Com.X Administrator Guide 1.3 Ver3.0.5

Com.
X
IP PBX / Gateway
Administrator Guide
(Com.X revision 1.3 software release)
Version 3.0.5, 09 January 2015
2010 2014 Far South Networks
Document History
Version
Date
Description of Changes
3.0.0
26/07/2013
Document ported from release 1.2 version 2.0.5
3.0.1
01/08/2013
Include Ntework and Firewall administration.
3.0.2
26/08/2013
Added 1.3 sip handset provisioning
3.0.3
20/01/2014
Added faxing over IP
3.0.4
26/02/2014
Added VPN and VLAN and network QoS configuration sections. Added
Campon Callback and Autoprovisioning of phones on a VLAN.
3.0.5
09/01/2015
Added Fop2 usuage, Added Polycom handets and disoclaimer on Yealinki

V72 handsets. Introduced Call centre stats and ATMOS voice logger as value
added applications.
Com.X Administrator Guide

Page 3
Table of Contents
1 INTRODUCTION...............................................................................................................................13
1.1 OVERVIEW........................................................................................................................................13
2 MANAGING THE UNIT...................................................................................................................14
2.1 ACCESSING THE UNIT..........................................................................................................................14
2.1.1. Factory network configuration............................................................................................ 14
2.1.2. Authentication......................................................................................................................14
2.1.3. Remote web login.................................................................................................................14
2.2 CONFIGURATION MANAGEMENT............................................................................................................15
2.2.1. GUI layout............................................................................................................................15
2.2.2. Configuration status.............................................................................................................15
2.2.3. Applying and reverting a configuration...............................................................................16
2.2.4. Changing the default administrator's password..................................................................16
2.2.5. Rebooting and Shutting down..............................................................................................17
2.3 SYSTEM MAINTENANCE......................................................................................................................18
2.3.1. Backup and Restore.............................................................................................................18
2.3.1.1 Backup contents........................................................................................................................... 18
2.3.1.1.1 Factory default configurations............................................................................................. 18
2.3.1.1.2 Configuration, Recordings................................................................................................... 18
2.3.1.1.3 Voicemail............................................................................................................................ 18
2.3.1.1.4 CDR..................................................................................................................................... 19
2.3.1.1.5 Excluded from backup......................................................................................................... 19
2.3.1.2 Immediate backup........................................................................................................................ 19
2.3.1.3 Scheduled backup........................................................................................................................ 19
2.3.1.4 Managing existing archives.......................................................................................................... 20
2.3.1.5 Restoring an archive onto the same system .................................................................................. 20
2.3.1.6 Restoring an archive onto a replacement system.......................................................................... 21
2.3.2. Licensing..............................................................................................................................23
2.3.2.1 Adding a License......................................................................................................................... 23
2.3.3. Unlicensed Functionality.....................................................................................................24

2.3.4. ViBE Licensing.....................................................................................................................24
2.3.5. Updating the software..........................................................................................................25
2.3.5.1 Updating the PBX software......................................................................................................... 25
2.3.5.1.1 Managing Packages from the GUI....................................................................................... 25
2.3.5.2 Updating Packages....................................................................................................................... 26
2.3.5.2.1 Check for updates................................................................................................................ 26
2.3.5.2.2 Download Updates.............................................................................................................. 26
2.3.5.2.3 Install/Upgrade.................................................................................................................... 26
2.3.6. Upgrading a 1.2 system to 1.3.............................................................................................27

3 COMMISSIONING THE UNIT....................................................................................................... 28
3.1 GLOBAL SETTINGS ..............................................................................................................................28
3.2 NETWORK CONFIGURATION..................................................................................................................30
3.2.1. Connections..........................................................................................................................30
3.2.2. Ethernet Interface Configuration.........................................................................................31
3.2.2.1 Listing ethernet interfaces ............................................................................................................ 31
3.2.2.2 Configuring an Interface.............................................................................................................. 31
3.2.2.3 Providing DHCP Server IP allocation to clients........................................................................... 33
3.2.2.4 Providing Comma iTA boot services........................................................................................... 35
3.2.3. Configuration of PPPoE Interfaces.....................................................................................36

3.2.4. X.21 Interface configuration...............................................................................................37
3.2.4.1 Frame Relay PVC Configuration................................................................................................. 39
3.2.5. Creating Virtual ports for VLAN and PVC .........................................................................41

3.2.5.1 VLAN.......................................................................................................................................... 41
3.2.5.2 VPN Interface.............................................................................................................................. 44
3.2.5.2.1 VPN Server.......................................................................................................................... 45
3.2.5.2.1.1 Routes.......................................................................................................................... 46

Page 4
3.2.5.2.1.2 Generating certificates for clients ............................................................................... 46
3.2.5.2.1.3 Revoking Client certificates ......................................................................................... 47
3.2.5.2.2 VPN Client.......................................................................................................................... 47
3.2.6. Adding IP Routes to Interfaces............................................................................................ 49

3.2.6.1 Removing IP Routes from an Interface ........................................................................................ 49
3.3 FIREWALL CONFIGURATION.................................................................................................................50

3.3.1. Zones....................................................................................................................................50
3.3.2. Policies.................................................................................................................................51
3.3.2.1 Applying Default Policies............................................................................................................ 52
3.3.2.2 Managing Policies........................................................................................................................ 53
3.3.2.3 Editing Policies............................................................................................................................ 53
3.3.2.4 Creating a New Policy................................................................................................................. 54
3.3.3. Rules.....................................................................................................................................55
3.3.3.1 Port Forward................................................................................................................................ 55
3.3.3.2 Inbound Redirect......................................................................................................................... 57
3.3.3.3 Inbound Access............................................................................................................................ 58
3.3.3.4 Ping.............................................................................................................................................. 59
3.3.3.5 Blacklist....................................................................................................................................... 60
3.3.4. Advanced..............................................................................................................................60
3.3.5. Failed Rules ........................................................................................................................ 60
3.3.6. Network Quality of Service..................................................................................................61
3.3.6.1 Loading network QoS defaults .................................................................................................... 61
3.3.6.2 Editing classification and priority................................................................................................ 62
3.4 HARDWARE DEVICE CONFIGURATION....................................................................................................62

3.4.1. Configured devices...............................................................................................................63
3.4.2. Detecting and scanning for devices.....................................................................................63
3.4.3. Adding Detected Devices.....................................................................................................63
3.4.4. Manually adding devices.....................................................................................................64
3.4.5. Configuring devices............................................................................................................. 64
3.4.5.1 Configuring Comma devices........................................................................................................ 64
3.4.5.2 Configuring Generic devices........................................................................................................ 66
3.5 MANAGED SIP PHONE PROVISIONING.....................................................................................................67

3.5.1. Adding Hardware (Managed SIP Device)...........................................................................68
3.5.1.1 Manually add Managed SIP Phone.............................................................................................. 68
3.5.1.2 Adding an Auto-detected Managed Sip phone............................................................................. 69
3.5.1.3 Provisioning via Spreadsheet based configuration ....................................................................... 72
3.5.1.3.1 Adding hardware via spreadsheet........................................................................................ 72
3.5.1.3.2 Exporting spreadsheet of current hardware and extensions .................................................. 73
3.5.2. Adding a Managed SIP phone from a VLAN interface........................................................73

