

and access to snom ONE star codes. Because the system recognizes the user when calling into the system 

from the cell phone, the user can easily access the Personal Virtual Assistant (page 310). Advantages of cell 

phone integration are highlighted in Table 25-1. 

Table 25-1. Cell Phone Features 


Simultaneous Cell phone integration enables the user to specify another phone number that will ring 

ring at the same time as the extension. This could be a cell phone or any other number. 

Calling Users can set their own calling schedule. This schedule determines when the cell phone 

schedule will respond to office extension calls. 

Ring delay Ring delay determines how long the office extension will ring before the cell phone 

redirection takes place. This is helpful if the user is in and out of the office throughout 

the day. 

Safety net for 

office phone 

If the extension is disconnected from the system at any time, calls will automatically be 

sent to the user’s cell phone. When reconnected, the extension will resume ringing. 

Note: Cell phone integration is different from having two phones registered to the same extension. Registering 

two phones to the same extension allows users to merely receive calls at one of two locations, while cell 

phone integration allows users to receive calls at any location, as well as be reached at numerous phone numbers. 

Cell phone configuration allows users to be accessible at all times, even in the event of a power failure. 

If the office extension (or remote extension) is disconnected from the system, calls will automatically be sent 

to the user’s cell phone. Once the phone is reconnected, incoming calls will automatically be rerouted to the 

extension. 


304 


Figure 25-1 illustrates the role that the cell phone plays in handling an incoming call. First, the cell phone 

must be configured to be included in all calls. Next, the call rings both phones. In the event the user decides 

not to answer the call, the caller can leave a message. Users have numerous options for listening to the message. 

1. Include cell phone in calls to extension. 

Cell phone number: 

Cell phone number: 

Confirmation: 

Include the cell phone in calls to extension: 

2. Incoming calls ring both your desk phone and your cell phone. 

3. Voicemail messages can be picked up from either the desk phone, 

cell phone, soft phone, or through email: 

Home Inbox 243 emails x traveling Wednesday x RE: New images for cell p x 

Delete Reply Forward Spam Move Actions View: All 

From Subject Date 

voicemail@snom.com Voicemail from (879) 123-4567 to 

Wed 11/25, 8:31 AM 

Vistaprint Our Thanks to You, Vistaprint Resellers! Wed 11/25, 8:31 AM 

Vistaprint Your order has been shipped 

Wed 11/25, 8:31 AM 

Figure 25-1. Cell Phone Integration in Action 

Cost Savings 

Once the user’s cell phone has been integrated to the extension, the callback feature can be used to reduce 

the costs of international dialing. (The callback feature is also available through the calling card account.) 

Callback 

The callback feature is ideal from not only a cost savings perspective but also from a billing perspective. Users 

can call into the system to request a callback from any destination they designate. Regardless of where each 

call is made from and to, all calls will be listed in the same account. Single-location billing allows employees 

who work from home to keep their personal lines separate from their business calls. 


Chapter 25: Cell Phones 305 

This feature is designed for users making calls from international destinations back to their home country. 

Neither the user nor the party receiving the call will be charged for the call. The call will be charged to the 

user’s extension. If the user has a cell phone plan that charges only for outbound calls, international calls will 

be free if placed using the callback feature. 

Note: The callback feature can be used from a phone that has not been integrated to the user’s account (e.g., 

a phone from the user’s hotel room); however, it is much simpler from a phone that is registered. 

Request a Call Back 

Once the user’s cell phone has been integrated as part of the user’s extension, the callback feature is easily 

accessible through the Personal Virtual Assistant and requires neither an extension number nor a PIN when 

the callback is initiated. 

1. Dial into the system. The Personal Virtual Assistant announces the options. 

2. Press 2 to initiate the callback. 

The phone will be silent for approximately 30 seconds before the system calls you back. 

3. Enter the destination number followed by the # sign. The system will connect the call, billing both 

calls to the user’s extension number. 

Calling Card Account 

Before you can use the calling card, you need to know the extension number of the account (see your administrator). 

A calling card account can can be used in numerous ways: 

■ As a standard calling card to reach out to a third-party web server for call authentication and pin 

use. 

■ To dial into the system and have the system dial out. 

■ To dial into the system and request a callback. 

The calling card is ideal when traveling because it allows users to place outbound calls from the system without 

logging into their extension. And regardless of where they are calling from, users can present the caller-ID 

of their office (they can also block their caller-ID). 

The calling card allows users to make international calls at much lower rates than if dialing the calls directly. 

For example, dialing the Caribbean from the United States costs in excess of $1.00 per minute if dialed directly. 

With the calling card and SIP trunking, it could cost as little as $0.20 per minute. 

To initiate a callback using the calling card account: 

A calling card account enables users to call into the system from a phone that has not been configured to 

their account and initiate a callback. 

1. Dial into the calling card system using the main number. 

2. The auto attendant announces the options. 

3. Enter the extension number of the calling card account. 

4. Enter your extension number. 

5. Enter your PIN. 

6. Enter the destination number followed by the # sign (a phone number or an extension number is 

allowed). 

The system will connect the call, billing the call to the user’s extension. 


306 


Once the user’s cell phone has been configured to the extension, the user can use the calling card to make 

these calls from the cell phone. Although calls made using the calling card still consume the user’s cell phone 

minutes, the rates are less expensive than calling directly. Users can view calls placed from the calling card 

by going to Lists > Call Log. 

Configuring the Cell Phone 

Add a Cell Phone to Your Account 

This section shows you how to configure a cell phone to the account. Navigate to Settings > Redirection. 

Details for each setting are shown after Figure 25-2 

Figure 25-2. Cell Phone Settings 

■ Cell phone number: While most users populate this field with a cell phone number, any phone number 

can be used. The number should be entered without any spaces or hyphens (e.g., 9781234567). 

■ Confirmation: This setting allows users to require confirmation (i.e., pressing 1 on the cell phone) 

before the call will be connected. The benefit of requiring confirmation is that if the cell phone is 

temporarily unavailable (i.e., out of range, without power, etc.), the call will not automatically go to the 

user’s voicemail and the caller is likely to call back. 

■ Include the cell phone in calls to extension: This feature gives users the freedom to pick up calls 

from either their extension or cell phone. When this feature is activated, both phones will ring. 

Call forward rings only the phone where the call is being redirected to. The cell phone can be set to 

ring from 1 to 30 seconds after the extension starts ringing. 



■ When calling the extension in a hunt group: This setting allows users to tell the system to include 

their cell phone in calls made to a hunt group. 

■ When calling the extension in an agent group: This setting allows users to tell the system to include 

their cell phone in calls made to an agent group. 

■ Include cell phone for web callback: This setting is linked to the click-to-dial feature which is 

available from the address book, missed calls, and call log pages. Click-to-dial works as follows: A 

user clicks a contact, and the system calls the user’s extension before it dials the contact. The Include 

cell phone for web callback setting allows the user to include a cell phone in the mix. When 

this setting is set to on, the system will call both the extension and the user’s cell phone. After one 

of the phones is answered, the PBX dials the contact. This setting is useful if the user is traveling 

and does not have access to the office phone. 

■ Specify time when system calls the cell phone: The options within this setting allow users to 

specify the times at which the system will ring their cell phone. This setting can prevent calls in the 

middle of the night. 

Note: The user’s time zone setting must be set to reflect the user’s time zone; otherwise, calls may be 

received outside the parameters set for the cell phone. 

Users have the following options when configuring their cell phone: 

— No specific time excluded: The cell phone will ring every time the user’s extension rings, 

regardless of the time. 

— Explicitly specify available times: This setting allows a user to specify a schedule for the 

system to use when calling the cell phone (Figure 25-3). Users must populate the fields for 

each day of the week. 

■ Users can use either the 24-hour format (e.g., 21:00 for 9:00 p.m.) or the English AM/ 

PM style (e.g., 9:00P for 9:00 p.m.). However, both formats require a start time and 

an end time (e.g., HH:MM-HH:MM). 

■ To specify more than one time segment, the following format must be used: 9:00-12:00 

1:00P-7:00P (one space separates the two segments). 

■ To specify the days on which the schedule should not be activated, such as a holiday or 

any other day, the holiday field must be used: 

— To specify 1 day, use the MM/DD format (e.g., 11/24 for November 24). 

— To specify 2 days, use the MM/DD format with a space between the days (e.g., 

12/24 12/25 for December 24 and 25). 


308 


Figure 25-3. Cell Phone Calling Schedule 

■ When this cell phone calls the PBX: The Personal Virtual Assistant provides special prompts to 

users who call into the system from their cell phone. This setting allows you to turn this feature off. 

Safeguard Against Loss-of-Signal Events 

In the event of a power outage (or if the user’s phone has been turned off), incoming calls will typically be 

terminated on the voicemail of the cell phone. To prevent missing calls in this way, users can activate the 

Confirmation feature so that the system will wait until the user accept the call (by pressing 1) before actually 

connecting the call. The system will not send the call to voicemail. This way, the caller is likely to call back. 

Users can configure this setting by selecting Press “1” to connect the call as shown below: 

Ring My Cell Phone When Voicemail Arrives 

Users who have decided against forwarding their calls to the cell phone can still be alerted when a new 

message arrives on their extension. Once a caller has left a message, the system will call the user on the cell 

phone. The user can press 1 to pick up the message. 

To activate this feature: 

1. From the Settings tab, click Mailbox. Then click Yes to activate the Call cell phone feature as 

shown below: 



2. Tell the system not to include your cell phone in any incoming calls. To do this, modify your cell 

phone redirection settings, as shown below: 

Note: Putting your phone into DND mode (using *78) will prevent incoming calls from ringing 

your cell phone; however, you will be notified of new voicemail messages as long as you activate the 

Call cell phone setting. 

Feature Codes 

The feature codes used in this section are related to cell phone use (*51, *52, or *00) but they cannot be 

executed from the cell phone. 

Retrieve Call From Cell Phone (*51) 

This feature allows users to pick up an in-process cell phone call on their extension phone. This feature can 

be used only for calls that have been made from the cell phone (through the Personal Virtual Assistant) or 

which have been redirected to the user’s cell phone. Calls that have been made directly to the user’s cell 

phone (without involving the system) cannot be picked up on the user’s extension phone. 

Scenario: You’re on your way to the office and you need to call a client, so you call into the system and place 

an outbound call. You speak with the client on your way to work, and once you arrive at the office, you 

decide to pick the call up from your office phone. 

To transfer a cell phone call to your extension: 

1. From your cell phone, call into the main number of the PBX (your cell phone must be associated 

with your account). 

2. From the Personal Virtual Assistant menu, press 1 for “outbound call.” 

3. Enter a destination number, then press the # key. 

4. When you’re ready to pick the call up from your office phone, dial *51 from the office phone. 

Once the call has been connected, your cell phone will disconnect and will no longer accumulate 

minutes. 


310 


Move Current Call to Cell Phone (*52) 

This feature allows users to move a current call from the extension phone to the cell phone. (Note: The user’s 

cell phone must be configured to the extension before this feature can be used.) 

Scenario: You’re on your extension in the middle of a conversation and need to leave the office. Instead of 

ending the conversation, you put the caller on hold and transfer the call to your cell phone. 

To transfer an extension call to your cell phone: 

1. Place the extension call on hold. 

2. Dial *52 from your office phone. 

3. Answer the call on your cell phone. 

Call Cell Phone of the Extension (*00) 

This feature allows users to call someone else’s cell phone without knowing that person’s cell phone number. 

(Note: The individual’s cell phone number must be configured as part of the extension that is being called.) 

Scenario: You notice that your boss has just left the building when an important call comes in. You have no 

idea what his cell phone number is, but you must speak with him. You know what his extension number is, 

so you use the *00 star code followed by his extension number. Because the system sees the extension and the 

cell phone number as being the same, the system is able to ring his cell phone. 

To call someone’s cell phone without knowing the number: 

1. Dial *00 followed by the person’s extension number. 

2. Press the start button from the telephone keypad. 

* 0 0 5 5 5 

The system will connect you to the individual’s cell phone. 

Personal Virtual Assistant 

Once a user’s cell phone has been configured as part of the extension, the system will recognize the user 

when the user call into the system from that phone. Instead of hearing the auto attendant, the user will hear 

the Personal Virtual Assistant, which will present the user with a new set of prompts. These prompts allow 

the user to make outbound calls, receive callbacks, go to voicemail, or go to the auto attendant. 

Note: Administrators can disable the Personal Virtual Assistant using the When this cell phone calls the 

PBX setting (page 308) if the auto attendant is preferred instead. 

To access the Personal Virtual Assistant: 

1. From your cell phone (or another phone that has been configured to your extension), call into the 

system and press 3 (“Go to your mailbox”). 

2. At anytime during the announcement, you can enter a star code. For example, to set call forwarding, 

use *71 (call redirection) and enter a call forwarding number. In this example, all calls will be 

forwarded to 781-123-4567. 

Voice 

Mail * 7 1 7 8 1 1 2 3 4 5 6 7 # 


1 

2 

3 

4 

Make an Outbound Call 

1. Enter your access code. 

2. Enter the destination number 

followed by the # key. 

Receive a Callback 

When the system rings your phone: 

1. Answer the phone. 

2. Enter the outbound number and 

press the # key. 

Go to Your Voicemail 

You can access feature codes 

from the voicemail system. 

For example, enter the feature feature code 

followed by by the # key (*67#). 

Go to the Auto Attendant 

Press 1 for Sales. 

Press 2 for Support. 

Press 3 for Accounting. 

Press 4 for the dial-by-name directory. 

Press 9 for all other inquiries. 

Figure 25-4. Personal Virtual Assistant 


Use feature codes from your cell phone. 