3.5.3. Removing a hardware device ..............................................................................................75
3.5.4. Restore Factory Defaults of a Managed handset.................................................................75
3.5.5. Configuring Extensions on a Managed Sip phone...............................................................75
3.5.5.1 Enabling Lines............................................................................................................................ 75
3.5.5.2 Assigning Extensions to Managed SIP Devices........................................................................... 76
3.5.5.3 Creating New Extensions............................................................................................................. 76
3.5.5.4 Assigning free extensions to available lines on Managed SIP Device .......................................... 77
3.5.5.5 Configuring a port on the Phone to register as an extension on a remote system ......................... 77
3.5.6. Codecs..................................................................................................................................78
3.5.7. Removing an association to an extension............................................................................79
3.5.7.1 Unassigning Extensions............................................................................................................... 79
3.5.7.2 Deleting Extensions..................................................................................................................... 79
3.5.8. Advanced Configuration options for Managed SIP Devices...............................................79

3.5.8.1 Configuring Soft Keys ................................................................................................................ 80
3.5.8.2 Configuring keys individually...................................................................................................... 80
3.5.8.3 Configuring Multiple DSS keys to map to consecutive extensions. ............................................. 80
3.5.8.4 Adding an Extension Module....................................................................................................... 81
3.5.9. Supported Devices................................................................................................................81

3.6 CONFIGURING COMMA DEVICE PORTS....................................................................................................82
3.6.1. Configuring Analogue Trunk (FXO) ports...........................................................................82
3.6.2. Configuring Analogue Extension (FXS) ports.....................................................................83
3.6.3. Configuring BRI ports..........................................................................................................84

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3.6.4. Configuring PRI...................................................................................................................85
3.6.5. Removing devices.................................................................................................................85
3.6.6. Echo cancellation.................................................................................................................85
3.6.7. Earthing the Com.X..............................................................................................................87
3.6.8. Configuring devices for Fax / modem support.....................................................................88
3.6.8.1 Echo cancellation......................................................................................................................... 88
3.6.8.2 Telco issues.................................................................................................................................. 88
3.6.8.3 Timing......................................................................................................................................... 88
3.6.8.4 Additional configuration for faxing over PRI.............................................................................. 89
3.6.8.5 Fax over IP.................................................................................................................................. 89
3.6.8.5.1 G.711 voice band data (ITU-T V.152)................................................................................. 89
3.7 EXTENSION CONFIGURATION................................................................................................................ 90

3.7.1. Extensions and Extension Ports...........................................................................................90
3.7.2. Listing existing extensions and ports...................................................................................90
3.7.3. Adding Extensions................................................................................................................90
3.7.3.1 Adding extensions from ports...................................................................................................... 91
3.7.3.2 Adding Unassigned extensions.................................................................................................... 91
3.7.3.3 Adding Generic SIP extensions.................................................................................................... 91
3.7.4. Configuring extensions........................................................................................................92

3.7.4.1 Basic configuration...................................................................................................................... 92
3.7.4.2 Advanced configuration............................................................................................................... 93
3.7.4.3 Campon Callback......................................................................................................................... 93
3.7.4.4 Configuring Follow-me / Divert.................................................................................................. 94
3.7.5. Logging into phones with an extension (hot-desking).........................................................96

3.7.6. Configuring SIP extensions..................................................................................................97
3.7.7. Remote SIP extensions.........................................................................................................97
3.7.8. Assigning extensions............................................................................................................97
3.7.9. Unassigning extensions........................................................................................................97
3.7.10. Configuring multiple extensions........................................................................................ 98
3.7.11. Configuring Multiple Sip Extensions from a Spread Sheet...............................................98
3.7.12. Deleting extensions............................................................................................................98
3.8 TRUNK GROUPS AND VOIP PROVIDERS.................................................................................................98
3.8.1. Listing existing trunk groups and VOIP providers..............................................................98
3.8.2. Analogue trunk groups.........................................................................................................98
3.8.2.1 Adding trunk groups.................................................................................................................... 98
3.8.2.2 Editing trunk groups.................................................................................................................... 99
3.8.2.3 Adding trunks to a group........................................................................................................... 100
3.8.2.4 Removing trunks from a group.................................................................................................. 100
3.8.2.5 Removing a trunk group............................................................................................................ 100
3.8.2.6 Viewing group members............................................................................................................ 101
3.8.3. SIP trunks...........................................................................................................................101

3.8.3.1 SIP call setup............................................................................................................................. 101
3.8.3.2 Adding a SIP trunk.................................................................................................................... 101
3.8.3.3 Sip trunk configuration.............................................................................................................. 101
3.8.3.4 Editing a SIP trunk..................................................................................................................... 106
3.8.3.5 Removing a SIP trunk................................................................................................................ 106
3.8.4. Inter-Asterisk Exchange (IAX) trunks................................................................................106

3.8.4.1 Adding an IAX trunk................................................................................................................. 106
3.8.4.2 Editing an IAX trunk................................................................................................................. 108
3.8.4.3 Removing a SIP trunk................................................................................................................ 108
3.9 BUDGET TRUNKS.............................................................................................................................108

3.9.1. Configuring Budget Trunks................................................................................................108
3.9.2. Budget Trunk Behaviour....................................................................................................110
3.10 OUTBOUND ROUTE CONFIGURATION...................................................................................................110
3.10.1. Viewing configures outbound routes................................................................................110
3.10.2. Adding outbound routes...................................................................................................110
3.10.3. Configuring outbound routes...........................................................................................111
3.10.4. Dial Patterns....................................................................................................................111
3.10.4.1 Adding dial patterns to an outbound route............................................................................... 111
3.10.4.2 Dial pattern syntax................................................................................................................... 111
3.10.5. Adding trunks to an outbound route................................................................................112


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3.10.6. Removing Trunks from a route.........................................................................................112
3.10.7. Deleting outbound routes.................................................................................................112
3.10.8. Outbound route priority...................................................................................................112
3.11 INBOUND ROUTE CONFIGURATION......................................................................................................112
3.11.1. Viewing existing inbound routes......................................................................................113
3.11.2. Adding a new inbound route............................................................................................113
3.11.3. Configuring inbound routes.............................................................................................114
3.11.4. Configuring call destinations...........................................................................................115
3.11.4.1 Conferences............................................................................................................................. 115
3.11.4.2 Day/Night................................................................................................................................ 115
3.11.4.3 Extensions................................................................................................................................ 115
3.11.4.4 Ring Groups............................................................................................................................. 115
3.11.4.5 Termination............................................................................................................................. 115
3.11.4.6 Voicemail................................................................................................................................. 115
3.11.4.7 IVR.......................................................................................................................................... 115
3.11.4.8 Phonebook............................................................................................................................... 116
3.11.5. Deleting an inbound route............................................................................................... 116

3.12 FLEXPATH CONFIGURATION.............................................................................................................116
3.12.1. FlexPath Fundamentals...................................................................................................116
3.12.2. FlexPath Route Types...................................................................................................... 117
3.12.2.1 Include..................................................................................................................................... 117
3.12.2.2 Match....................................................................................................................................... 117
3.12.2.3 Start......................................................................................................................................... 118
3.12.3. FlexPath Destinations......................................................................................................118

3.12.4. Viewing existing FlexPaths..............................................................................................118
3.12.5. Adding FlexPaths.............................................................................................................118
3.12.6. Deleting FlexPaths...........................................................................................................119
3.12.7. Adding routes to a FlexPath............................................................................................ 119
3.12.8. Changing route priority in a FlexPath............................................................................120
3.12.9. Editing routes in a FlexPath............................................................................................120
3.12.10. Deleting routes from a FlexPath....................................................................................120
3.12.11. FlexPath use cases........................................................................................................120
3.12.11.1 Class of service..................................................................................................................... 120
3.12.11.2 Partitioning / multi-tenant..................................................................................................... 121
3.12.11.3 Simple partitioned DID routing:............................................................................................ 122
3.12.11.4 Trunk to Trunk gateway........................................................................................................ 122
3.12.11.5 Legacy PBX Remote extension gateway............................................................................... 123
3.12.11.6 Legacy PBX feature set extension......................................................................................... 124
3.12.11.7 Immediate dialing................................................................................................................. 125
3.12.11.8 Transformations.................................................................................................................... 125
3.12.11.9 Assigning DIDs to BRI ports................................................................................................ 126
3.12.11.10 Load-balancing calls........................................................................................................... 126
3.12.11.11 FXO Trunk pick-up to test for trunk service....................................................................... 127
3.13 CALL RECORDING...........................................................................................................................127

3.13.1. Configuring individual extensions...................................................................................128
3.13.2. Recording all calls...........................................................................................................128
3.13.3. On-demand recording......................................................................................................128
3.13.4. Call recording conditions................................................................................................ 128
3.13.5. Gateway Recording..........................................................................................................129
3.13.5.1 Gateway Recording Encryption ............................................................................................... 129
3.13.5.2 Gateway recording in a-law..................................................................................................... 129
3.13.6. Viewing call recordings...................................................................................................129

3.14 POWERING THE COM.X VIA AN UNINTERRUPTED POWER SUPPLY.........................................................130
3.14.1. Configuring Comma-Ups.................................................................................................130
3.14.2. Supported UPS Devices...................................................................................................131
3.15 SECURING THE SYSTEM................................................................................................................... 131
3.15.1. General.............................................................................................................................131
3.15.2. Network............................................................................................................................131
3.15.3. VoIP.................................................................................................................................132
3.15.4. PBX..................................................................................................................................132

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3.16 CALL BEHAVIOR ON NO ANSWER / BUSY...........................................................................................132
3.16.1. Unattended transfer........................................................................................................132
3.16.2. Attended transfer.............................................................................................................133
3.16.3. Call parking....................................................................................................................133
3.16.4. Ring-back........................................................................................................................133
3.16.4.1 Disabling Ringback Functionality............................................................................................ 135
4 ADVANCED CONFIGURATION..................................................................................................137
4.1 ANNOUNCEMENTS............................................................................................................................137
4.2 CALLBACK......................................................................................................................................137
4.3 CONFERENCES.................................................................................................................................139
4.4 DAY / NIGHT CONTROL.................................................................................................................... 140
4.5 FEATURE CODES..............................................................................................................................141
4.6 GENERAL SETTINGS.........................................................................................................................143
4.6.1. Dial command options.......................................................................................................143
4.7 IVR..............................................................................................................................................148
4.8 MUSIC ON HOLD..............................................................................................................................151
4.8.1. Configuring streaming music.............................................................................................151
4.8.2. Playback configuration......................................................................................................151
4.8.3. Adding a music category....................................................................................................152
4.8.4. Adding a streaming category.............................................................................................152
4.9 OUTBOUND ROUTES..........................................................................................................................153
4.10 PAGING........................................................................................................................................154
4.11 INTERCOM.....................................................................................................................................155
4.12 PARKING LOT............................................................................................................................... 156
4.13 PHONEBOOK..................................................................................................................................156
4.14 PIN SETS......................................................................................................................................157
4.15 QUEUES....................................................................................................................................... 158
4.16 RING GROUPS...............................................................................................................................161
4.17 SYSTEM RECORDINGS.....................................................................................................................163
4.18 TIME CONDITIONS......................................................................................................................... 164
4.19 TIME GROUPS...............................................................................................................................165
4.20 VOICEMAIL BLASTING....................................................................................................................166
4.21 DISA......................................................................................................................................... 166
4.22 LISTENING IN ON CALLS / SPY FEATURE.............................................................................................167
5 MONITORING THE SYSTEM......................................................................................................168
5.1 GUI STATUS PAGE......................................................................................................................... 168
5.1.1. Status Display....................................................................................................................168
5.1.2. Services menu.....................................................................................................................168
5.2 DETAILED LOGGING..........................................................................................................................168
5.3 CALL DETAIL RECORDS (CDR)........................................................................................................169
5.3.1. Reviewing Call Detail Records..........................................................................................169
5.3.2. Enabling logging to CSV....................................................................................................170
5.3.3. Accessing the CDR database............................................................................................. 170
5.4 MONITORING OF REMAINING CAPACITY ON THE UNIT..............................................................................170
5.4.1. Deleting of call recordings................................................................................................171
5.4.2. Changing the memory consumption threshold...................................................................171
5.4.3. Auto-removal of audio files................................................................................................171
5.5 RETRIEVING MONITORING INFORMATION...............................................................................................171
5.5.1. Monitoring functions..........................................................................................................171
6 ADVANCED ADMINISTRATION................................................................................................175
6.1 CONSOLE ACCESS.............................................................................................................................175
6.1.1. Authentication....................................................................................................................175
6.1.2. Accessing the unit...............................................................................................................175
6.1.2.1 Serial access............................................................................................................................... 175
6.1.2.2 Secure shell access..................................................................................................................... 175

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6.1.2.3 Editing system files.................................................................................................................... 176
6.1.2.4 Changing the shell login password............................................................................................. 176
6.2 REBOOTING THE SYSTEM...................................................................................................................176

6.3 CONFIGURING EMAIL SERVICES...........................................................................................................176
6.3.1. Configuring voicemail to email..........................................................................................176
6.3.2. Configuring sendmail for mail relay..................................................................................177
6.3.3. Sending a test email........................................................................................................... 178
6.3.4. Monitoring outgoing email................................................................................................ 178
6.4 ASTERISK PBX SOFTWARE MANAGEMENT...........................................................................................178
6.4.1. Accessing the Asterisk PBX console.................................................................................. 178
6.4.2. Stopping and restarting the Asterisk PBX software...........................................................179
6.4.2.1 Restarting Asterisk..................................................................................................................... 179
6.4.2.2 Stopping Asterisk....................................................................................................................... 179
6.4.2.3 Starting Asterisk........................................................................................................................ 179
6.4.2.4 Notify when no calls.................................................................................................................. 179
6.5 FREEPBX ACCESS...........................................................................................................................179