Making Calls 

*66 Redial 

*69 Call Return 

*90 

Intercom 1 


*77 

Transfer 2 

Call Park3 *85 

Call Park Retrieve3,5 *86 

Call Pickup3,5 *87 


*67 Block CID (Activate) 

*68 Block CID (Deactivate) 

*88 

*89 


(Activate) 


(Deactivate) 

snom ONE Star Codes 

Call Forwarding 

*70 Hot Desking 

Call Forward All 

(Activate) 2 

Call Forward All 

(Deactivate) 

Call Forward Busy 

(Activate) 2 

Call Forward No Answer 

(Activate) 

Call Forward No Answer 

(Deactivate) 

2 

*71 

*72 

*73 

*74 Call Forward Busy 

(Deactivate) 

*75 

*76 

*78 DND (Activate) 

*79 DND (Deactivate) 

*80 

Set Night Mode for 

Domain Accounts 3 

Voicemail 

*95 Send Voicemails as Emails 

(Activate) 

*96 Send Voicemails 

(Deactivate) 

*97 Go to Voice Mail 

*98 Record New Greetings 

*99 

Clear the Message 

Indicator 

Call Center 

*64 

*65 

3, 5 

Agent Log In 

3, 5 

Agent Log Out 

Call Barge In1 *81 

Teach Mode1 *82 

Listen In1 *83 

Cell Phones 

*51 Retrieve Call from Cell Phone 

*52 Move Current Call to Cell Phone 

Call Cell Phone of the Extension1 *00 

Miscellaneous 

*53 Conference 

*61 Show Account Balance 

*62 Wakeup Call 

*63 Request Call Details 

*84 Clean up an Extension 

*91 Add to White List 

*92 Add to Black List 

Record ON4 *93 

Record OFF4 *94 


312 



Appendices 



WORKING WITH CSV FILES 

appendix a 

CSV (comma-separated values) files can be used to expedite the creation of all types of accounts and the domain 

address book. A CSV file is a plain text file that stores tabular data from database-style tools. Records 

appear on separate lines, and the fields within each record are traditionally separated by a comma (snom 

ONE requires a semicolon). CSV files can be created from many document types, including Excel and 

Word. 

The first step in any conversion process is knowing the file requirements and determining which parameters 

are needed for a particular account. This information is detailed below. 

Structural Requirements 

Structure 

CSV files need to follow to the structural requirements that have been set by snom. For example, if you are 

preparing a file for an address book import, it cannot include fields that are not part of snom ONE’s address 

book structure. Extra columns (e.g., “Nick Name,” as shown below) will need to be removed. 

First Name;Last Name;Nick Name;Phone;Speed Dial 

First Name;Last Name;Phone;Speed Dial 

Other requirements are listed below: 

■ The first line of your CSV file requires a “header.” The header specifies which fields will be used and 

can include only the acceptable parameter names (lowercase letters are required). 

First Name;Last Name;Phone;Speed Dial Incorrect syntax 

first;name;number;speed Correct syntax 

■ Each record must be on one line (lengthy records can wrap onto the next line). When lines do 

wrap, be sure to surround the field that contains the embedded line-break with double-quotes. 

■ Fields must be separated with semicolons. During CSV file conversions, commas are automatically 

used to separate the fields. Before importing the file to snom ONE, you will need to replace them 

with semicolons. 

Smith,Mary,9783731234 Incorrect syntax 

Smith;Mary;9783731234 Correct syntax 

■ Fields with embedded semicolons must be delimited with double-quotes. 

■ Fields that contain double quote characters must be surrounded by double-quotes, and the embedded 

double-quotes must each be surrounded by a pair of consecutive double quotes. 

Company “A,B,C” Incorrect syntax 

“Company ““ABC””” Correct syntax 

For most files being converted to CSV, the two criteria that usually need to be addressed are the file structure 

and the field separator (the comma changes to a semicolon). 


316 


The following four lines meet the criteria stated above. Records are on separate lines, semicolons act as field 

separators, and embedded quotes have been addressed. 

Cynthia Smith;102 Tremont Street;Bradford;MA;01834;9781234567 

Bob Pine;22 King Street;Boston;MA;02134;6171234567 

Cathy Chere;13 Tremont Street;Bradford;MA;01834;9781234567 

Pat Rushetta;18 Elm Street;Bradford;MA;01834;9781234567 

“Company ““ABC”””;38 Main Street;Boston;MA;02134;6171234567 

Parameters 

The parameters required to create a domain address book will differ from those required to generate a large 

number of extensions. The tables in the following sections list the required parameters for each CSV file 

type. (Address books are discussed on page 324.) 

Extensions 

Not all parameters listed in Table A-1 are required. Parameters can be added or removed based on need, and 

not every extension will necessarily have a value for every parameter you’ve included. In those cases, be sure 

to retain the semicolon (;) for that field, and make sure the number of values in each line corresponds to the 

number of parameters at the top line. The line below shows a missing ANI: 

type;alias;ani;password;web_pass;first_name;display_name;mb_pin;email_ 

address;mac 

extensions;651;;1234;1234;Carl;Clever;1234;cc@abc.com;000BBBCCCDDDE 

Table A-1 Extension Account Parameters 

Parameter Definition Value 

type Account type extensions 

alias Extension number and alias 508 Johnny* 

ani ANI 

password SIP password 1234 

web_pass Web interface password 1234 

first_name User’s first name John 

display_name User’s last name Smith 

mb_pin User’s PIN 1234 

email_address User’s mail address 

mac User’s MAC IP address 192.168.1.4 

picom Extensions permitted to barge in 509 509 * 

profile Button profile for this phone 12-button 

*When aliases are used in the CSV file, they will show up on the accounts page as a separate entry, giving the 

impression that each extension has been created twice. While this will not be a problem, you can avoid this 

by entering aliases manually once the account has been created. 

Sample Syntax—Extension 


address;mac 


Appendix A: Working with CSV Files 317 

extensions;650;9786501234;1234;1234;Fred;Feuerstein;1234;ff@abc. 

com;000BBBCCCDDDD 

extensions;651;9786511234;1234;1234;Carl;Clever;1234;cc@abc. 

com;000BBBCCCDDDE 

Agent Groups 

The possible parameters for the agent group are listed in Table A-2 (not all are required). 

Table A-2 Agent Group Parameters 


type Account type acds 

alias Extension number of the agent group 663 

agents Extension numbers of the agents 41 42 43 

gap_time The number of seconds that the system should 

wait between each announcement 

15 

jump Extensions that are permitted to jump out of 

the agent group 

41 

Sample Syntax—Agent Group 

type;alias;agents;gap_time;jump 

acds;663;40 41 42;15;* 

acds;664;40 41 42;15;40 

Hunt Groups 

The possible parameters for the hunt group are listed in Table A-3 (not all are required). 

Table A-3 Hunt Group Parameters 


type Account type hunts 

alias Extension number of the hunt group account 74 

display Display name for the hunt group hunt_74 

st1_ext Extensions that will be part of stage 1 40 41 

st1_dur Number of seconds the system should ring stage 1 10 

st2_ext Extensions that will be part of stage 2 43 45 

st2_dur Number of seconds the system should ring stage 2 15 

st3_ext 

st3_dur 

(for unwanted stages, leave empty) 

st4_ext Extension or phone number for the final stage 70 

Sample Syntax—Hunt Group 

type;alias;display;st1_ext;st1_dur;st2_ext;st2_dur;st3_ext;st3_dur;st4_ext 

hunts;74;hunt_74;40 41;10;42;15;43 45;10;70 

hunts;75;hunt_75;40 41;10;42 43;15;45 46;10;70 


318 

Conference Account 


The possible parameters for the conference account are listed in Table A-4 (not all are required). 

Table A-4 Conference Account Parameters 


type Account type conferences 

alias Extension number of the account 76 

name Name the conference conf_76 

intro Type of introduction to use when the user enters ask_name 

the conference 

none 

pin Access code for the moderator 777777 

upin Access code for the user 888888 

Sample Syntax—Agent Groups 

type;alias;name;intro;pin;upin 

conferences;76;conf_76;ask_name;777777;888888 

conferences;77;conf_77;none;111112;222222 

Paging 

The possible parameters for paging are listed in Table A-5 (not all are required). 

Table A-5 Paging Account Parameters 


type Account type hoots 


display Display name for the paging account page_662 

ext Corresponds to the Destination setting of the 

paging account (when using unicast mode, enter a 

group of extension numbers; when using multicast 

mode, enter an IP address). 

40 

perm Extensions that are permitted to monitor the account 

41 42 43 

mode Type of paging account unicast 

multicast 

record Indicates whether pages will be recorded true 

false 

Sample Syntax—Paging Account 

type;alias;display;ext;perm;mode;record 

hoots;662;page_662;40;41 42 43;unicast;true 

hoots;661;page_661;*;*;multicast;false 

Service Flags 

The possible parameters for service flags are listed in Table A-6 (not all are required). 


Table A-6 Service Flag Parameters 


type Account type hoots 



display Display name for the account srv_658 

mode Determines whether this service clock 

flag will be automatic (clock) or 

manual (when clock is chosen, the 

system will automatically use the 

schedule that is specified below). 

manual 

sound night 

mon When the clock mode is used, 9:00-12:00 13:00-17:00 

tue the auto attendant will route calls 9:00-17:00 

wed 

thur 

according to the schedule that is 

indicated here. 

9:00-12:00 1:00P-4:00P 

9:00-17:00 

fri 

sat 

sun 

9:00-17:00 

holiday Indicates a holiday schedule (i.e., the 12/24 1/1 

days on which the schedule should 

not be activated) 

Sample Synax—Service Flags 

type;alias;display;mode;sound;mon;tue;wed;thur;fri;sat;sun;holiday 

srvflags;658;srv_658;clock;night;9:00-12:00 13:00-17:00;9:00-17:00;9:00- 

12:00 1:00P-4:00P;9:00-17:00;9:00-17:00;;;12/24 1/1 

srvflags;659;srv_659;manual;;9:00-12:00;9:00-17:00;9:00-12:00 3:00P- 

5:00P;9:00-17:00;9:00-17:00;;;1/1 3/19 

Converting an Excel File to CSV 

Once you have become familiar with the parameters that can be used for a particular account, you can prepare 

the file for CSV conversion. 

1. Open the file that contains your source data. In the example shown below, the source file is an Excel 

file. 

2. Review the contents of your file to determine which material does not conform to the specified 

parameters for a particular account type. As a reminder, the allowable parameters for the extension 

account are listed below: 


320 



address;mac 

Note: Your file cannot contain columns that do not conform to these parameters. In the exercise 

sample shown in Step 1 above, columns C, D, E, and F contain nonconforming material and will 

not be needed for the CSV conversion. 

In addition to removing data, you may also need to add data. For example, it is recommended that 

you add columns for data such as ANIs, passwords, PIN codes, and MAC addresses so that you will 

not have to manually enter this data later on. To simplify this process, you can assign identical passwords 

and PIN codes to each extension and have the users create unique numbers once their phones 

have been registered. 

3. Delete the nonconforming columns (Edit > Delete). The material that remains must correlate with 

the allowable parameters, as shown below: 

4. When creating a block of extensions using a CSV file, the type and alias fields are mandatory. 

To accommodate these parameters, add two additional columns to your file. (To add a column using 

Excel, place your cursor in the first column, then click Insert > Columns.) 

5. To accommodate other new material you would like to include, such as passwords, PIN codes, and 

other information, add columns for this information as well. This will simplify the process later on. 

6. Rename columns so that they conform to the extension parameters (Table A-1): 



7. Populate the empty fields. Begin by populating the fields that will have identical information, such as 

type, password, web_pass, and mb_pin. Type only the first entry as shown below: 

These parameters are outlined as follows: 

■ type: When adding extensions, the value will always be extensions. 

■ password: This field can be populated with any type of password; however, the quickest 

method is to use the same password for all users and have the users change their own password. 

■ web_pass: See the previous comment. 

■ mb_pin: This field requires a PIN. The quickest method is to use the same password for all 

users and have the users change their own password. 

8. To quickly populate the columns using Excel’s Auto Fill feature, use the following instructions: 

Click on the cell containing the text you would like to repeat. (Extension numbers will be done 

later.) Place your cursor at the bottom-right corner until you see a plus sign. Then, holding down the 

Alt key, drag the plus sign down to the end of the column. 

9. Repeat Step 8 until the necessary columns have been populated. 

Next, enter extensions for each account. Adding a block of extensions using Excel can be expedited 

using the Auto Fill tool, as explained below: 

■ Choose the extensions you that you will be using (make sure none of them are currently in use). 

For the example, in this exercise, extensions 509 through 516 will be used. 


322 


■ Enter the first extension, as shown below: 

■ Click the bottom right-hand corner, then while holding down the Ctrl key, drag the double 

plus sign to the end of the column. The numbers will appear in sequential order, as shown 

below: 

10. Populate the remaining columns. These will need to be done manually: 

■ ani: 

■ email_address: Supply the user with an email address. 

■ mac: Enter the MAC address of the phone that will be assigned to this extension. 

11. Once you are satisfied with the contents of the file, save the file as a CSV file type (File > Save As). 

12. Click OK when the following message appears: 

13. Click Yes at the following message: 



14. You now have a CSV file (the file name is at the top left-hand corner). Proceed to the next section to 

ensure that the integrity of the file has not been compromised in the conversion process. 

Changing the Field Separator and Validating the File 

Now that the CSV has been created, you will need to go into the file to change the field separator from the 

default comma to a semicolon. You will also need to ensure that the integrity of the data has not been compromised. 

This includes checking for extra commas, extra quotation marks, and any other formatting which 

Excel may have inserted. This will be done from a plain text editor. 

1. Open the new file from a plain-text program such as Notepad or TextEdit. 

2. Check to make sure there are no extra commas other than between fields. Check also for extra quotation 

marks or other formatting which Excel may have inserted. In the CSV file shown above, Excel 

has inserted commas at the end of a few of the rows. Those will need to be deleted. 

3. Change the comma to a semicolon by clicking Edit > Replace. 

The final file is shown below. The file is now ready to be used for the address book upload. 

4. To import the CSV file into snom ONE, see “Importing the CSV File” on 

page 327. 


324 


Creating a Domain Address Book 

Address Book Parameters 

When creating CSV files for a domain address book, be sure your data conforms to the structure of snom 

ONE’s address book. Depending upon the particular file you will be converting to CSV, this may mean that 

you will need to remove information that does not conform to this layout. The acceptable parameters are 

listed in Table A-7. 

Table A-7 Address Book Parameters 

Parameter 

display number 

Definition Value 

domain Domain name (not required for single domains) 

first Last name Smith 

name First name Fred 

number Phone number: The following characters are ignored by the 

system (although it is permissible to use these characters, they 

will not appear in the address book): 

- (hyphen) 

, (comma) 

(space) 

/ (backslash) 

. (period) 

The plus (+) sign is permitted when entering global numbers, 

e.g., 

+49 (30) 386-12345. 

(978) 543 6545 

speed Speed dial numbers cannot overlap with numbers that are in 

use as star codes (see Chapter 23 for the list of star codes). 

*12 

type 

user 

Black list/white list indicator: By default, all address book 

entries are classified as white list contacts. To specify that an 

entry should be placed on the black list, enter “black” into 

this field. 

black 

Converting the Address Book to CSV 

Once you have become familiar with the parameters that can be used for domain address books, you can 

prepare the file for CSV file conversion. 

1. Open the source file. In the example shown below, the source file is an Excel file. 

2. Assess the contents of the file to determine which material does not conform to the parameters listed 

in Table A-7. As a reminder, the allowable parameters are listed below: 



display_number;domain;first;name;number;speed;type 

Keep in mind that your file cannot contain columns that stray from these parameters. In the sample 

shown in Step 1, columns C, D, E, and F contain nonconforming material. 