6.6 MANUALLY BACKING UP AND RESTORING THE CONFIGURATION................................................................180
6.7 SETTING THE SYSTEM'S DATE, TIME AND TIMEZONE................................................................................180
6.8 VIDEO CALL SUPPORT (H.263 AND H.263+)......................................................................................181
6.9 RECOVERING / RESETTING THE COM.X COMMA USER PASSWORD.............................................................181
6.10 ADDING STATIC ROUTES TO THE NETWORK.........................................................................................181
6.11 BOOTING THE COM.X IN RECOVERY MODE........................................................................................ 182
6.11.1. Booting in recovery mode................................................................................................182
6.12 VALUE ADDED APPLICATIONS.........................................................................................................182
6.12.1. ATMOS Voice Logger......................................................................................................182
6.12.2. Call Centre Analyser .......................................................................................................182
6.13 THIRD PARTY MODULES.................................................................................................................183
6.14 TELEPHONE MANAGEMENT SYSTEMS (TMS)....................................................................................183
6.14.1. MAN 3000........................................................................................................................183
6.14.1.1 Installing MAN 3000 support.................................................................................................. 183
6.14.1.2 CDR format............................................................................................................................. 187
6.14.1.3 Defining outbound line numbers.............................................................................................. 189
6.14.1.4 Logging records from a specific date....................................................................................... 189
6.14.2. Asterisk TMS systems.......................................................................................................190

6.15 SETTING UP A STREAMING MUSIC SERVER...........................................................................................191
6.16 MICROSOFT OUTLOOK INTEGRATION.................................................................................................192
6.16.1. Outcall..............................................................................................................................192
6.16.2. asttapi...............................................................................................................................194
6.17 FAIL2BAN INTEGRATION.................................................................................................................. 196
6.17.1. Configuration...................................................................................................................196
6.18 VIBE..........................................................................................................................................196
6.19 FLASH OPERATOR PANEL ................................................................................................................197
6.19.1. Flash Operator Panel demo version................................................................................197
6.19.2. FOP2 access....................................................................................................................197
6.19.3. Setting the operator extension..........................................................................................197
6.19.4. Limiting Flash Operator Panel access to operator only................................................. 197
6.19.5. Accessing Flash operator panel.......................................................................................198
6.19.6. Using FOP2.....................................................................................................................199
6.19.7. Generating internal phone calls......................................................................................199
6.19.8. Answering Calls...............................................................................................................200
6.19.9. Transferring calls.............................................................................................................200
6.19.10. Queue Management.......................................................................................................201
6.19.10.1 Adding/removing dynamic agents.......................................................................................... 201
6.19.10.2 Picking up a call from a queue............................................................................................... 201
6.19.11. Listening to calls............................................................................................................202

6.19.11.1 Listen .................................................................................................................................... 202
6.19.11.2 Listen and Whisper................................................................................................................ 202
6.20 NAGIOS........................................................................................................................................202
6.20.1. Hosts.................................................................................................................................203

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6.20.2. Accessing Nagios.............................................................................................................203
6.20.3. Services............................................................................................................................203
6.20.4. Disabling active Nagios Checks on a Service or Port.....................................................204
6.20.5. Customised Monitoring and Reporting............................................................................205
7 TROUBLESHOOTING...................................................................................................................206
Index of Tables
Table 1: Default network interface configuration.......................................................14
Table 2: Global settings............................................................................................30
Table 3: Com.X Firewall default policies. .................................................................53
Table 4: Comma device configuration.......................................................................65
Table 5: Generic device configurationSip Handset Provisioning...............................66
Table 6: FXO port configuration................................................................................83
Table 7: FXS port configuration................................................................................83
Table 8: BRI port configuration.................................................................................84
Table 9: PRI port configuration.................................................................................85
Table 10: Extension configuration.............................................................................92
Table 11: Advanced extension configuration............................................................93
Table 12: Follow-me configuration............................................................................96
Table 13: Trunk group configuration (addition)..........................................................99
Table 14: Trunk group configuration (editing)..........................................................100
Table 15: SIP trunk basic configuration...................................................................102
Table 16: SIP trunk advanced configuration...........................................................105
Table 17: IAX trunk basic configuration...................................................................107
Table 18: IAX trunk advanced configuration...........................................................107
Table 19: Outbound route configuration..................................................................111
Table 20: Inbound route configuration.....................................................................115
Table 21: FlexPath route types...............................................................................117
Table 22: FlexPath destinations..............................................................................118
Table 23: FlexPath configuration............................................................................119
Table 24: FlexPath route configuration...................................................................119
Table 25: Ring-back behavior.................................................................................135
Table 26: Announcement configuration...................................................................137
Table 27: Callback configuration.............................................................................138
Table 28: Conference configuration........................................................................139
Table 29: Day/night configuration...........................................................................141
Table 30: Feature code configuration......................................................................142
Table 31: General PBX configuration settings.........................................................147
Table 32: Interactive voice response configuration.................................................150
Table 33: Playback configuration............................................................................152
Table 34: Music category configuration...................................................................152
Table 35: Streaming music category configuration.................................................153
Table 36: Outbound route configuration..................................................................154
Table 37: Paging configuration...............................................................................155
Table 38: Parking lot configuration..........................................................................156
Table 39: Phone book entry configuration options..................................................157
Table 40: Pin set configuration................................................................................157
Table 41: Queue configuration options...................................................................161
Table 42: Ring group configuration oprions............................................................163
Table 43: System recording configuration...............................................................163
Table 44: Time condition configuration...................................................................164
Table 45: Time group configuration.........................................................................165
Table 46: Voicemail blasting configuration..............................................................166

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Table 47: Logging levels.........................................................................................169
Table 48: Monitoring functions................................................................................174
Table 49: Man3000 CDR format column headers...................................................187
Table 50: Man3000 CDR format field descriptions..................................................189
Table 51: Outcall configuration...............................................................................192
Table 52: asttapi configuration................................................................................195
Table 53: Services and interfaces monitored by the Nagios application.................204
Illustration Index
Figure 1: Com.X web portal......................................................................................19
Figure 2: Remote web login......................................................................................19
Figure 3: GUI layout and configuration status...........................................................19
Figure 4: Review / apply task list...............................................................................20
Figure 5: Start menu options.....................................................................................20
Figure 6: Start menu options.....................................................................................21
Figure 7: System maintenance - backup...................................................................22
Figure 8: The new backup is listed under the appropriate section.............................23
Figure 9: Backups can be scheduled to back up the Com.X settings automatically. 23
Figure 10: The schedule (if configured) is displayed on the title bar. ........................23
Figure 11: Managing individual backup archives. .....................................................24
Figure 12: Save the backup archive to an external location......................................24
Figure 13: Restore the selected backup....................................................................24
Figure 14: Tasks to be executed are listed with an option to cancel.........................25
Figure 15: Submit the desired archive for upload. ....................................................25
Figure 16: Updating the MAC address of a replacement unit....................................26
Figure 17: Manage Licenses via the GUI..................................................................27
Figure 18: Select the appropriate license type before uploading the license.............27
Figure 19: File retrieved and ready for upload...........................................................27
Figure 20: Licenses submitted for authentication......................................................28
Figure 21: Only system maintenance options available from the GUI of a system with
no active licenses......................................................................................................28
Figure 22: Select the Packages tab..........................................................................29
Figure 23: Managed packages listed with version and status...................................29
Figure 24: Options available for managing packages...............................................29
Figure 25: The comx-base-1.2-mpx packages is selected. ......................................30
Figure 26: Global Settings........................................................................................31
Figure 27: Options available for global network configurations. ...............................33
Figure 28: Interface list, status is boldly indicated by colour highlighting...................34
Figure 29: Ethernet LAN port configuration options..................................................35
Figure 30: DHCP and Comma server configuration options. ....................................36
Figure 31: Adding a route to all addresses in the 10.0.0.X range by declaring the
subnet, /24............................................................................................................... 37
Figure 32: Managing IP routes on an interface. .......................................................38
Figure 33: PPPoE interface configuration options. ...................................................39
Figure 34: VLAN and PVC port creation. .................................................................40
Figure 35: VLAN Configuration Options....................................................................41
Figure 36: Ports can be allocated a particular zone into which they fall. ..................42
Figure 37: Ports can be allocated into one of four default zones. .............................42
Figure 38: Policies are listed showing source and destination zones. ......................43
Figure 39: Add the default firewall policies to your configuration. .............................44
Figure 40: Policy Management and preference manipulation. .................................45
Figure 41: Editing an existing Firewall policy. ...........................................................45
Figure 42: Logging Options for firewall policies. .......................................................45