3. Delete the nonconforming columns (Edit > Delete). The material that remains must correlate with 

the allowable parameters, as shown below: 

4. Change the first row to reflect the actual names of the parameters: 

5. If you would like to add another parameter, you could do so now. For this particular file, you could 

add the type parameter, so that white list contacts can be distinguished form black list contacts; 

however, there are so few black list contacts that this distinction could be made manually just as easily. 

6. Once you are satisfied with the contents of the file, save the file as a CSV file type (File > Save As). 

7. Click OK when the following message appears: 


326 


8. Click Yes at the following message: 

9. You now have a CSV file, as is indicated by the file name in the top left-hand corner. Proceed to 

the next section to ensure that the integrity of the file has not been compromised in the conversion 

process. 

Changing the Field Separator and Validating the File 

Now that the CSV has been created, you will need to go into the file to change the field separator from the 

default comma to a semicolon. You will also need to ensure that the integrity of the data has not been compromised. 

This includes checking for extra commas, extra quotation marks, and any other formatting which 

Excel may have inserted. This will be done from a plain text editor. 

1. Open the new file from a plain-text reading program such as Notepad or TextEdit. 

2. Check to make sure there are no extra commas other than between fields. Check also for extra quotation 

marks or other formatting which Excel may have inserted. In the CSV file shown above, Excel 

has inserted commas at the end of a few of the rows. Those will need to be deleted. 

3. Change the comma to a semicolon by clicking Edit > Replace. The final file is shown below. The file 

is now ready to be used for the address book upload. 

4. To import the CSV file into snom ONE, proceed to the next section, “Importing the CSV File.” 


Importing the CSV File 

Importing a CSV File for the Extension Account 


This section shows you how to import the CSV files for the extension account into snom ONE (the same 

procedure can be used for all account types). 

1. Open the CSV file in a text editor (for example, Notepad). 

2. Select the entire contents and copy it. (It will remain on your clipboard for a short time.) 

3. Click Accounts within the domain, click Create. 

4. From the dropdown list, choose Import CSV. 

5. Paste the contents of the CSV file into the space provided: 



328 


7. To verify that the accounts were created, click List as shown below: 

The extensions are listed with the other extensions of that domain. 

If you don’t see the extensions, go to the troubleshooting section on page 329. 

Importing CSV Files for Domain Address Books 

This section explains how to import a CSV file that will be used as a domain address book. Use the following 

instructions from the snom ONE home page: 

1. From the domain, click the Settings tab, then choose Address-Book: 

2. Scroll to the bottom of the page and click the Browse button. 

3. Locate your CSV file and click Open to import it. 

Note: If you would like to retain any existing address book entries, set Clear Addrbook to off. Otherwise, 

the import will remove any previous entries. 

4. Click Create. The new entries will be displayed in the address book. 


Troubleshooting 


Problem Possible Cause(s) Correction 

Extensions do not appear. Missing header Be sure your CSV file contains a header row 

and that it includes the correct syntax (see page 

315). 

Incorrect syntax. Syntax must conform to the syntax shown in the 

parameter tables beginning on page 316. 

Information appears in the 

wrong place (e.g., the first name 

appears in the web password 

field). 

After the CSV file was imported, 

the Accounts page reflects 

two entries for each extension. 

Too many tabs 

between fields. 

Aliases will show 

up as extensions on 

the Accounts page, 

but internally, 

there will be only 

one extension. This 

is not a problem. 

Remove extra tabs from the CSV file. 

If you want to clean up the file, you can go 

back to defaults and take the space and the alias 

portion out of the CSV file and import it again. 

(Aliases will need to be entered manually using 

this method of CSV import.) 


330 



SIP OVERVIEW 

appendix B 

SIP (Session Initiation Protocol) is a signaling protocol that is used to control multimedia communication 

sessions, such as voice and video calls, over Internet Protocol (IP). SIP is analogous to HTTP for voice and 

is essentially the glue that ties communications systems together, much like HTTP ties clients and servers 

together for worldwide communication. More and more vendors are implementing SIP as a standard 

telephony platform, and it is becoming increasingly popular. This appendix provides an introduction to SIP 

and is designed to introduce you to key concepts and mechanisms of the SIP protocol. 

What is SIP? 

Background 

SIP was originally designed in 1996 to create a mechanism for inviting people to large-scale multipoint conferences 

on the Internet multicast backbone. The first version (SIP 2.0) was defined by RFC 2543, and by 

November of 2000, the protocol had been refined and clarified in RFC 3261. Though IP telephony didn’t 

actually exist at the inception of RFC 3261, SIP evolved to provide a missing piece to the Internet architecture: 

A way for users to explicitly invite others users to join sessions over the Internet. 

What is a SIP Session? 

A SIP session is a related progression of events devoted to a particular activity occurring over the Internet. 

Activities can include two-way telephone calls, video conferencing, streaming multimedia distribution, instant 

messaging, presence, and online games. 

UAC UAS 

1 

INVITE 

5 ACK 

100 Trying 

180 Ringing 

200 OK 

SIP knows nothing about the details of the sessions it controls. It only initiates, terminates, and modifies 

the sessions. Although SIP can work in a framework with other protocols— SOAP, HTTP, XML, VXML, 

WSDL, UDDI, and SDP—it does not perform any of their functions. 

SIP Components 

Entities interacting in a SIP scenario are known as User Agents (UAs), and there are two types of UAs: clients 

and servers. 


2 

3 

4

332 


■ User Agent Client (UAC): The UAC generates “methods” and sends them to servers (e.g., it sends 

an INVITE request and initiates a call). 

1 

UAC UAS 

INVITE 

5 ACK 

Methods Responses 

100 Trying 

180 Ringing 

200 OK 

■ User Agent Server (UAS): The UAS receives the methods, processes them, and generates responses 

(e.g., it sends a 200 Ok response to indicate a successful session). The UAS may issue multiple 

responses to the UAC. 

User Agent Clients 

The UAC is often associated with the end user, since applications running on systems are used by people. 

The UAC can be any end-user device, such as a cell phone, multimedia handset, personal computer (PC), 

personal digital assistant (PDA), or a softphone. 

The requests generated by the UAC are sent to a server (typically a proxy server) and are known as “methods,” 

which will be discussed later. 

Note: Non-IP devices like dumb phones can also be turned into SIP UAs by using an inexpensive analog 

telephone adapter (ATA) to make them SIP-aware. An ATA is a box with one or two analog ports with RJ11 

jacks used for connecting regular analog phones to the VoIP network. Popular ones include the SIPura or 

Linksys PAP2. 

User Agent Servers 

Servers possess a predefined set of rules to handle the requests sent by clients and are usually part of the network. 

There are several types of servers: 

Proxy Server— Proxy servers help track down addresses of recipients whose exact addresses are not known 

in advance. If the proxy server cannot find the address of the recipient, it will send the request to other 

proxy servers. Destinations include another extension on the same proxy server, the next-hop proxy server in 

the routing table, or a media server. SIP proxy servers use presence services to track users, which means users 

can be located regardless of their physical location. Proxy servers are the most common server in the SIP 

environment. 

Registrar Server— A SIP registration server is responsible for registering devices. It does this by authenticating 

the device with a user name and password and keeping a table of IP addresses and extensions/phone 

numbers. This authentication process is similar to logging into a web server, which requires a user name and 

password. The registrations server makes it possible for users to alter the address at which they are contactable. 

Registrations play an important role in the process since SIP devices that do not register cannot be 

called and SIP devices that do not successfully authenticate cannot make outbound calls. A media server is 

a device that handles any kind of media or RT, such as a voicemail server, a conference server, an IVR server, 

and a music on hold server. 


2 

3 

4

Proxy Server 

User Agent User Agent 

Appendix B: SIP Overview 333 

User Agent 

Figure B-1. Different Types of User Agent Servers (UASs) 

Back-to-Back User Agent (B2BUA) 

Registrar Server 

A B2BUA essentially bolts two user agents together in a back-to-back fashion, similar to two people standing 

back to back. A B2BAU establishes a two-legged call, keeping the SIP server in the middle of the call to 

orchestrate the details. One side of the session acts as the SIP UA server that receives the calls; the other side 

acts as the SIP UA client that establishes the other leg of the call. This “middle position” of the SIP server 

allows the system to execute difficult call scenarios, like recording a call, stepping out of the voicemail system 

(by pressing 0), barging into a call, and many other call scenarios that are very hard to do without this center 

position. 

Answering 

SIP UA 

Originating 

SIP Endpoint 

Figure B-2. Back-to-Back User Agent (B2BUA) 

SIP Language 

Similarities with HTML and SMTP 

Originating 

SIP UA 

Answering 

SIP Endpoint 

SIP shares some common characteristics with HTTP and SMTP. Like the latter two, SIP is an ASCII textbased 

protocol which makes it easy to read and troubleshoot. The text below is a SIP trace that shows a user 

inviting another use to a session. 

Users are identified by a SIP address, known as a Uniform Resource Identifier (URI). A SIP URI is similar 

to an email address and is typically built around the user’s phone number or host name (e.g., sip:[your_ 

number]@companyA.vonage.net). This allows users to be redirected to another phone as easily as they 

would be redirected to another web page. 


334 

Types of Messages 


SIP communication consists of two types of SIP messages—methods and responses. Methods are sent from 

the client to the server and are used to indicate the purpose of the request. The following six methods are 

part of RFC 3261. 

Table B-1 RFC 3261 Methods 

SIP Method Description 

INVITE Establishes a session 

ACK Confirms an INVITE request 

BYE Ends a session 

CANCEL Cancels establishing of a session 

REGISTER Registers a user with a proxy/registrar 

OPTIONS Communicates information about the capabilities of the other side 

Table B-2 Non-RFC 3261 Methods 

SIP Method Description RFC 

PRACK Provisional acknowledgement 3262 

SUBSCRIBE Subscribes for an Event of Notification from the notifier 3265 

NOTIFY Notifies the subscriber of a new event 3265 

PUBLISH Publishes an event to the server 3903 

INFO Sends mid-session information that does not modify the 

session state 

2976 

REFER Asks the recipient to issue a SIP request (call transfer) 3515 

MESSAGE Transports instant messages using SIP 3428 

UPDATE Modifies the state of a session without changing the state 

of the dialog 

3311 

Responses are sent from the server to the client and are used to indicate the status of the transaction. Responses 

are delivered in integer form (from 100 to 699) and are categorized as shown in Table B-3. 

Table B-3 Categories of Response Codes 

1xx Informational responses 

2xx Success responses 

3xx Redirection responses 

4xx Request failures 

5xx Server errors 

6xx Global failures 

Common response codes are shown below (for a complete list, see Table B-8). 

Table B-4 Common Responses 

Code Definition Occurrence 

100 Trying When the other side is trying to process the request 

180 Ringing When the other side is ringing 

183 Session progress 



Code Definition Occurrence 

200 Ok When the event is successful 

302 Moved temporarily When forwarding has been activated on the phone number 

that is being called 

305 Use proxy 

400 Bad request When a client enters a bad request 

401 Unauthorized When authentication is required 

403 Forbidden 

404 Not found When a phone number cannot be located 

415 Unsupported 

media type 

When the SDP contains no common codecs (this can happen 

if one side supports only G.729 and the PBX has no G.729 

license defined) 

500 Server internal error When the SIP server is out of resources 

501 Not implemented 

606 Not acceptable When a global failure occurs (usually at the carrier level) 

Content of Messages 

SIP messages consist of the following three parts: 

■ SIP URI: The SIP URI is typically built around the user’s phone number. This first line also indicates 

either the purpose of the request or the response given by the called party. 

■ Message body: SIP requests and responses can both contain message bodies. The content of the 

message body is usually a session description and contains syntax as shown in the message below. 

SIP URI 

Headers 

Message 

Body 

Start Line 

(Response) 

Headers 

Figure B-3. Message Content 

INVITE sip:513@192.168.1.119:1036;transport=tls;line=islsito6 SIP/2.0 

Via: SIP/2.0/TLS 192.168.1.251:5061;branch=z9hG4bK-2ec1bde71fb1935517545f7c96ad68 

From: "Cell Phone BC" ;tag=21838 

To: "Jerry McDonald" 

Call-ID: 3eb9a3b0@pbx 

CSeq: 14649 INVITE 

Max-Forwards: 70 

Contact: 

v=0 

o=- 20105 20105 IN IP4 192.168.1.251 

s=c=IN 

IP4 192.168.1.251 

t=0 0 

m=audio 62116 RTP/AVP 0 8 9 2 3 18 101 

a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:Egjy8tgmY65IlnTnNvyedrLzY6Vo1i6HXY49b1e 

a=rtpmap:0 pcmu/8000 

a=rtpmap:8 pcma/8000 

a=rtpmap:9 g722/8000 

SIP/2.0 200 Ok 

Via: SIP/2.0/UDP 192.168.1.251:5060;branch=z9hG4bK-690e2b838d8204228c527d8bf99470 

From: "Jennifer Hanson" ;tag=45498 

To: "Jennifer Hanson" 

Call-ID: ft836yqi@pbx 

CSeq: 19493 MESSAGE 



336 


■ Headers: SIP header fields provide additional information about the message. Common headers 

are shown below (for a complete list, see Table B-7). 

Table B-5 Common SIP Headers 

SIP Header Description 

To Address of the called party or the destination party 

From Calling party or the caller-ID in some cases 

Via Path taken by the request so far 

Call ID A unique number used to identify the call 

CSeq Used for keeping track of the conversation number 

Contact Used for identifying the user agent and the version of software 

used by the user agent 

Describing the SIP Session 

Besides methods and responses, SIP messages can also carry session descriptions in the form of Session 

Description Protocol (SDP). SDP is used for describing multimedia sessions for the purposes of session 

announcement, session invitation, and other forms of multimedia session initiation. It includes information 

such as IP addresses, port numbers, and times and dates when the session is active. SDP session descriptions 

are text-based and consist of lines of text that are categorized as shown below (refer to Table B-6 for details 

on the syntax): 

Session-level 

information 

Media-level 

information 

v=0 

o=root 839347042 839347042 IN IP4 192.168.1.101 

s=call 

c=IN IP4 192.168.1.101 

t=0 0 

m=audio 52460 RTP/AVP 0 8 9 99 3 18 4 101 

a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:zw00j/jEiq68b1fY0mJ 

a=rtpmap:0 pcmu/8000 

a=rtpmap:8 pcma/8000 

a=rtpmap:9 g722/8000 

a=rtpmap:3 gsm/8000 

a=rtpmap:18 g729/8000 

a=fmtp:18 annexb=no 

a=rtpmap:4 g723/8000 

a=rtpmap:101 telephone-event/8000 

a=fmtp:101 0-16 

a=ptime:20 

a=rtcp-xr:voip-metrics stat-summary=loss,dup,jitt 

a=sendrecv 

In the example shown above, the session-level section consists of five lines of text (always beginning with 

v=0, which indicates the version number of SDP), and the media-level section consists of one audio stream. 

The a=rtpmap lines convey information about the media format used. 

Table B-6 Session Description Protocol (SDP) Syntax 

Type Description 

v Protocol version (optional) 

b Bandwidth information (optional) 


Type Description 

o Owner/creator and session identifier (optional) 

z Time zone adjustments 

s Session name (optional) 

k Encryption key (optional) 

i General information about the session (optional) 

a Session attribute (optional) 

u URI containing a description of the session (optional) 

t Time the session is active (optional) 

e Email address of the contact person (optional) 

r Times when the session will be repeated 

p Phone number to obtain information about the session (optional) 

m Media name and transport address 

c Connection information (not required if included in all media) 

i Information about a media line (optional) 

SIP Call Proces 

Session Establishment 

1 

INVITE 

5 ACK 

100 Trying 

180 Ringing 

200 OK 


UAC UAS 

Call Flow 

1. The UAC sends an INVITE message to 41@pbx.company.com (UAS). 

2. The UAS receives the request and responds with a 100 Trying. 

3. The UAS sends a 180 Ringing response to the UAC when the phone begins ringing. 

4. Once the call is picked up, the UAS send a 200 Ok message to the UAC. 

5. The UAC sends an ACK request to confirm the 200 Ok response was received. 

Note: The ACK method completes what is known as the three-way handshake—confirmation that a session 

has been successfully established. The INVITE is the only method where this occurs, and this is due to the 

large gap of time that often occurs between the INVITE itself and the 200 OK response (when a user can’t 

find the phone, is running to the phone, etc.). So the ACK tells the called party that the caller hasn’t hung 

up and has accepted the call. 

INVITE sip:41@pbx.company.com;user=phone SIP/2.0 


2 

3 

4

338 


Via: SIP/2.0/TLS 192.168.0.32:2061;branch=z9hG4bK-9gg3wzak;rport 

From: “Roland Wright” ;tag=g5ua0i7fz6 

To: 

Call-ID: 3c2812339279-zvojwzvof6we 



Contact: ;reg-id=1 

P-Key-Flags: resolution=”31x13”, keys=”4” 

User-Agent: snom360/7.3.14 

Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, 

MESSAGE, INFO 


SIP/2.0 100 Trying 

Via: SIP/2.0/TLS 192.168.0.32:2061;branch=z9hG4bK-9gohvng3wzak;rport=2061 

From: “Roland Wright” ;tag=g5ua0i7fz6 

To: ;tag=eed7a3b4e0 




SIP/2.0 180 Ringing 

Via: SIP/2.0/TLS 192.168.0.34:5061;branch=z9hG4bK-e299f160c512cb066a3a536253a 

a4d44;rport=5061 

From: “Roland Wright” ;tag=1521860827 

To: “Rachel Reed” ;tag=ozac09qwnh 

Call-ID: f6f17567@pbx 



Require: 100rel 

RSeq: 1 

Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, 

MESSAGE, INFO 

Allow-Events: talk, hold, refer, call-info 


[7] 2010/04/14 16:10:31: SIP Rx tls:192.168.0.31:1037: 

SIP/2.0 200 Ok 

Via: SIP/2.0/TLS 192.168.0.34:5061;branch=z9hG4bK-08d0f56ed1e8d82deb32779c5a2 

cc55b;rport=5061 




CSeq: 28593 PRACK 



ACK sip:41@192.168.0.31:1037;transport=tls;line=zs4m8lei SIP/2.0 

Via: SIP/2.0/TLS 192.168.0.34:5061;branch=z9hG4bK-6dcf1018159b8e96b7b6d62a758 

d77fd;rport 




CSeq: 28592 ACK 


Contact: 



Session Termination 

Call Flow 

1 

BYE 


UAC UAS 

200 OK 

1. Once the caller hangs up, a BYE is sent to the UAS. 

2. The UAS receives the request and responds with a 200 Ok. 

BYE sip:40@192.168.0.32:2061;transport=tls;line=i339wesg SIP/2.0 

Via: SIP/2.0/TLS 192.168.0.34:5061;branch=z9hG4bK-eec7092e42e0694ecd8be3d3e94 

00816;rport 

From: ;tag=eed7a3b4e0 

To: “Roland Wright” ;tag=g5ua0i7fz6 


CSeq: 23626 BYE 


Contact: 

RTP-RxStat: Dur=20,Pkt=970,Oct=170720,Underun=0 

RTP-TxStat: Dur=18,Pkt=975,Oct=171600 


[7] 2010/04/14 16:10:50: SIP Rx tls:192.168.0.32:2061: 

SIP/2.0 200 OK 

Via: SIP/2.0/TLS 192.168.0.34:5061;branch=z9hG4bK-eec7092e42e0694ecd8be3d3e94 

00816;rport=5061 

From: ;tag=eed7a3b4e0 

To: “Roland Wright” ;tag=g5ua0i7fz6 


CSeq: 23626 BYE 


User-Agent: snom360/7.3.14 

RTP-RxStat: Total_Rx_Pkts=976,Rx_Pkts=972,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0 

RTP-TxStat: Total_Tx_Pkts=974,Tx_Pkts=974,Remote_Tx_Pkts=751 


REGISTER Request 

The Contact field is used by the SIP UA when using the REGISTER method. Once it receives the IP and 

configuration from the registrar, the SIP UA can begin registering (scenario 1 shown below). If the UA is 

challenged for authenication, it will resend the registration request with the necessary credentials (realm, 

username, password), if it has been set in the server. A 200 OK acknowledges a successful registration (scenario 

2 on page 341). 


2

340 

Scenario 1 

1 

Call Flow 

REGISTER 


200 OK 

Registrar 

1. The UAC tries to register the user’s SIP address. 

2. The address is located, so a 200 OK response is sent to the UAC. 

Scenario 1 

Call Flow 

REGISTER sip:cs410.snom.com SIP/2.0 

Via: SIP/2.0/TLS 192.168.1.103:2077;branch=z9hG4bK-rib4emgq;rport 

From: “Hanna Flanders” ;tag=fwxg2s0u4i 

To: “Hanna Flanders” 

Call-ID: 3c267028e8ad-q7dwiql86rvc 

CSeq: 6134 REGISTER 


Contact: ;regid=1;q=1.0;+sip.instance=”” 

[7] 2010/04/14 16:05:16: SIP Tx tls:192.168.1.103:2077: 

SIP/2.0 200 Ok 

Scenario 2 

1 

4 

REGISTER 

REGISTER 

407 Proxy Authenication Required 

200 OK 

1. The UAC tries to register the user’s SIP address. The Web server (running the Web site) thinks that 

the HTTP data stream sent from the UAC was correct, but access to the URL resource requires the 

prior use of a proxy server that needs some authentication which has not been provided, so it responds 

with a 407 Proxy Authentication Required response. 

2. The user logs in (enters user ID and password) with the proxy server. The UAC resends the REGIS- 

TER method. 

3. The UAS responds with a 200 OK. 


3 

3 

6

Scenario 2 


REGISTER sip:itsp.com SIP/2.0 

Via: SIP/2.0/UDP 75.150.83.2:5060;branch=z9hG4bK-79d064228874f311c131a824e14e 

910f;rport 

From: “itsp” ;tag=45682 

To: “itsp” 

Call-ID: niyrncl8@pbx 



Contact: ;+sip. 

instance=”” 

SIP/2.0 407 Proxy Authentication Required 

v: SIP/2.0/UDP 75.150.83.2:5060;branch=z9hG4bK-79d064228874f311c131a824e14e91 

0f;rport=5060 

f: “itsp” ;tag=45682 

t: “itsp” 

i: niyrncl8@pbx 


Proxy-Authenticate: Digest realm=”itsp.com”, domain=”sip:itsp.com”, nonce=”ab 

45353d389d01ba2086232c09417752”, opaque=””, stale=TRUE, algorithm=MD5 

l: 0 

[7] 2010/04/14 16:05:17: SIP Tx udp:204.11.192.31:5080: 

REGISTER sip:itsp.com SIP/2.0 

Via: SIP/2.0/UDP 75.150.83.2:5060;branch=z9hG4bK-a071e61e35d725f2c9a17c9bf2ed 

b7f7;rport 

From: “itsp” ;tag=45682 

To: “itsp” 

Call-ID: niyrncl8@pbx 



Contact: ;+sip. 

instance=”” 

User-Agent: snom-PBX/4.0.1.3453 

Supported: outbound 

Proxy-Authorization: Digest realm=”itsp.com”,nonce=”ab45353d389d01ba2086232c0 

9417752”,response=”664617e317a5c34448f276db3932be2f”,username=”17772830314”,u 

ri=”sip:itsp.com”,algorithm=MD5 

Expires: 3600 


SIP/2.0 200 Ok 

CANCEL Request 

The CANCEL request cancels pending transactions and generates an error response to the pending request. 

CANCEL has no effect on requests that have already received a final response (e.g., an ACK). Therefore, 

the CANCEL method is often associated with methods that require lengthy response times, such as the 

INVITE. A UAS that receives a CANCEL request for an INVITE but hasn’t yet sent a final response would 

cease ringing and then respond to the INVITE with a specific error response (a 487, as shown below). 


342 

1 

4 

ACK 


UAC UAS 

CANCEL 

200 OK 

487 Request Terminated 

Figure B-4. CANCEL Request—INVITE Response 487 Request Terminated 

Call Flow 

1. A user makes a call, but after a few rings decides to hang up, so a CANCEL request for the previous 

INVITE is sent by the client to the server. 

2. The called party’s phone stops ringing, and a 200 OK response for the CANCEL is sent by the server 

to the client. The server then sends back a 487 Transaction Terminated response. 

3. The client acknowledges the response by sending an ACK. 

CANCEL sip:50@localhost;user=phone SIP/2.0 

Via: SIP/2.0/UDP 192.168.0.32:2048;branch=z9hG4bK-lg7xoa360cuh;rport 

From: “PC PBX 51” ;tag=u58kfgll03 

To: 

Call-ID: 3c2c575f0429-38s24k0wrdrx 

CSeq: 2 CANCEL 


Reason: SIP;cause=487;text=”Request terminated by user” 


[7] 2010/04/28 11:21:27: SIP Tx udp:192.168.0.32:2048: 

SIP/2.0 200 Ok 

Via: SIP/2.0/UDP 192.168.0.32:2048;branch=z9hG4bK-lg7xoa360cuh;rport=2048 


To: ;tag=4de5095638 



Contact: 



[7] 2010/04/28 11:21:27: SIP Tx udp:192.168.0.32:2048: 

SIP/2.0 487 Request Terminated 

Via: SIP/2.0/UDP 192.168.0.32:2048;branch=z9hG4bK-lg7xoa360cuh;rport=2048 


To: ;tag=4de5095638 



Contact: 

Supported: 100rel, replaces, norefersub 

Allow-Events: refer 


2 

3


Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE 

Accept: application/sdp 



[7] 2010/04/28 11:21:27: SIP Tx udp:192.168.0.31:1024: 

CANCEL sip:50@192.168.0.31:1025;line=npbudr1b SIP/2.0 

Via: SIP/2.0/UDP 192.168.0.33:5060;branch=z9hG4bK-9611418f8bbb34f24b5482fb861 

59ddd;rport 

From: “Fifty One” ;tag=33392 

To: “Robyn Wright” 

Call-ID: a8c5d213@pbx 




[7] 2010/04/28 11:21:27: SIP Rx udp:192.168.0.31:1024: 

SIP/2.0 200 OK 


59ddd;rport=5060 


To: “Robyn Wright” ;tag=55ay4g1375 




[7] 2010/04/28 11:21:27: Call a8c5d213@pbx#33392: Clear last request 

[7] 2010/04/28 11:21:27: SIP Rx udp:192.168.0.31:1024: 

SIP/2.0 487 Request Terminated 


59ddd;rport=5060 







[7] 2010/04/28 11:21:27: Call a8c5d213@pbx#33392: Clear last INVITE 

[7] 2010/04/28 11:21:27: SIP Tx udp:192.168.0.31:1024: 

ACK sip:50@192.168.0.31:1025;line=npbudr1b SIP/2.0 


59ddd;rport 




CSeq: 21261 ACK 


Contact: 


[5] 2010/04/28 11:21:27: INVITE Response 487 Request Terminated: Terminate 

a8c5d213@pbx 

[7] 2010/04/28 11:21:27: SIP Rx udp:192.168.0.32:2048: 

ACK sip:50@localhost;user=phone SIP/2.0 

Via: SIP/2.0/UDP 192.168.0.32:2048;branch=z9hG4bK-lg7xoa360cuh;rport 


To: ;tag=4de5095638 


344 



CSeq: 2 ACK 




snom ONE Settings and SIP Traces 

Viewing SIP Traces 

This section shows you how to turn SIP logging on and view SIP traces. 

1. Navigate to Admin > Settings > Logging. 

2. Under General Logging, set the log level to 7. 

3. Under Specific Events, set Log SIP Events to yes. 


5. To view the SIP trace, navigate to Status > Call Log. 

Logfile Syntax 

Figure B-6 shows a sample logfile and the syntax that typically appears in one. 



Date and time 

Transport 

layer 

Transmit 

Log level 

[7] 2010/04/07 14:27:31: SIP Tx tls:187.147.41.75:2137: 

SIP/2.0 200 Ok 

Protocol 

Via: SIP/2.0/TLS 192.168.1.131:2137;branch=z9hG4bK-tbn06xq6tjgv;rport=2137;received=187.147.41.75 

& release 

From: "Maribel" ;tag=dp4b46ul1a 

To: "Maribel" ;tag=bca20066a3 

Call-ID: 3c267053b07e-b0lqd4vaze6r 

Transport layer 

SIP CSeq: 8211 REGISTER 

security 

headers Contact: ;expires=58 

(From and To) Content-Length: 0 Transport 

Receive 

layer 

Packet [7] 2010/04/07 14:27:31: SIP Rx udp:76.127.234.149:5060: 

type REGISTER sip:cs410.snom.com SIP/2.0 

Via: SIP/2.0/UDP 76.127.234.149:5060;branch=z9hG4bKb9f8adb39884fe0e3fd1996e7b97bfd3.0 

Via: SIP/2.0/UDP 76.127.234.149:1030;rport=1030;branch=z9hG4bK-a8ru3e03jsdp.om8vRLKP6dg__ 

From: "Jane Smith" ;tag=lcnnszda0o 

To: "Jane Smith" 

User Call-ID: 3c267027d74e-hmz8xouk5s3x 

agent CSeq: 177 REGISTER 


Contact: ;expires=3600;reg-id=1;q=1.0;+sip.instance=" Call Log. 