Page 11
Figure 43: Options for new rules and policies. .........................................................46
Figure 44: Configuring a new firewall policy. ............................................................46
Figure 45: Rule Wizard . Types of configurable rules. ..............................................46
Figure 46: Inbound Redirect rule wizard. .................................................................48
Figure 47: Inbound access rule wizard......................................................................49
Figure 48: Ping firewall rule wizard...........................................................................50
Figure 49: Blacklist rule configuration wizard. ..........................................................51
Figure 50: This rule features an unknown interface and so has been disabled and
marked in red............................................................................................................ 52
Figure 51: The network can bge scanned for devices from the Hardware panel.......52
Figure 52: The number of detected devices is displayed on the Hardware panel.
Scanning the network increases the number of detected devices. ...........................52
Figure 53: Selecting the appropriate detected device after a scan of the network. . .53
Figure 54: Selection of device types to be added. ....................................................53
Figure 55: Comma device configuration....................................................................55
Figure 56: A Managed SIP device will already have some fields populated. ............56
Figure 57: Manual Configuration of a Managed SIP Device......................................57
Figure 58: The Com.X forwards and proxies the necessary traffic to the handsets so
that they can sit safely behind a firewall and still be configured remotely..................57
Figure 59: The GUI of the device is accessible through the Com.X GUI by selecting
Connect....................................................................................................................58
Figure 60: Enabling lines on a handset.....................................................................59
Figure 61: Right click the appropriate line in the ports tree to assign it a new
extension..................................................................................................................59
Figure 62: Consecutive extensions can be created and assigned to multiple lines on
the same device........................................................................................................59
Figure 63: Four extensions will be created and assigned to the lines of the same
handset..................................................................................................................... 60
Figure 64: Click and drag the desired line over the desired free extension. Note the
green tick icon on the line being dragged. ................................................................60
Figure 65: Strong passwords are generated for extensions. ....................................60
Figure 66: Extensions can be configured to support multiple Codecs.......................61
Figure 67: Available destinations for DSS keys include
extensions,conferences,queues,ring groups, parking lots and individual parking bays.
.................................................................................................................................. 62
Figure 68: Multiple DSS keys can be configured to transfer to consecutive extensions
.................................................................................................................................. 63
Figure 69: The above procedure results in a configuration as such. ........................63
Figure 70: Keys on the extensions module can also be configured. .........................64
Figure 71: Select Add-> Spreadsheet to add and provision handsets from a
spreadsheet. ............................................................................................................64
Figure 72: Example spreadsheet for Maqnaged SIP Device provisioning and
extension creation.....................................................................................................65
Figure 73: Echo cancellation.....................................................................................71
Figure 74: Assigning extensions...............................................................................74
Figure 75: Trunk group configuration (editing)..........................................................81
Figure 76: SIP trunk basic configuration options.......................................................85
Figure 77: SIP provider deletion confirmation...........................................................88
Figure 78: Budget configuration settings...................................................................90
Figure 79: CDR Call logs........................................................................................110
Figure 80: Configuration and synchronization status display..................................145
Figure 81: Services can be enabled/disabled, restarted and enabled/disabled on start
up. .......................................................................................................................... 145

Page 12
Figure 82: Call record search options.....................................................................146
Figure 83: CDRs can be exported in CSV or PDF format.......................................147
Figure 84: "ifconfig eth0" shows network configurations of eth0 (Lan1)..................152
Figure 85: 'sudo asterisk -rx "stop when convenient" ' stops the asterisk service, so
that no calls are cut off, then 'sudo reboot' reboots the Com.X system. .................153
Figure 86: If an email address is listed in this field, voicemail recordings and
notifications will be sent to the address. .................................................................153
Figure 87: Sendmail configuration..........................................................................154
Figure 88: Man3000 settings menu.........................................................................160
Figure 89: Man3000 port settings...........................................................................160
Figure 90: Man 3000 TCP/IP configuration.............................................................161
Figure 91: Man3000 format settings........................................................................162
Figure 92: Importing a Man3000 format template...................................................162
Figure 93: Outcall dialing rules configuration..........................................................168
Figure 94: Outcall server configuration...................................................................168
Figure 95: Outcall extension configuration..............................................................169
Figure 96: Outlook contacts integration...................................................................169
Figure 97: asttapi configuration...............................................................................170
Figure 98: Calling a contact....................................................................................171
Figure 99: Selecting Asterisk as dial option............................................................171
Figure 100: An Operator extension can be manually configured. ...........................173
Figure 101: FOP access can be limited to the operator only, if required. ...............174
Figure 102: Select "Operator panel" to access the Flash operator panel................174
Figure 103: Login with an extension number and voicemail pin..............................174
Figure 104: Extension 1000 logged in. Notice the nine icons above the extensions
each of which performs a function..........................................................................175
Figure 105: Extension 2000 is logged in on this operator panel. Notice only four
function buttons above the listed extensions...........................................................175
Figure 106: Trunks, queues and parking slots are also displayed. Notice the Sip
Trunk has one active call, indicated by the orange colour and the external caller,
5000........................................................................................................................ 175
Figure 107: Dialling extension 2000 from extension 1000. Notice the blue outline
around extension 2000...........................................................................................176
Figure 108: Notice the change of colour of extension 1000, as well as the incoming
call from "1000" shown on extension 2000..............................................................176
Figure 109: Picking up a call to extension 1003. Pushing the "Call Pickup" button will
connect the caller to the extension of the FOP user. ..............................................176
Figure 110: The call between extension 1000 and external caller 5000 will be
transferred to extension 1001.................................................................................176
Figure 111: Select the active call, then click "Listen"...............................................177
Figure 112: The Listen and whisper feature allows the operator to listen in on other
calls, and talk to the local user on the call. .............................................................177
Figure 113: The landing page of the Com.X Gui.....................................................178
Figure 114: Options available for configuring Nagios monitoring on a port or a
service.................................................................................................................... 180

Page 13
1 Introduction
1.1 Overview
This document is intended for Com.X PBX and Gateway administrators and details
access to, commissioning, management and monitoring of the Com.X range of
products, release 1.3, including the Com.X5, Com.X10 and Com.X2 products. It
forms part of a suite of manuals and guides for the Com.X product range, which also
includes the Com.X End User manual, the Com.X Installation guide and the Com.X
Trouble-shooting guide release 1.2
Note: Com.X systems running 1.2 software can be upgraded to 1.3 software and
licensed.
Com.X gateway products can also be remotely licensed and upgraded to fully
functional 1.3 PBX systems.
Please contact your distributor for pricing and other details.

Page 14
2 Managing the unit

2.1 Accessing the unit
2.1.1. Factory network configuration
The Com.X units ship with 2 - 4 external ethernet ports labeled lan1, lan2 etc.
(enumerated from eth0 in the Operating System). Com.X1 and Com.X10 models
have an internal ethernet port (eth3), and certain Com.X5 models have an internal
port for DSL applications (wan). The ports are factory-configured as follows.
Interface
Models
DHCP?
IP
Netmask
lan1 / eth0
X1, X2, X5,

X10
DHCP Client
lan2 / eth1
X1, X2, X5,

X10
No
192.168.101.1
255.255.255.0
lan3 / eth2
X1, X5, X10
No
192.168.102.1
255.255.255.0
lan4 / eth3
X5
No
192.168.103.1
255.255.255.0
internal / eth3
X1, X10
No
192.168.103.1
255.255.255.0
wan
X5
No
Table 1: Default network interface configuration
Note: Com.X5 systems utilize an internal PCI bus and offer external connection to
the lan4/eth3 network interface.
Warning: Changing the configuration of the internal eth3 interface may result in loss
of service on Com.X1/Com.X10.
2.1.2. Authentication
Web login has been pre-configured with the default administrator username 'admin'
and password 'admin'. It is advisable to change this password directly after
installation.
2.1.3. Remote web login

Once the unit has been powered on and started up, it can be accessed from a
Firefox, Internet Explorer or Chrome web browser with the URL set to the IP address
of the unit. Select the Administrator button to log in. Enter your username and
password and select 'Login' to authenticate. Please see section 2.2.4 for more detail
on changing the administrator's password.

Page 15
Figure 1: Com.X web portal
Figure 2: Remote web login
Note: Should your GUI eventually time out rather than loading correctly, you can
now select to Restart the GUI and to restart the system directly from the web portal
log in.
2.2 Configuration management

2.2.1. GUI layout
The Com.X GUI facilitates status monitoring and control of the unit and peripheral
devices via a set of menus accessible from the menu bar. After any changes have
been made to the configuration, these need to be reviewed and applied before they
take effect.
Figure 3: GUI layout and configuration status
2.2.2. Configuration status

The current configuration status, indicating pending tasks and status alerts, can be
viewed from the configuration menu:
When the status on the Status tab is SYNCHRONIZED, the configuration is up to
date and synchronized with the underlying system configuration resources. If the
status is PENDING or MODIFIED, the Review/Apply button will be emphasized and
the administrator needs to review and apply the configuration using the Review/Apply
button in order to synchronize.
At the top right of the GUI a status icon indicates the over-all system status. If this
indicator is a green check
, the system is synchronized and all services are
running. If the status indicator is an orange/yellow warning symbol

, the
configuration requires audit and apply, or, the system is in error and requires

Page 16
attention. Clicking the status indicator navigates to the Configuration status panel on
the GUI. The unknown indicator shows that system status is not available
(possibly still being acquired after restart or GUI login.)
See section 5.1 for further details on the GUI status page.
2.2.3. Applying and reverting a configuration

Select the Review/Apply button. A list of tasks to be applied will appear. Review the
task list and select Apply to proceed.
To revert changes and return to the configuration last applied, select the Revert
button.
To apply changes without disrupting service to users, select Apply when no calls.
The system will monitor the number of active calls, and if all calls clear, will initiate
apply. The system will remain pending Apply until all calls have terminated.
To continue configuring the system without applying, select Cancel.
During apply, individual task progress and success or failure is indicated. Once the
process has completed, a confirmation is displayed.
Figure 4: Review / apply task list
Figure 5: Start menu options.
2.2.4. Changing the default administrator's password

The password for the default administrative user (admin) can be changed by
selecting the Change Password option from the Start menu. Enter the existing
password, the new password and its confirmation and Accept to apply the change.

Page 17
Figure 6: Start menu options.
Note: changing the administrator's password on the GUI also changes the
underlying advanced FreePBX GUI's password accordingly.
Note: changing the administrator's password does not change the SSH login
password for the system, the ARI call monitor password or the MySQL database
password. Please see section 3.15 for more information on securing the system.
2.2.5. Rebooting and Shutting down

Select Start and then Restart or Shutdown (please wait 2 minutes before power
cycle).
Note: Any changes that have not been applied will be lost. When the system has
been shut down, it remains powered on but halted. Only a physical power cycle will
power it on again.
Note: Startup typically takes 2 3 minutes, but the system will check its hard disk
integrity every 20th start-up. This could delay start-up by a few minutes. If uncertain
of the system state, please connect a serial cable to determine the boot state.

Page 18
2.3 System Maintenance

2.3.1. Backup and Restore
The Com.X configuration can be backed up to the unit's internal hard drive, or
downloaded to another machine. Backups can be scheduled daily, weekly and
monthly, and individual sections of the configuration can be configured for backup,
each with its own schedule. Immediate backups can also be initiated. Backups are
stored on the system in /var/lib/asterisk/backups/
From the Start menu, select System maintenance and then Backup. Please see
section 6.6 for more information regarding manual backup and restore.
Figure 7: System maintenance - backup
2.3.1.1 Backup contents

The following selection of backup sets are provided for backup:
2.3.1.1.1 Factory default configurations
Com.X systems ship with a default factory configuration, which includes the default
configuration of telephony ports in the system. This configuration is reflected on
hardware the Hardware panel. Factory default configurations cannot be deleted from
the GUI, but can be stored offline and may also be uploaded to the system.
2.3.1.1.2 Configuration, Recordings
System configuration, including network, device, trunk, extension, voicemail,
inbound, outbound and FlexPath configurations, as well as all recordings present on
the system, including system recordings, auto attendant and music.
2.3.1.1.3 Voicemail

Page 19
All user voice mailboxes with messages and the listened-to state of the messages.
This could result in a substantial archive.
2.3.1.1.4 CDR
Backs up the system's Call Data Record database.
2.3.1.1.5 Excluded from backup
The following items are excluded from backup and should be provided for by the
system installer, using scheduled scripts or other mechanisms:
Custom FOP2 configuration (FOP2 configuration is generated by the Com.X

GUI on apply. Custom configurations are not backed up.)
NFS configuration
Mail relay configuration Third party module / service configuration (e.g. P$X,
FOP, Queue statistics, PPPoE, TNG, Man3000, device mount configurations,
etc)
Static network routes
2.3.1.2 Immediate backup

To initiate an immediate backup, select Backup now from the Options menu for the
desired backup set. After successful completion, the backup archive will be listed
under the appropriate section.
Figure 8: The new backup is listed under the appropriate section.
2.3.1.3 Scheduled backup

To schedule a backup, select the Options menu for the desired backup set, select
Schedule and select the desired backup frequency. To remove a schedule, select
Manual Only.
Figure 9: Backups can be scheduled to back

up the Com.X settings automatically.
The schedule setting is reflected in the backup set's title bar.
Figure 10: The schedule (if configured) is displayed on the title bar.