The SIP trace can be pasted into an email or a text editor and used for troubleshooting. 

Trunk and Extension Settings 

Settings made from the web interface (see Figure B-6 on next page) will affect what is in the SIP INVITE. 


346 

“Domain” (Trunk setting) 


IP address of PBX 

1. INVITE sip:509@itsp.com;user=phone SIP/2.0 

2. Via: SIP/2.0/UDP 192.168.0.34:5060;branch=z9hG4bK-6574779d5edc4d91e2f 

dc47a8fba26ec;rport 

3. From: “snom ONE” ;tag=677614076 

4. To: 

5. Call-ID: 317ab0eb@pbx 

“Account” (Trunk setting) 

6. CSeq: 23418 INVITE 

7. Max-Forwards: 70 

8. Contact: 

9. Supported: 100rel, replaces, norefersub 

10. Allow-Events: refer 

11. Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE 

12. Accept: application/sdp 

13. User-Agent: snom-PBX/4.0.0.3204 

14. Remote-Party-ID: "Hanna Smith” ; 

party=calling;screen=yes 

15. Content-Type: application/sdp “First Name” 

“Last Name” 

16. Content-Length: 298 

(Extension settings) 

Figure B-6. Trunk Settings 


Supporting Tables 

Table B-7 SIP Headers 


Header Abbrev. Header Abbrev. 

Accept Organization 

Accept-Contact a P-Access-Network-Info 

Accept-Encoding P-Answer-State 

Accept-Language P-Asserted-Identity 

Accept-Resource-Priority P-Associated-URI 

Alert-Info P-Called-Party-ID 

Allow P-Charging-Function- 

Addresses 

Allow-Events u P-Charging-Vector 

Answer-Mode P-DCS-Billing-Info 

Authentication-Info P-DCS-LAES 

Authorization P-DCS-OSPS 

Call-ID i P-DCS-Redirect 

Call-Info P-DCS-Trace-Party-ID 

Contact m P-Early-Media 

Content-Disposition P-Media-Authorization 

Content-Encoding e P-Preferred-Identity 

Content-Language P-Profile-Key 

Content-Length l P-Refused-URI-List 

Content-Type c P-Served-User 

CSeq P-User-Database 

Date P-Visited-Network-ID 

Encryption Dep.* Path 

Error-Info Permission-Missing 

Event o Priority 

Expires Priv-Answer-Mode 

Flow-Timer Privacy 

From f Proxy-Authenticate 

Hide Dep.* Proxy-Authorization 

History-Info Proxy-Require 

Identity y RAck 

Identity-Info n Reason 

In-Reply-To Record-Route 

Join Refer-Sub 

Max-Forwards Referred-By 

MIME-Version Replaces 

Min-Expires Resource-Priority 

Min-SE Response-Key Dep.* 

Retry-After Target-Dialog 


348 


Header Abbrev. Header Abbrev. 

Route Timestamp 

RSeq To t 

Security-Client Trigger-Consent 

Unsupported 

SIP-If-Match User-Agent 

Subject s Via v 

Subscription-State Warning 

Supported k WWW-Authenticate 

Suppress-If-Match 

* These headers have been deprecated. 

Table B-8 SIP Response Codes 

1xx—Informational Responses 423 Interval Too Brief 

100 Trying 424 Bad Location Information 

180 Ringing 428 Use Identity Header 

181 Call Is Being Forwarded 429 Provide Referrer Identity 

182 Queued 433 Anonymity Disallowed 

183 Session Progress 436 Bad Identity-Info 

2xx—Successful Responses 437 Unsupported Certificate 

200 OK 438 Invalid Identity Header 

202 Accepted 480 Temporarily Unavailable 

3xx—Redirection Responses 481 Call/Transaction Does Not Exist 

300 Multiple Choices 482 Loop Detected 

301 Moved Permanently 483 Too Many Hops 

302 Moved Temporarily 484 Address Incomplete 

305 Use Proxy 485 Ambiguous 

380 Alternative Service 486 Busy Here 

4xx—Client Failure Responses 487 Request Terminated 

400 Bad Request 488 Not Acceptable Here 

401 Unauthorized 489 Bad Event 

402 Payment Required 491 Request Pending 

403 Forbidden 493 Undecipherable 

404 Not Found (User not found) 494 Security Agreement Required 

405 Method Not Allowed 5xx—Server Failure Responses 

406 Not Acceptable 500 Server Internal Error 

407 Proxy Authentication Required 501 Not Implemented 

408 Request Timeout 502 Bad Gateway 

409 Conflict 503 Service Unavailable 

410 Gone 504 Server Time-out 

412 Conditional Request Failed 505 Version Not Supported 

413 Request Entity Too Large 513 Message Too Large 

414 Request-URI Too Long 580 Precondition Failure 



415 Unsupported Media Type 6xx—Global Failure Responses 

416 Unsupported URI Scheme 600 Busy Everywhere 

417 Unknown Resource-Priority 603 Decline 

420 Bad Extension 604 Does Not Exist Anywhere 

421 Extension Required 606 Not Acceptable 

422 Session Interval Too Small 


350 



SNOM ONE AND EXCHANGE 

appendix C 

This appendix shows you how to set up Microsoft Exchange 2007/2010 UM with your snom ONE software. 

Although this setup provides basic functionality to get you started, keep in mind that both Exchange 

2007/2010 UM and snom ONE can be customized extensively, which allows for more complex configurations 

than what is shown here. The configuration resembles the setup of the Microsoft Speech Server, as we 

suspect the underlying engines are alike. 

When using snom ONE with Exchange, you can redirect calls, voicemail, and email to the mailbox of the 

Exchange server. You can also use the calendar and the address book. Exchange’s address book allows you 

to initiate outbound calls and perform voice-activated dialing for internal or external calls. Exchange’s auto 

attendant allows you to use your voice to call extensions. 

Note: Exchange does not send an MWI (message waiting indication) when a new message has arrived. 

Configuring Exchange for snom ONE 

1. Install the Unified Messaging role on your Exchange 2007 Server. 

2. Run Exchange Management Console. The following image displays the entries that already exist on 

the Exchange server. 

Create a New Unified Messaging Dial Plan 

1. Using the Exchange management console, select Organization Configuration/Unified Messaging. 

2. Under Actions, select New UM Dial Plan. 

3. Name the dial plan snomone, and set the number of digits in extension numbers to 3. 

4. Click New, then click Finish. 


352 


5. Create a new unified messaging IP gateway. Under Actions, select New UM IP Gateway. Name the 

gateway snomONE. 

6. Select IP Address and enter the IP address of your snom ONE server. 

7. Click the Browse button, and select the snomone dial plan you created. 

8. Click New, then click Finish. 

Associate the Dial Plan with the Unified Messaging Server 

1. Using the Exchange management console, select Server Configuration/Unified Messaging. Under 

Actions, click Properties and then the UM Settings tab. 

2. Under Associated Dial Plans, click the Add button. 

3. Select the snomone dial plan you created. 

4. Click OK. 


Enable Mailbox Recipients for Unified Messaging 

Appendix C: snom ONE and Exchange 353 

1. Using the Exchange management console, select Recipient Configuration/Mailbox. 

2. Select the mailbox you wish to enable and then under Actions, click Enable Unified Messaging. 

3. Next to Unified Messaging Mailbox Policy, click the Browse button and then select snomONE 

Default Policy. 

4. Enter a mailbox extension for this user that matches the user’s extension number on the snom ONE 

server. 

5. Enable Manually specify PIN. 

6. Click Enable. 

7. Repeat this section for each mailbox. 

Configuring the snom ONE Server 

Create a new trunk to connect to Exchange: 

1. Click the Trunks tab. 

2. Name the new trunk MS-Exchange or any suitable name. 

3. Set its type to SIP Gateway. 

4. Click the Create button. 

5. From the list of trunks, click MS-Exchange. 


354 


6. In the Domain field, enter the FQDN or IP address of your Exchange Server (i.e., exchange. 

company.com). 

7. In the Proxy Address field, enter sip:exchange.company.com:5060;transport=tcp, but 

replace exchange.company.com with the FQDN or IP address of your Exchange Server. The 

example assumes the address is 100.200.100.200. 

8. Enable Accept Redirect. 

9. If you want the PBX to do in-band DTMF to out-of-band DTMF (RFC 4733) transcoding on 

this trunk, set Trunk requires out of band-DTMF tones to Yes . For this to work properly, Inband 

DTMF detection must be set to on (go to Admin > Settings > General). 


Figure C-7. Trunk Settings for Configuration with Exchange 2007/2010 UM 

Add the Exchange Gateway to Your Current Dial Plan 

1. From the Dial Plans tab, click the edit icon next to your dial plan. 

2. From the dropdown in the top entry, select MS-Exchange. 

3. Enter 7* as the pattern and * as the replacement. 


Appendix C: snom ONE and Exchange 355 

Note: The digit(s) that are entered into the pattern field must match the digit(s) that were used in 

the External Voicemail System field (Domain > Settings). For example, if you used 99$u in that 

field, then you can use 99* as the pattern in the dial plan. 


Set the snom ONE External Voicemail System 

The redirection target should be a telephone number, rather than a SIP URI, and it must be possible to dial 

that number through the dial plan (the extension must have the permission to do that). The replacement 

fields known from the caller-ID representation in the trunks can be used (e.g., in the image shown below, 

$u will be replaced with the extension number). The redirection to the mailbox works only if the mailbox is 

enabled for that account. 

1. Go to the voicemail settings in the domain, and clear the Mailbox Direct Dial Prefix field; otherwise, 

it will interfere with the mailbox redirection logic and create an endless redirection loop. 

2. Set the External Voicemail System field as explained in the previous section. 

3. Set Offer Camp On to No. 



356 



GlOSSARy 

Address mask 

SIP (Session Initiation Protocol) 

A bit mask used to identify which bits in an IP address correspond to the network portion of the 

Internet address and which ones correspond to the subnet portions of the address. The address mask is 

often referred to as the subnet mask because the network portion of the address can be determined by 

the class inherent in an IP address. The mask is 32 bits long: ones are in positions that correspond to 

the network and subnet numbers, and zeros are in the host number positions. 

Address resolution 

The translation of an Internet address into the hardware address or IP of a network interface card 

(NIC). Because names are easier for people to remember, they are usually used in place of the numerical 

machine address to identify a piece of hardware. However, before instructions can be executed, the 

name must be translated into the numerical address of the hardware. This function is done by Address 

Resolution Protocol (see “ARP”). The terms “address resolution” and “name resolution” are synonymous. 

ANI (Automatic Number Identification) 

The phone number of the calling party. Used for billing and lookup purposes. 

ARP (Address Resolution Protocol) 

A network protocol used for determining a network host’s link layer or hardware address when only 

its Internet layer or network layer address is known. Using the IP address of the target node, the client 

station broadcasts an ARP request onto the network. The node that has that address responds by 

sending back its physical address so that packets can be transmitted. Because ARP requests are broadcast 

onto the network, every station in the subnet is required to process the request. ARP returns the 

layer 2 address for a layer 3 address. 

ATA (Analog Telephony Adapter) 

A device used to connect one or more standard analog telephones to a digital and/or non-standard 

telephone system such as a Voice Over IP based network. 

Bandwidth 

The range from highest to lowest frequencies transmitted on a network. Bandwidth measures network 

capacity. 

BlF (Busy lamp Field) 

A collection of lights or indicators on a phone that allows users to monitor the dialog state of another 

phone/user extension. This is indicated by the LEDs adjacent to the particular key. BLFs are often 

used by receptionists and secretaries to aid in routing incoming calls. 

BRI (basic rate interface) 

An ISDN interface that consists of two 64 kb B-channels and a 16 kb signaling channel. Defined by 

the ITU (International Telecommunication Union) in the I.430. 

Broadcast 

A transmission mode in which a station sends a message to all stations on a network. For example, an 

authorized user of the voicemail system might broadcast a message to all the phones on a particular 

domain to advise users of scheduled downtime for system maintenance. 


358 


CAlEA (Communications Assistance for law Enforcement Act) 

A U. S. wiretapping law passed in 1994 to protect public safety and ensure national security. Its objective 

is to preserve law enforcement’s ability to conduct lawfully-authorized electronic surveillance while 

preserving public safety, the public’s right to privacy, and the telecommunications industry’s competitiveness. 

The law defines the existing statutory obligation of telecommunications carriers to assist law 

enforcement in executing electronic surveillance pursuant to court order or other lawful authorization. 

CAS (content-addressed storage) 

A mechanism for storing information that is in its final form (the object cannot be duplicated or 

modified once it has been stored). CAS assigns an identifier to the files so that they can be retrieved 

based on its content, rather than on their storage location. It is typically used for storing mortgage 

records, insurance policies, health records, and emails. 

CDR (Call Detail Report) 

A record produced by the system which contains the details of a call that has terminated on the system. 

CDRs include the date and time the call started, the number making the call, the number receiving 

the call, and the call duration. 

Client/server 

A network architecture whereby a machine is designated to serve the needs of client machines. Application 

services, such as file storage, network administration, and security are provided by the server 

to the client. Client machines are complete, standalone computers, and servers include mainframes, 

minicomputers, and personal computers. Client/server computing melds personal computing with 

centralized data processing and supports many users simultaneously. 

Codecs (coder decoder) 

A device used to convert binary signals transmitted on their digital networks to analog signals converted 

on their analog networks. 

— GSM - 13 Kbps, 20ms 

— iLBC - 15Kbps - 20ms & 13.3 Kbps - 30ms 

— G.711 - 64 Kbps (alaw/ulaw) 

— G.722 - 48/56/64 Kbps 

— G.723.1 - 5.3/6.3 Kbps, 30ms frame size 

— G.726 - 16/24/32/40 Kbps 

— G.728 - 16 Kbps 

— G.729 - 8 Kbps, 10ms 

— Speed - 2.15 to 44.2 Kbps 

— LPC10 - 2.5 Kbps 

— DoD CELP - 4.8 Kbps 

CPU (Central Processing Unit) 

The central unit in a computer containing the logic circuitry that interprets and executes most of the 

commands from the computer’s hardware and software. It is often called the “brains” of the computer, 

as well as the processor, microprocessor, and central processor. 

DHCP (Dynamic Host Configuration Protocol) 

The TCP/IP protocol for allocating IP addresses dynamically when they are needed. Devices running 

DHCP do not need a pre-configured IP address to join an IP-based network. As devices join the 

network, IP addresses are given, and as devices exit the network, IP addresses are released. DHCP is 

used by Internet service providers (ISPs) to allow customers to join the Internet. The DHCP protocol 


educes system administration workload, allowing networks to add devices with little or no manual 

intervention. 

DID (Direct Inward Dial) 

A feature offered by telephone companies for use with their customers’ private branch exchange (PBX) 

systems. In DID service, the telephone company provides one or more trunk lines to the customer for 

connection to the customer’s PBX and allocates a range of telephone numbers to this line (or group 

of lines) and forwards all calls to such numbers via the trunk. As calls are presented to the PBX, the 

dialed destination number (DNIS) is transmitted, usually partially (e.g., last four digits), so that the 

PBX can route the call directly to the desired telephone extension within the organization without the 

need for an operator or attendant. 

DID numbers are assigned to a communications gateway connected by a trunk to the public switched 

telephone network (PSTN) and the VoIP network. The gateway routes and translates calls between 

the two networks for the VoIP user. Calls originating in the VoIP network will appear to users on the 

PSTN as originating from one of the assigned DID numbers. 

DISA (Direct Inward Service Access) 

The process of how incoming calls are handled by a telephone system. It allows the external subscriber 

to make calls by dialing the extension or trunk subscriber directly without assistance of an operator. It 

compensates for the shortcomings of DIL and DDI. 

DMZ (DeMilitarized Zone) 

A computer or network zone that sits between a trusted zone (typically a LAN) and an untrusted zone 

(typically the Internet). A DMZ is usually separated by firewalls at each border and is used for servers 

(e.g., FTP) which are accessed from untrusted networks. 

DND (Do Not Disturb) 

A feature that prevents an extension from ringing. DND functionality allows users to silence their 

phones for all incoming calls, even calls made to a hunt group or an agent group, and takes precedence 

over call forwarding. DND can be configured with an override permission that allows a designated extension 

to call an extension that has been placed into DND. The DND button can be used to activate 

DND on a single phone; however, a user will need to activate the *78 star code so that the system will 

know to put the extension and any additional phone number(s) that have been configured through it 

into DND. 

DNIS (Dialed Number Identification Service) 

A telephone service that identifies for the receiver the telephone number that was dialed by the caller. 

Once the call enters the PBX system, the DNIS will identify which number was dialed and record that 

information. 

DNS (domain name system) 

The Internet’s name/address resolution service that translates alphabetic domain names into numeric 

IP addresses. For example, the domain name www.pbx.com might translate to 198.105.232.4. If 

a computer cannot access DNS, the user’s web browser will not be able to find web sites and the user 

will not be able to receive or send email. The DNS system consists of three components: DNS data, 

name servers, and Internet protocols for getting the data from the servers. 

Domain name server 

A computer that runs a program that converts a fully qualified domain name (FQDN) into its numeric 

IP address and vice versa. 


359

360 

DTMF (Dual-Tone Multi-Frequency) 


The signal that is generated when a user presses the touch keys of an ordinary telephone. Also known 

as “Touchtone,” DTMF has essentially replaced pulse dialling. When a user presses touch keys, two 

tones of specific frequencies are generated (one from a high-frequency group and the other from a lowfrequency 

group), so it’s impossible for the voice to imitate the tones. 

E1 (or E-1) 

Similar to the North American T-1, E1 is the European format for digital transmission. E1 carries 

signals at 2 Mbps (32 channels at 64 kbps, with 2 channels reserved for signaling and controlling), and 

the T1 carries signals at 1.544 Mbps (24 channels at 64 kbps). E1 and T1 lines may be interconnected 

for international use. 

E.164 [RFC 2916] 

An international numbering plan that provides unique phone numbers worldwide. Each assigned 

number contains a country code (CC), a national destination code (NDC), and a subscriber number 

(SN). An E.164 number can be up to 15 digits. E.164 may be replaced with ENUM, which is currently 

being worked on by the ITU and the IETF. ENUM will encompass both traditional analog 

phones and digital devices, including computers and other devices on the Internet. See also “ENUM.” 

E911 (Enhanced 911) 

A location technology advanced by the FCC that allows mobile phones to identify to emergency 

dispatchers the location from which a wireless call is being made. A 911 call made from a traditional 

phone with ground wires is routed to the nearest public safety answering point (PSAP). The PSAP 

receives the caller’s phone number and the exact location of the phone from which the call was made 

and distributes the emergency call to the proper services. In contrast, 911 callers using a mobile phone 

had to access their service providers in order to get verification of subscription service before the call 

was routed to a PSAP. A 1996 FCC ruling changed this. Now a 911 call must now go directly to the 

PSAP without receiving verification of service from a specific cellular service provider. The call must 

be handled by any available service carrier even if it is not the cellular phone customer’s specific carrier. 

The FCC rules also require that all mobile phones manufactured for sale in the United States after 

February 13, 2000, that are capable of operating in an analog mode, must include Enhanced 911 support. 

Since May 19, 2005, E911 support is required for certain VoIP service. 

ENUM (Electronic NUMbering) [RFC 3761] 

A protocol from the IETF that uses the domain name system (DNS) to convert a telephone number 

to an IP address and vice versa, enabling it to be resolved by the Internet’s DNS system like traditional 

Web site domains. The goal of the ENUM standard is to provide a single number to replace the 

multiple numbers and addresses for an individual’s home phone, business phone, fax, cell phone, and 

e-mail. 

FTP (File Transfer Protocol) 

A standard Internet protocol used to upload and download files between computers that are connected 

to the Internet. FTP uses the Internet’s TCP/IP protocols as does HTTP, which transfers displayable 

Web pages and related files, and SMTP, which transfers e-mail. 

FQDN (fully-qualified domain name) 

The complete domain name for a specific computer, or host, on the Internet. The FQDN consists of 

three parts: the hostname, the domain name, and the top-level domain (e.g., www.webopedia.com and 

sip.itsp.com). 


FXO (Foreign eXchange Office) interface 

A device usually found on the customer end (e.g., a plug on a phone or fax machine, or the plug on 

an analog phone system) that receives the analog line. Because the FXO port is attached to a device, 

such as a fax or phone, the device is often called an FXO device. FXO and FXS are always paired. See 

“FXS.” 

FXS (Foreign eXchange Subscriber) interface 

A device usually located on the telephony company’s property that sends power through a channel to a 

phone on the other end. It provides battery power, a dial tone, and a ring. By necessity, it must connect 

to an FXO device. See “FXO.” 

GSM (Global System for Mobile communication) 

A wireless telephone standard in Europe and other parts of the world. GSM uses a variation of time 

division multiple access (TDMA), which is the most widely used of the three digital wireless telephony 

technologies (TDMA, GSM, and CDMA). GSM digitizes and compresses data, then sends it 

down a channel with two other streams of user data, each in its own time slot. It operates at either the 

900 MHz or 1800 MHz frequency band. 

H.323 

A carrier-grade communication protocol that has been approved by the world governments as the international 

standard for voice, video, and data conferencing. It defines an entire, unified system for 

performing these functions and leverages the strengths of the IETF and ITU-T protocols. 

Hub 

A device for connecting one or more computers to other computers, networked devices, or to other 

networks. Compared to routers and switches (which are also connecting types of devices), a hub is the 

least complicated way to get the job done. Hubs work at the physical layer (layer 1) of the OSI model. 

All computers on the hub are able to see everything all the other computers on the hub see. Also see 

“switch” and “router.” 

IETF (Internet Engineering Task Force) 

A body organized by the Internet Society (ISOC) that defines standard Internet operating protocols. 

Its mission is to produce high-quality, relevant technical documents, known as Requests for Comments 

(RFC), that influence the way people design, use, and manage the Internet. The IETF is 

supervised by the ISOC’s Internet Architecture Board (IAB), and its members are drawn from ISOC’s 

individual and organization membership. See also “ISOC.” 

IP-PBX (Internet Protocol Private Branch Exchange) 

A telephone switch (see “PBX”) located on a customer’s premises that utilize VoIP to manage and 

deliver calls. 

IPv4 

IPv6 

A revision of Internet Protocol that uses 32 bit addressing. See also “IPv6.” 

A network packet routing protocol that uses 128 bit addressing. IPv6 was developed to address the 

shortcomings of IPv4, such as the 32 bit address that limited the maximum number of devices that 

could be addressed and to extend the capabilities of IP to meet future QoS capabilities. IPv6 addresses 

have 8 hexadecimal numbers, separated by colons (e.g., 0800:5008:0001:0000:0002:1005:AA 

BC:BA43). 


361

362 


ISDN (Integrated Services Digital Network) 

An international communications standards for simultaneous digital transmission of voice, video, 

data, and other network services over the traditional circuits of the public switched telephone network. 

ISDN supports data transfer rates of 64 Kbps (64,000 bits per second). BRI (basic rate interface) and 

PRI (primary rate interface) are the two types of ISDNs. The popularity of ISDN has decreased in 

areas where DSL and cable modem service is offered. 

ISOC (Internet Society) 

A nonprofit organization founded in 1992 to provide leadership in Internet related standards, education, 

and policy. ISOC is supported by more than 90 organizational members and 26,000 individual 

members. 

ITSP (Internet Telephone Service Provider) 

A company that offers an Internet data service for making telephone calls using VoIP. Most ITSPs 

use SIP, H.323, or IAX for transmitting telephone calls as IP data packets. Customers may use VoIP 

phones or traditional telephones with an analog telephony adapter (ATA). 

ITU (International Telecommunication Union) 

A telecommunications standards body that is guided by the United Nations. It was founded as the 

International Telegraph Union in Paris on May 17, 1865. The ITU acts as the global focal point for 

governments and the private sector in developing networks and services and is comprised of more than 

185 countries and produces over 200 standards recommendations annually in the areas of information 

technology, consumer electronics, broadcasting, and multimedia communications. 

IVR (Interactive Voice Response) 

A telephone technology that allows a caller to respond to configured voice menus through voice and 

touch tone. The IVR system responds with pre-recorded audio to further direct callers on how to 

proceed. 

Jitter 

The statistical variance of the inter-arrival times of RTP data packets. If a VoIP device sends out one 

RTP packet every 20 milliseconds for an audio stream, the packets will not necessarily arrive precisely 

every 20 milliseconds, so if the packets are played out as they arrive, the output will produce jitter and 

will be of poor quality. Jitter buffers can be used to change the variable delays into constant delays. 

lAN (local Area Network) 

A computer network covering a small physical area, like a home, office, or small group of buildings, 

such as a school, or an office park. LANs are connected primarily through Ethernet and can be connected 

to other LANs over any distance via telephone lines and radio waves. LANs have a high data 

transfer rate and are not very expensive to set up. See also “WAN.” 

MAC (Media Access Control) address 

A hardware address that uniquely identifies most network adapters or network interface cards (NICs) 

by the manufacturer for identification. The manufacturer’s registered identification number is usually 

part of the MAC address if it was assigned by the manufacturer. The MAC address is used by the Media 

Access Control protocol sub-layer of the Data-Link Layer (DLC) of telecommunication protocols. 

MIPS (million instructions per second) 

An old method for measuring a computer’s speed and power and, by implication, for determining the 

amount of work a computer can do. It measures the approximate number of machine instructions the 

computer can execute in 1 second (i.e., it measures CPU speed). Because there are so many variables 

with computer performance (e.g., varying amounts of time for different instructions, importance of 


I/O speed, etc.), MIPS ratings are not used that often anymore. However, a MIPS rating can give you 

a general idea of a computer’s speed. 

MoH (Music on Hold) 

Recorded music that is used to fill the silence that would be heard by telephone callers who have been 

placed on hold. 

MOS (mean opinion score) 

A measurement that is used to evaluate the quality of human speech at the point of termination on 

any type of phone connection (audio, voice telephony, or video). A group of listeners simultaneously 

hear a reader recite a series of test sentences, presented by a male voice, then repeated using a female 

voice (computer-generated voices are also used). The listeners individually grade the test sentences using 

a scale that ranges from 1 to 5 (“bad” to “excellent”) and the mean score is taken. Anything below 

a 3 generally indicates issues with the signal processing or the bandwidth applied to the signal. Scores 

of 4 or 5 are usually considered stable and within industry standards, and thus acceptable. 

Multicast 

A network technology used for delivering information to a group of destinations at the same time. The 

most efficient delivery strategy is used to ensure that the messages are delivered over each link of the 

network only once. Copies are created only when the links to the multiple destinations split. 

MWI (Message Waiting Indicator) 

The red light for voicemails on a desk phone. 

NAD (Network Access Device) 

A device that provides access to a larger communication network (e.g., connects a LAN to a WAN). 

Also referred to as dialer or least cost router, a NAD reroutes calls through a network to provide the 

best rates for the call destination. A NAD normally includes a router, modem, and a monitored power 

supply. Most NADs can report power failures and reconnect themselves back to the network when 

disconnected. 

NANPA (North American Numbering Plan Administration) 

The numbering plan for the Public Switched Telephone Network for Canada, the United Stated and 

its territories, and the Caribbean. The plan has evolved over time into a system of three-digit area 

codes and seven-digit telephone numbers. 

NAT (Network Address Translation or Network Address Translator) 

The method for translating an IP address used within one network to a different IP address known 

within another network (one network is designated the inside network and the other is the outside 

network). NAT allows as a router, for example, to act as an agent between the public network (e.g., the 

Internet) and a private network (i.e., a local network), which means that a single, unique IP address 

can represent an entire group of computers. 

NTP (Network Time Protocol) [RFC 958] 

A protocol for synchronizing the clocks of computer systems using a set of distributed clients and 

servers. NTP uses UDP on port 123 as its transport layer and uses a jitter buffer to resist the effects of 

variable latency. 

PBX (Private Branch eXchange) 

A telephone exchange that serves a particular business or office, as opposed to one that is owned by a 

common carrier or telephone company and is used by many businesses or the general public. Users of 

the PBX share a certain number of outside lines for making telephone calls external to the PBX. PBXs 


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364 


have evolved over time, beginning as a manual switchboard or attendant console that was operated by 

a telephone operator (circuit switching) to the modern IP PBX. See also “IP PBX.” 

PCM (pulse code modulation) 

A sampling technique that converts analog signals to digital form. The analog signal is represented by 

a series of pulses and non-pulses (1 or 0, respectively), and the PCM performs the analog-to-digital 

conversion of the speech waveform by sampling the signal 8000 times a second, encoding and transmitting 

the samples in a serial bit stream as 8-bit digital words. Two standards are used for coding the 

sample level: the U-Law standard (North America and Japan) and the A-Law standard (used in most 

other countries). PCM makes it possible to digitize all forms of analog data, including full-motion 

video, voices, music, telemetry, and virtual reality. 

POTS (Plain Old Telephone Service) 

The standard telephone service that most homes use. The POTS network is also called the PSTN. 