Page 20
2.3.1.4 Managing existing archives
As scheduled backups complete and immediate backups are initiated, the existing
archives are listed under the appropriate sections on the backup panel. The most
recent backups appear at the bottom of each section. To sort by a particular column,
click on the column header.
Figure 11: Managing individual backup archives.
Individual backup archives can be managed by right-clicking on the entries. To

download an archive to a local disk, select Save As and follow the browser prompts.
Figure 12: Save the backup archive to an external

location.
2.3.1.5 Restoring an archive onto the same system

To restore from an existing archive, right-click the archive in the list and select
Restore.
Figure 13: Restore the selected backup.
The list of steps to be executed during the restore are displayed. Select Continue to
restore, or cancel.

Page 21
Figure 14: Tasks to be executed are listed with an option to cancel.
Reboot the system once restore has completed (please see 2.2.5).
To restore from a data file saved at an external location, first upload the archive to
the Com.X and then follow the procedure above. Select Upload from the options
menu on the appropriate backup set in order to upload an archive.
2.3.1.6 Restoring an archive onto a replacement system
In case of hardware failure, it might be necessary to restore the configuration of a
system onto a replacement unit.
to restore from a data file saved at an external location, first upload the backup to the
Com.X.
Select upload in the options menu of the relevant section. Enter or browse to, the
location of the backup archive and submit.
Figure 15: Submit the desired archive for upload.
Then follow the procedure in 2.3.1.5 to restore the backup onto the system.
Note: After a configuration has been restored, you will be required to apply the
changes. This gives the opportunity to change any settings (network configurations
in particular may need changing, to avoid loss of service) before they become active
on the system.

Page 22
Once restored, select Start Configuration and then select the Hardware panel. The
restored device should be listed, but the MAC address will not match that of the
device in the replacement unit. As a result some telephony spans may not be
available.
To update the restored configuration with the replacement device's MAC address,
edit the hardware entry (double-click or right-click-edit) and select the selection
button to the right of the MAC address field and select Update from detected.
Note: It is important to note that the telephony port configuration of the replacement
unit must match that of the system to be replaced. The restore process will only list
detected devices for which the port configurations match.
Select the appropriate detected device, accept and apply the changes. A system
restart may be required to ensure all telephony spans are working.
Figure 16: Updating the MAC address of a replacement unit
Note: Updating of the MAC address is necessary only on iTA devices. Units that
contain only MPX telephony devices (such as a Com.X 5 with no external iTA's ) will
not require MAC addresses to be updated.
These archives contain the relevant licenses for Farsouth (gateway or PBX) and third
party software.
Note: If a configuration back up is made on a system featruing budget trunks, the
remaining minutes field is set to the minutes remaining on the system at the time the
back up is made. If this back up is then restored to the system, the trunk will be
created with the remaining minutes field already set. This field should be manually
reset after any configuration archive restoration.

Page 23
2.3.2. Licensing
Licenses are used to enable functionality on 1.2 systems. Licenses are available for
full PBX functionality, gateway and PBX major release upgrades and for certain third
party installations, like ViBE (see section 2.3.4 for more information on ViBE
licensing) and FOP2 (see section 6.19 for further information on FOP2).
2.3.2.1 Adding a License
License archives of the form <serialnumber>_XXXXXXXXXXXXXXXXXXX .zip
obtained from Far South Networks are uploaded to the system via the GUI. These
archives contain the relevant licenses for Farsouth (gateway or PBX) and third party
software. These licenses will only work on the unit for which they are issued.
From the system maintenence page, select the packages tab, and then options,
licensing, and Add License.
Figure 17: Manage Licenses via the

GUI
This will allow you to upload a license from your computer. Select they type of
License you would like to add and then Browse your computer for the correct license
file.
Figure 18: Select the appropriate license type before

uploading the license.
As the license file is retrieved the status bar will turn blue. Accepting will upload the
selected file.

Page 24
Figure 19: File retrieved and ready for

upload.
The .zip folder is then submitted for authentication. After which the GUI restarts.
Figure 20: Licenses submitted

for authentication
After uploading a license to the Com.X, allow a few seconds for the GUI to update
and refresh. If the license is accepted the appropriate GUI functionality will be
available.
Note: It may be necessary to manually restart the GUI from the command line after
adding a license: sudo /etc/init.d/comma-gui restart
2.3.3. Unlicensed Functionality

Unlicensed functionality is presented as grey and is inaccessible in the GUI, as such,
the GUI of a Com.X gateway will show several greyed-out PBX configuration options.
This functionality can be remotely enabled by adding a Com.X PBX license, which
can be purchased from Far South Networks, through your distributor.
Figure 21: Only system maintenance options

available from the GUI of a system with no active
licenses.
2.3.4. ViBE Licensing

In order to operate a ViBE enabled link or network a once-off license is required for
each ViBE-enabled CPE. To activate the ViBE application, this license needs to be
installed on the internal ViBE CPE of the Com.X.

Page 25
ViBE licenses can be purchased through your Far South Networks distributor and are
uploaded to the GUI as in section 2.3.2.1.
Please refer to section 6.18 for further details on the ViBE application.
2.3.5. Updating the software

Com.X software updates are published from time to time on the Far South Networks
repository. The updates are freely available to licensed units with valid software
maintenance contracts.
2.3.5.1 Updating the PBX software
Each Com.X unit ships with the Far South Networks repository configured in
/etc/apt/sources.list:
The software packages on Com.X units need to be updated from time to time to
ensure optimal functinality and to incorporate new features.
Updates are released as a part of two top-level managed packages, namely the
comx-base and comx-gui packages, that can be updated from the Com.X GUI.
Note: Units need to have internet access in order to retrieve update packages from
the Far South Networks repository.
Note: On Com.X5 units, the comx-base package is somewhat different to the

equivalent package for Com.X1, Com.X2 and Com.X10 units. As such the
corresponding package on Com.X5 builds is named comx-base-1.2-mpx.
2.3.5.1.1 Managing Packages from the GUI

The status of the two top-level packages is displayed on the Packages tab of the
System Maintenance page of the Com.X GUI.
Select Start, System Maintenance and then the Packages tab.
Figure 22: Select the

Packages tab
A list showing the com.x-base, comx-gui and vibe

packages, as well as their versions and status is displayed.
Figure 23: Managed packages listed with version and status.
The State column provides information comparing the version of the installed
package to the version of the latest package available on the Far South Networks
repository.

Page 26
Figure 24: Options available for managing packages.
2.3.5.2 Updating Packages

2.3.5.2.1 Check for updates
From the Packages tab of the System Maintenance page, select Check For Updates.
On this command, the Com.X connects to the repository and compares current
package versions to those available.
2.3.5.2.2 Download Updates
To download updates for a particular package, or multiple packages, select the
package in the package list, or select multiple packages using Ctrl-click. A selected
package is distinguished by a blue highlight.
Figure 25: The comx-base-1.2-mpx packages is selected.
With one or more packages selected, click Options and then Download updates.
This will pull the latest packages from the Far South Networks repository and store
them on the Com.X, however it will not install the updates until requested to do so.
This allows updates to be retrieved immediately, but installed at a convenient time,
when service can be interrupted.
Note: If no packages are selected, the GUI will ask if you would like to download
updates for the two top-level packages, comx-base and comx-gui
2.3.5.2.3 Install/Upgrade
Select individual packages or multiple packages as in section 2.3.5.2.2 and then click
on the Install/Upgrade option in the Options list. This will install any updates to the
selected package that have been downloaded but not installed. If there are no
dowloaded updates, then the Com.X will download and install the updated packages
from the Far South Networks repository.
Note: Installation of updates may incur a brief loss of service.