POTS is generally restricted to about 52 Kbps (52,000 bits per second). Non-POTS telephone services, 

such as ISDN and FDDI, have high-speed, digital communications lines. 

Power over Ethernet (PoE) 

A technology that allows the LAN switching infrastructure to provide power over a copper Ethernet 

cable to an endpoint or powered device. With PoE, the electrical current is not carried by power cords. 

PSTN (Public Switched Telephone Network) 

The network of the world’s public circuit-switched telephone networks. Originally a network of fixedline 

analog telephone systems, the PSTN is now almost entirely digital in its core and includes mobile 

as well as fixed (plain old telephone service, POTS) telephones. The PSTN is largely governed by technical 

standards created by the ITU-T, and uses E.163/E.164 telephone numbers for addressing. 

PRI 

A type of ISDN service designed for larger organizations. PRI service is generally transmitted through 

a T-1 line (or an E1 line in Europe). 

Proxy Server 

A server (a computer system or an application program) that acts as an intermediary for requests from 

clients seeking resources from other servers. The VoIP proxy server is used in a DMZ of a company’s 

secure internal communication network and receives VoIP control messages and VoIP media streams. 

Using the MAC address and source IP address contained in the control message, the proxy server 

pushes a policy change to the internal network’s external firewall to open call control protocol ports 

and Real Time Protocol (RTP) ports only for packets from the source IP address. The VoIP proxy 

server hides the company’s internal network address and directs incoming VoIP packets to an IP-PBX 

connected to the company’s internal network. 

QoS (Quality of Service) 

A broad collection of networking technologies and techniques that are used to provide guarantees on a 

network’s ability to deliver predictable results. In VoIP networking, quality refers to being able to listen 

and speak in a clear and continuous voice, without unwanted noise. When applying QoS to VoIP 

networks, emphasis is placed on latency (packet delivery delays), jitter (variations in packet delivery 

delays), packet loss (dropped packed due to too much traffic in the network), and burstiness of loss 

and jitter. 

RAM (Random Access Memory) 

A form of computer data storage that allows stored data to be accessed in any order (i.e., “random access”). 

RAM is used by a computer’s operating system, application programs, and currently used data, 


so that they can quickly be reached by the computer’s processor. RAM is quickly readable and writeable 

compared to other kinds of computer storage (e.g., the hard disk, floppy disk, and CD-ROM); 

however, data in RAM remains only as long as the computer is running. Once the computer has been 

turned off, RAM loses its data. When the computer is turned on again, the operating system and other 

files are once again loaded into RAM. 

RFC (Request for Comment) 

A memorandum published by the Internet Engineering Task Force (IETF) describing methods, 

behaviors, research, or innovations applicable to the working of the Internet and Internet-connected 

systems. RFCs are used either for peer review or to simply convey new concepts and information. The 

IETF adopts some of the proposals published as RFCs as Internet standards. 

Router 

A device for connecting one or more computers to other computers, networked devices, or to other 

networks. Compared to hubs and switches (which are also connecting types of devices), a router is 

the smartest and most complicated of the three. Routers can be programmed to understand and route 

the data its being asked to handle. Configuration is done through a user interface. Larger routers are 

capable of being programmed to communicate with other routers to determine the best method of 

getting network traffic from point A to point B. Hubs work at the data link and network layers (layers 

2 and 3) of the OSI model. 

RTCP (Real-time Transport Control Protocol) [RFC 1889, RFC 3550] 

An Internet protocol used to synchronize streams (e.g., the time stamps of a video stream with the 

time stamps of an audio stream). RTCP works hand in hand with RTP (which does the delivery of the 

actual data) and sends control packets to participants in a call (in a manner scalable to large multicast 

networks). The primary function is to provide feedback on the quality of the data distribution being 

provided by RTP. See “RTP.” 

RTP (Real-time Transport Protocol) [RFC 1889, RFC 3550] 

An Internet protocol used for transmitting real-time data, such as audio, video or simulation data, over 

multicast or unicast network services. RTP addresses jitter and out-of-sequence datagrams by assigning 

timestamps and sequence numbers to the packets. The sequence number in the RTP header enables 

the receiver to put the RTP packets in order once they are received. The timestamps are used to restore 

the original timing relationship of the data. 

SBC (session border controller) 

A device used in Voice over Internet Protocol (VoIP) networks to control the signaling (and even the 

media streams) involved in initiating, conducting, and terminating telephone calls or other interactive 

media communications. Its purpose is to ease the load on the call agents within the network. A common 

location for an SBC is in the communication path between any two parties engaged in a VoIP 

session. A stand-alone SBC is often placed at a connection point, or a border, between a private local 

area network (LAN) and the Internet. 

SDP (Session Description Protocol) [RFC 2327] 

An Internet protocol that specifies how the information necessary to describe a protocol should be 

encoded. It includes information such as IP addresses, port numbers, and times and dates when the 

session is active. 

SIP (Session Initiation Protocol) [RFC 3261, 3262, 3263, 3264, and 3265] 

A signalling protocol for initiating and terminating an interactive user session that involves multimedia 

elements such as video, voice, chat, gaming, and virtual reality (it is used mainly for voice and 

video calls over the Internet or data networks). 


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366 

SIP Trunk 


A service offered by an ITSP that allows businesses that have a PBX for their internal calls to use VoIP 

to go outside the enterprise network by using the same connection as the Internet connection. Before 

SIP trunks can be deployed, a business must have a PBX with a SIP-enabled trunk side, an enterprise 

edge device that understands SIP, and an ITSP. See “ITSP.” 

SlA (Service level Agreement) 

A contract between a network service provider and a customer that details the services that the network 

service provider will furnish to the customer. The details of the SLA typically expand upon the 

services, priorities, responsibilities, guarantees, and warranties. Specifics might include the percentage 

of time that the service will be available, the number of users that can be served simultaneously, dial-in 

access availability, usage statistics, etc. The SLA allows the customer to measure, justify, and even compare 

the services with those of outsourcing network providers. 

SNMP (Simple Network Management Protocol) [RFC 1157] 

A UDP-based network protocol used mostly in network management systems to monitor networkattached 

devices for conditions that warrant administrative attention. It consists of a set of standards 

for network management, including an application layer protocol, a database schema, and a set of data 

objects. 

SOAP (Simple Object Access Protocol) 

An XML message-based protocol specification that allows applications running on a decentralized, 

distributed environment to exchange information. It relies on Extensible Markup Language (XML) as 

its message format, and usually relies on other application layer protocols, such as RPT and HTTP for 

message negotiation and transmission. A typical example of how SOAP can be used is as follows: A 

SOAP message is sent to a web service enabled web site (for example, a Call Detail Record database) 

with the necessary parameters to populate a database. The site then returns an HTML-formatted 

document that includes a table taken from the database that lists the calls for the day. the requested 

data (prices, location, features, etc.). The data is then integrated directly into a third-party site. 

Softswitch (software switch) 

A term used to describe the software that is used to bridge a public switched telephone network 

(PSTN) and VoIP. This is done by separating the call control functions of a phone call from the media 

gateway (transport layer). The softswitch is typically used to control connections at the junction point 

between circuit and packet networks. 

SRTP (Secure Real-time Transport Protocol) [RFC 3711] 

A protocol that provides confidentiality, message authentication, and replay protection to Internet 

media traffic such as audio and video. SRTP is an extension of RTP. 

STUN (Simple Traversal of UDP through NAT) [RFC 5389] 

A protocol for assisting devices behind a NAT firewall or router with their packet routing. The main 

objective of STUN is to overcome some of the problems associated with the lack of standardized 

behaviors in NATs. STUN works with a variety of NATs and application programs. However, STUN 

does not allow incoming UDP packets through symmetric NATs and cannot be used to obtain an 

Internet Protocol address (IP address) to communicate with a peer behind the same NAT. 

Switch 

A small device for connecting one or more computers to other computers, networked devices, or to 

other networks. Compared to a hub (which is also a connecting type of device), the switch is more 

intelligent. It is capable of picking up on traffic patterns and learning where particular addresses are. 


In this way, it sends traffic only where it needs to go rather than to every port, making the network 

significantly faster. Switches work at the data link layer (layer 2) of the OSI model. 

T1 (also called channelized T1) 

A digital carrier modulation method in which a T1 line is divided into 24 channels, each having a 

maximum data speed of 64 thousand bits per second (Kbps), and each capable of supporting a unique 

application that can run concurrently with, but independently of, other applications on different 

channels. 

T.38 [RFC 3362] 

A protocol that describes how to send a fax over a computer data network. T.38 allows a fax to be converted 

into an image and then sent to other T.38 fax devices, where it is converted back to an analog 

fax signal. T.38 is supported by most VoIP gateways and ATAs. 

TCP (Transmission Control Protocol/Internet Protocol) [793] 

The communication protocol for the Internet. TCP/IP defines the rules that computers must follow to 

communicate with each other over the Internet. It can also be used as a communications protocol in a 

private network (either an intranet or an extranet). 

TFTP (Trivial File Transfer Protocol) [RFC 1350] 

A simple, less-capable form of the File Transfer Protocol (FTP). TFTP uses the User Datagram Protocol 

(UDP) and provides no security features, so it is typically used where user authentication and 

directory visibility are not required (e.g., by servers to boot diskless workstations, X-terminals, and 

routers). 

TlS (Transport layer Security) [RFC 4346, 5246] 

A protocol that ensures privacy between a client and a server. TLS authenticates both the client and 

the server and creates an encrypted connection between the two. TLS provides protection against 

eavesdropping by a third party and protection against tampering with a message. TLS is the successor 

to the Secure Sockets Layer (SSL). 

UDP (User Datagram Protocol) [RFC 768] 

A communications protocol that offers a limited amount of service when messages are exchanged between 

computers in a network that is using the Internet Protocol (IP). UDP merely performs IP traffic 

demultiplexing based on UDP port numbers, after which it provides a checksum that can be used by 

end systems to determine whether the datagrams received were corrupted by the network. 

Unicast 

The transmission method used for sending messages to a single network destination host on a packet 

switching network. Unicast messaging is used whenever a private or unique resource is requested. 

UPnP (Universal Plug and Play) 

A set of networking protocols designed to enable simple and seamless connectivity among consumer 

electronics, intelligent appliances, and mobile devices from many different vendors. 


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368 


URI (Uniform Resource Identifier) [RFC 2396] 

A string of characters for identifying all types of names and addresses on the Internet (e.g., a page of 

text, a video, a program, a SIP proxy). It enables devices the ability to interact with the resource over 

a network (typically the Internet) using specific protocols. Schemes specifying a concrete syntax and 

associated protocols define each URI. 

Mechanism to 

Access Resource 

Computer that Houses 

Resource Name of Resource 

http:// www.snom.com UserFiles/File1.pdf 

https:// www.my-cdr-domain.com cdr.php 

sip 123@domain.com 

VlAN (Virtual local Area Network) 

A software concept whereby a group of hosts with a common set of requirements communicate as if 

they were attached to the same broadcast domain, regardless of their physical location. A VLAN has 

the same characteristics as a physical LAN, but end stations in a VLAN can be grouped together even 

if they are not located on the same network switch. Network reconfiguration can be done through 

software instead of physically relocating devices. 

VoIP (Voice over Internet Protocol) 

A technology that uses Internet Protocol (IP) to transmit telephone calls over the Internet or a private 

internal network, rather than using circuit transmissions of the PSTN. With VoIP, the physical 

telephone is not necessary; users can communicate without a telephone set. A primary attraction of 

VoIP is its ability to reduce expenses, since telephone calls travel over the data network rather than the 

phone company’s network. 

WAN (Wide Area Network) 

A computer network that covers a broad area (e.g., any network that links across metropolitan, regional, 

or national boundaries). WANs are similar to the Internet in that they are not owned by a 

single organization. They exist under collective or distributed ownership and management. For WAN 

connectivity over the longer distances, ATM, frame relay, and X.25 are used. Computers connected to 

a WAN can be connected via the telephone system, leased lines, or satellites. WANs have a lower data 

transfer rate when compared to LANs. See also “LAN.” 

XMl (eXtensible Markup language) 

A flexible way to create common information formats and share both the format and the data on the 

Internet, intranets, etc. XML and HTML are similar, in that they both contain markup symbols to 

describe the contents of a page or file. However, HTML focuses on how data looks, whereas XML 

focuses on what data is. In HTML, text that is placed between markup tags and 

will simply take on the appearance (font, size, etc.) of a “title tag.” However, if the same text were 

placed between XML markup tags and , it would likely mean that the data 

was a title of a book. Then depending on how the application in the receiving computer wanted to 

handle the book title, it could be stored or displayed on a website with a selection of other books that 

can be purchased. 


INDEX 

Symbols 

911 54 

A 

access list 

blocking IP addresses 43 

address book 

add agent group callers to 146 

and anonymous calls 285 

and CMC 56 

create using a CSV file 324–330 

domain address book 275 

upload address books to phone 276 

user’s personal address book 273–276 

administrator 

admin password 24 

admin username 24 

resetting admin password 24 

web interface for admin 15 

administrator authority 

control over accounts 30 

control over ANI 30 

control over dial plans 30 

control over trunks 30 

control settings of user interface 45 

agent groups 141–156, 284 

add caller to address book 146 

agents that can jump out 145 

and cell phone settings 307 

and DND 289 

feature overview 141–142 

logging in and out 153–154, 290 

message-music cycle 143–144 

monitoring 148 

naming the agent group 144 

night service 152–153 

queueing 142 

queue manager 147 

recording agent group greetings 151 

ringback in the queue 150–151 

setting up an agent group 144–156 

the From header 145 

user input handling 150 

ANI 77 

anonymous calls 

blocking your caller-ID 285 

rejecting using *88 286 

attended transfer 282 

audio 

audio__* folder 20 

audio prompts 20 

busy.wav 20 

codec preference 34 

customized ringtones 206–208 

installing new languages 204–206 

IVR language 23 

ringback.wav 20 

tone language 23 

web language 24 

auto attendants 121–134 

black listed callers 124 

dial by name 123 , 128 

direct destinations 129–130 

nesting auto attendants 130–132 

user input 123 

welcome greetings 131–134 

welcome message 121 

B 

black list 

auto attendant 124 

classify a contact 292 

blind transfer 282 

block anonymous calls 286 

block caller-ID 285 

buttons 209–218 

and multiple identities 217 

button functionality 209–212 

button profiles 211–217 

key system configuration 215–217 

C 

call barge 109, 290 

call center 

agent log in/log out 290 

barging in on calls 290 

listening in on calls 291 

teach mode 290 

call details 

request using *63 star code 291 

caller-ID 77, 181 

and anonymous calls 286 

blocking 285 

Index 369 


370 

unblocking 285 


calling card 179–182. See also pre-pay 

account balance 291 

callback 179, 181 

credit for outbound calls 56 

dialing interpretation 181 

DISA 179 

pre-pay 179 

read out balance 182 

setting up the account 180–182 

calling features 

add to white list 292 

agent log in/log out 290 

black list 292 

block caller-ID 285 

call barge 290 

call forward 286 

call forward all calls 287 

call forward on busy 287 

call forward on no answer 288 

call park 283 

call pickup 283 

call record 292 

call return 281 

call teach 290 

clean up extension 292 

conferencing 284 

do not disturb 288 

hot desking 286 

intercom 281 

redial 280 

reject anonymous calls 286 

set night mode for domain 289 

show account balance 291 

transer 281 

wakeup call 291 

call listen-in mode 291 

call park 

park reminder 58 

call park and retrieve 283 

call pickup 283 

call recordings 

system performance 29 

call return 281 

call teach 109 

call teach mode 290 

CDR 

CDR URL setting 26 

CDR (Call Detail Record) 

midnight events 61 

CDRs 251–258 

automatically generated 251–253 

cdre, cdri, and cdrt folders 18 

global configuration file 18 

limiting the number of 26 

request call details 291 

setting a duration to keep 26 

cell phones 

and agent groups 307 

and blind transfer 282 

and conferencing 285 

and DND 288 

and hunt groups 307 

and loss-of-signal events 308 

call a cell phone without knowing number 310 

call from extension 310 

cost savings 304 

matching 55 

personal virtual assistant 311 

retrieve call from cell phone 309 

send calls to 310 

send call to cell phone 309 

specify ring schedule 307 

certificates 

buying 42 

making your own 42 

size and format 42 

system certificate 41 

clean up extension 292 

clean up extension permission 110 

CMC 56 

codecs 

codec preference 33 

G.711 A-law 34 

G.711 U-law 33 

lock codec during conversation 34 

transcoding 33 

conferencing 163–170 

and cell phones 285 

conference star codes 169 

participant list 168 

recording the conference 166 

the ad hoc conference 168–170 

the scheduled conference 164 –165 

three-way conferencing 284 

configuration files 

request configuration files 40 


estore/backup 40 

contacts 

searching from phone display 65 

searching from web interface 65 

CPU 

and agent groups 149 

limiting CDRs on system 26 

processor affinity mask 28 

requirements 2 

setting maximum number of calls 28 

CSV files 315–330 

button profiles 213–214 

CDR to CSV 255–256 

converting Excel to CSV 319–325 

domain address book 324–330 

file requirements 315 

required parameters 316–322 

troubleshooting 329 

D 

dialing 

dial by name 128 

direct inward dialing 75–78 

regular expressions 73–76 

dialing settings 

area code 54 

CMC authentication 56 

country code 53 

emergency numbers 54 

matching cell phone numbers 55 

NANPA 54 

PnP dialing scheme 62 

ROW 54 

dial plans 91–100 

building a dial plan 95–97 

configuring a CMC 91 

configuring a PIN 91 

preferencing 91 

regular expression matching 94 

replacement string 92–95 

sample dial plans 97–100 

wildcard patterns 93–94 

DID. See dialing 

direct destinations 129–130 

disk space. See system performance; CPU 

DND 

call even if set 110 

DND (do not disturb) 288 

and hunt groups/agent groups 288 

set using feature codes 288 

Index 371 

domain 

administering the domain 49–65 

assigning an administrator 50 

billing settings 56 

CDR settings 55 

default domain 52 

domain settings 53–66 

email settings 60 

FQDN 50 

from/to headers 55 


naming the domain 49–51 

recording defaults 60 

setting up a domain 

area code 54 

authentication user/password 63 

configuring domain with ANI 54 

country code 53–54 

dial plan 53 

dial plan scheme 62 

email settings 60 


From header 55 

IVR language 53 


music on hold 53 

naming the domain 53 

offering camp on 58 

park reminder 58 

set no-answer timeout for call forward 58 

time zone 53 

To header 55 

tone language 53 

voice mailbox settings 57 

voicemail settings 56 

web language 53 

voicemail/mailbox settings 56 

domain address book 

create one with CSV file 64 

downloads 

audio prompts 205 

MoH/paging application 245 

ringtone config. file 206 

snom ONE software 8 

DTMF 

inband 30 


E

372 


email 187–196 

add email accounts 188 

blacklist activity 195 

blocking user interface settings 48 

CDRs to email 192 

customizing email notifications 191–192 

Exchange 2007/2010 UM 351 

list of conference participants 168 

send call details to email account 291 

send voicemails to email account 301 

smtp server settings 187 

system notifications 191 

troubleshooting email server 37 

user settings 268–271 

welcome email for user 55 


Exchange 2007/2010 UM 351–356 

extensions 101–110 

admin-only settings 106 

creating new extensions 103–106 

importing bulk extensions 103 

limit users who can call 110 

permission settings 109–110 

registration settings 106–109 

restricting concurrent registrations 29 

working with default extensions 101 

F 

file system 17–26. See also configuration files 

directory tree 17 

folder overview 18, 19–23 

generated folder 233 

global configuration file 17 

of greetings 197 

of recorded phone calls 201–206 

pagmoh.xml 249 

tftp folder 238 

XML files 19 

forwarding calls 

using star codes 

forward all calls 287 

forward calls only when busy 287 

forward calls when no answer 288 


from/to headers 55–56 

G 

greetings 197–202 

agent group greetings 200–202 

auto attendant greetings 198– 199 

personal greetings 197– 198 

recording permission 126 

record new greetings 298 

group mail 301 

H 

hardware 

CPU 

limiting length of voicemails 57 


and agent groups/hunt groups 286 

html folder 22 

customizing HTML files 19 

HTTP 

logging HTTP events 37 

ports 31 

hunt groups 135–140, 284 

and call pickup 284 

and cell phone settings 307 

and DND 289 

customized ringtones 139 

naming the hunt group 137 

night service 140 

permission to monitor account 139 

record incoming calls to 140 

ring stages 135, 136, 138 

the From header 139 

I 

installation 

fresh installation 3 

software upgrade 7 

instant messaging 270 

intercom 110, 281 

IP addresses 

blocking access 43–46 

IVR 

language settings 23 

logging IVR events 37 

IVR node 171– 178 

DTMF match list 174 

external application server 176– 178 

permission to monitor account 175 

recording messages 175– 176 

setting up an IVR node 173– 176 


l

listen in 110 

logging 

log filename 36 

logging specific events 37 

log length 36 

log messages 36 

login levels 5 

M 

messages 

leaving a voicemail message 297 

Microsoft Exchange 

basic setup for Exchange Server 2007/2010 

351–356 


reset block CID 61 

reset DND flags 61 

reset hotdesking 61 

send daily CDR reports 61 

MoH 241–250 

audio_en directory 18 

audio_moh folder 18 

creating WAV files 243–244 

mohs folder 19 

setting up MoH 242 

types of 241–242 

monitoring 

agent group manager 147 

agent group queue 148 

blocking monitoring settings 46 

multicast 

IP addresses 34 

sip.mcast.net 30 

music on hold. See MoH 

MWI (message waiting indicator) 

clear the MWI 302 

N 

NANPA 54 

net start snomone 10 

net stop snomone 9 

night mode permission 110 

P 

paging 157–162 

configuring paging/MoH 245–250 

multicast 158 

multicast IP addresses 161–162 

setting up an account 158–161 

unicast 157 

parameters 

log_filename 10 

max_tcp_length 40 

ptime 34 

reg_status 18 

Index 373 

password 

administrator login 24 

password strength 25 

resetting admin password 24 

special password to bypass users’ passwords 63 

password settings 

blocking on user interface 46 

Personal Virtual Assistant 311 

PIN 

compare to “Trust Caller-ID” setting 54 

number of digits 57 

number of digits required 57 

special PIN to override users’ PINs 63 

plug and play. See PnP 

PnP 219–240 

customizing pnp.xml 238 

default dial plan scheme 62 

DHCP 219, 224–225 

dial plan scheme 62 

generated directory 19 

generated files 233–238 

logging PnP events 37 

manual setup 219, 220, 226–228 

mass deployment 219, 220, 226 

multiple extensions on one phone 230–232 

overriding PnP defaults 231–234 

PnP methods 219–221 

prepping the extension for 220 

provisioning parameters 61–63 

resetting the phone 229 

SIP SUBSCRIBE 219, 220, 225–226 

ports 31–35 

firewalls 33 

HTTP 31 

RTP 33 

SIP 31 

SNMP 34 

TCP/TLS NAT refresh 29 

TFTP 35 

UDP NAT refresh 29 

pre-pay 


374 

credit for outbound calls 56 


pre-pay feature 183–186. See also calling card 

access from extension 185 

access from PVA 186 

dollar amount on extension 185 

rates table 183 

ptimes 

multicast IP addresses 34 

R 

record calls 292 

recorded phone calls 201–206 

record location setting 27 

system-initiated recordings 202 

user-initiated recordings 201 

recordings 

domain defaults 60 

record hunt group calls 140 

recordings folder 19, 23 

redial last number called 

using star codes 280 

redirection settings 

blocking from user interface 47 

return a call 

using feature codes 281 

RFC 2833 30 

RFC 2976 334 

RFC 3262 334 

RFC 3265 334 

RFC 3311 334 

RFC 3428 334 

RFC 3515 334 

RFC 3903 334 

ringback 

audio_moh 18 

ringing transfer 282 

routing 

routing calls to extensions 72–79 

ROW 54 

RTP 

logging media events 37 

ports 33 

S 

screening calls 

anonymous calls 285 

SDP 

ptime parameter 34 

security 

blocking the access of IP addresses 43 

service flags 111–120 

configuring a service flag 112– 115 

linking flag to an account 114– 117 

message/voicemail option 116– 118 

night service 115– 116 

permission to monitor 114 

setting the service flag 117 

using buttons with 118–120 

settings 

changing hidden settings 18 

SIP 

From header 55 

logging SIP events 37 

multicast 30 

ports 31 

retrieve SIP logging 38 

setting registration times 29 

short headers 30 

SIP logging 38 

To header 55 

SIP overview 331–350 

CANCEL request 341 

components 331–333 

REGISTER request 339 

session establishment 337 

session termination 339 

SIP headers 347 

SIP language 333–337 

SIP response codes 348 

SIP traces 344 

SNMP 34 

snom ONE 

B2BUA 1 

CPU requirements 2 

documentation feedback 11 

downloads 8 

logging in 5, 261 

memory requirements 2 

naming your system 23 

restarting system 8 

software installation 3 

software upgrade 7 

web interfaces 15 

SOAP 

trusted IP 26 


star codes 

snom ONE star codes 280 

star codes (by number) 

*00 (call cell phone of extension) 310 

*51 (retrieve call from cell phone) 309 

*52 (move call to cell phone) 310 

*53 (conference) 285, 310 

*61 (show account balance) 291 

*62 (wakeup call) 291 

*63 (request call details) 291 

*64/*65 (login/logout) 290 

*66 (redial) 280 

*67 (block caller-ID - activate) 285 

*68 (block caller-ID - deactivate) 285 

*70 (hot desking) 286 

*71 (forward all calls) 287 

*72 (forward all calls - deactivate) 287 

*73 (forward calls on busy) 287 

*74 (forward calls on busy - deactivate) 288 

*75 (forward calls on no answer) 288 

*76 (forward calls on no answer - deactivate) 288 

*77 (transfer) 282 

*78 (DND) 288 

*79 (DND - deactivate) 288 

*80 (set night mode) 289 

*81 (call barge in) 290 

*82 (call teach mode) 290 

*83 (listen-in mode) 291 

*84 (clean up extension) 292 

*85 (call park) 283 

*86 (call park retrieve) 284 

*87 (call pickup) 284 

*88 (block anonymous calls) 286 

*89 (block anonymous calls - deactivate) 286 

*90 (intercom) 281 

*91 (add call to white list) 292 

*92 (add to black list) 292 

*93/*94 (call record) 292 

*95 (send voicemails to email) 301 

*97 (go to voicemail) 297, 301 

*98 (record new greetings) 298 

*99 (clear message indicator) 302 

support 

online help 10 

snom forum 11 

snom ONE wiki 10 

submitting tickets 11 

symbols 

yellow triangle 53 

system maintenance 


Index 375 

system performance. See also CPU 

binding pbxctrl to fixed CPU 28 

call duration 29 

call recordings 29 

concurrent registrations 29 

maximum number of calls 28 

registration times 29 

ring duration 29 

size of backup file 29 

transcoding 34 

UDP and TCP/TLS NAT refresh 29 

system restart 

from command line 9 

from Windows 8 

when system fails to start 10 

T 

technical support 

online help 10 

snom forum 11 

submitting tickets 11 

wiki 10 

TFTP 

ports 35 

tftp directory 23 

tftp folder 19, 238 

time zone 

setting local time zone 24 

To header 

setting a new To header for the domain 55 

transferring calls 

attended transfer 282 

blind transfer 282 

from mobile device 281 

transfer call to voicemail 282, 299 

troubleshooting 

blocked access (accesslist folder) 18 

CSV files 329 


retrieving SIP logging 38 

setting SIP logging 38 

tracking system activity (logging) 36 

trunks 67–90 

configuring trunks 80–90 

creating trunks 79 

direct inward dialing (DID) 75–78 

inbound calls 70–72 

logging trunk events 37 


376 

V 


outbound calls 77–80 

purpose of 67–68 

regular expressions 73–76 

routing calls to extensions 72–79 

SIP trunks 68–71 

voicemail 

accessing 296 

add a comment to a message 296 

calling own extension 57 

change your PIN 295, 296 

delete a message 296 

direct dial prefix 57 

disk space requirements 57 

envelop information 58 

external voicemail system 57 

fast-forward through messages 296 

go straight to voicemail 301 

group mail 301 

leave a message 297 

log in to voicemail 296 

mailbox escape account 57 

mailbox timeout 56 

message playback 296 

move or copy a message 296 

no answer timeout 58 

offer camp on 58 

PIN - number of digits 57 

receive voicemails in email account 301 

record a greeting 296, 298 

record a welcome message 296 

record your name 296 

repeat a message 296 

request a callback 296 

restrict length of voicemail 56 

restrict number of voicemails 56 

ring my cell phone when voicemail arrives 308 

skip a message 296 

transfer call to another’s voicemail 282 

transfer call to your voicemail 282 

voicemail prompts 20 

voicemail settings 

blocking from user interface 47 

W 

wakeup call 

using star feature codes 291 

WAV files 

MoH 243–244 

web interface 

customizing the accounts page 58–60 

white list 

add a contact 292

snom ONE Technical Manual - Atcom

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