Page 27
2.3.6. Upgrading a 1.2 system to 1.3

Com.X units featuring 1.2 software can be upgraded to use 1.3 software. Please
contact your Far South Networks distributor for further details or pricing information.

Page 28
3 Commissioning the unit

3.1 Global settings
A range of global settings are available for configuration,
Double-click a field to edit the field and enter required information.
Figure 26: Global Settings
Field
Content
External Address
If specified, outgoing SIP packets will include this address as

the source address of the SIP packets.
External is
Dynamic
When true, indicates that the SIP source host specified should
be treated as a dynamic host. This informs the remote peer to
resolve the address of the host for each dialog, in case the

Page 29
Field
Content
address has changed.
Additional local
subnet
The subnet (IP and subnet mask in the form A.B.C.D/E.F.G.H)

for any additional local subnet. Traffic on this subnet will be
treated as local by the Com.X for SIP purposes.
Jitter Buffer
Enabled
When enabled, a SIP jitter buffer will be instantiated for each

SIP call.
Jitter Buffer Max

Milliseconds
Indicates the maximum number of milliseconds that the jitter

buffer is allowed to buffer media before streaming out.
Jitter Buffer
Adaptive
When enabled, uses an adaptive buffering algorithm that

adjusts to the media stream conditions. When disabled, the jitter
buffer length is fixed to the maximum value specified.
Match Digit
Timeout
The timeout in milliseconds after which, if no additional digit is

pressed, the dialed digit series is deemed complete and the call
is attempted.
Notification email
Email address to which to send system notifications .
SMTP host
The mail relay server to use when sending emails (including

notification emails and voicemail to email.)
Please see section 6.3 for a detailed discussion on email
services.
SMTP port
SMTP host network port, default is 25
Masquerading
domain
The domain to masquerade from when sending emails.
SMTP user
SMTP authentication user to use for sending email via the

SMTP host
SMTP password
SMTP authentication password
iTA extra FIFO
Provides additional receive FIFO buffers (4ms instead of 2ms)

for analogue and PRI channels on attached iTAs in case of
network slips
Gateway
recording
If enabled, all calls on all trunks via the Com.X will be recorded.
Gateway
recording
encryption
If enabled, all calls on all trunks recorded will be encrypted

using the Com.X system's unique hardware key. Recordings
can only be decrypted on the same system, or using the
system's unique hardware key.
FOP only
If set to true, this will restrict access to the Flash operator panel
Please see section 3.13.5 for a detailed discussion on gateway

recording.

Page 30
Field
Content
available to
Operator
to the operator extension set in the general settings page of the

advanced menu. Please see Section 6.19 for more detail on the
operator panel.
Ringback on
transfer busy
Enabled by default. If enabled, calls that are transferred to a

busy extension will ring back to the transferring
extensionRingback on transfer no answer
Ringback on
Enabled by default. If enabled, calls that are transferred to an
transfer no answer extension and then unanswered, will ring back to the
transferring extension.
Ringback on
transfer voice
prompt
Enabled by default. If enabled a caller will hear a short

extension unavailble, one moment please message before a
call rings back to the transferring extension.
Table 2: Global settings
3.2 Network configuration

Configuration of all ethernet LAN and Wan ports is possible through the Network tab
of the Comma GUI. This module allows further control of router and firewall settings,
including configurations of zones on a port-by-port basis, general firewall policies,
and individual rules, providing exceptions to the configured policies.
3.2.1. Connections
The connections page allows for configuration of some global networking
parameters.
Double-click a field to edit the field and enter required information.
Figure 27: Options available for global network

configurations.

Page 31
Field
Content
Host Name
The host name assigned to the Com.X.
Domain
The system domain name. Use 'local' to indicate a local domain.
Primary DNS
Server
The system's primary Domain Name resolution Server

Note: the Com.X uses an open-source, synchronous DNS
lookup library. As a result, the Com.X makes use of a DNS
proxy to facilitate DNS lookups. If the primary DNS is
configured to an IP address outside the LAN (i.e. outside of the
network gateway router) and this DNS server or network
connectivity to the DNS server should become unavailable, the
Com.X and the DNS cache might both block on DNS lookup,
resulting in loss of SIP registration and general call failures.
Always configure the primary DNS as the LAN gateway router,
and the secondary DNS as an external DNS server if so
required.
Secondary DNS
Server
The system's secondary Domain Name resolution Server
DNS Associated
Interface
The interface through which the Com.X expects communication

from the DNS server.
NTP Time Server
The NTP server the Com.X should refer to for time

synchronization in the NTP hierarchy.
3.2.2. Ethernet Interface Configuration

The Com.X GUI provides functionality to configure all ethernet interfaces, as well as
provision of DHCP IP allocation for IP phones and booting of Comma iTA devices.
The Com.X unit can serve both DHCP and Comma addresses simultaneously,
provided the IP ranges do not overlap.
3.2.2.1 Listing ethernet interfaces
To access the list of network interface configurations, select Configuration from the
Start menu and then select the Network panel and the Interfaces section.
Interfaces are coloured coded, with active, healthy ports indicated by green
highlighting, and yellow highlights indicating a port is in a warning or down status.
Figure 28: Interface list, status is boldly indicated by colour highlighting.
3.2.2.2 Configuring an Interface

Double-click an ethernet interface to configure the following fields:

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Field
Content
Name
Operating System allocated interface name
Description
A description for the interface
DHCP Client
Determines whether the unit will try and obtain its IP

configuration from a DHCP server. If not selected, the fields
below need to be configured
Enabled
By default each port is enabled. Unselecting this option will

disable the port for all functionality.
IP Address
An IP address unique to the network. Auto-assigned if DHCP

Client is enabled.
Network Mask
The network mask to use on the network. Auto-assigned if

DHCP Client is enabled.
Gateway
The routing gateway for this interface. Auto-assigned if DHCP

Client is enabled.
Default Metric
The metric of a network route is a property used by a routing

protocol to determine whether one particular route should be
chosen over another.
In conflicting cases, traffic will be directed towards the gateway
containing the lowest value metric.

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Figure 29: Ethernet LAN port configuration options.
Apply the configuration.

Note:After applying network configuration changes, the unit needs to be rebooted
or the networking service restarted before the changes become active in the
configuration.
3.2.2.3 Providing DHCP Server IP allocation to clients

Double-click an ethernet interface and select the Servers tab.
Field
Content
DHCP Start Address
The start address for the range of IP addresses from

which new DHCP requests should be served
DHCP End Address
The end address for the range of IP addresses from

which new DHCP requests should be served
DHCP Router Address
The address of the IP router / gateway to be used

by the device requesting IP configuration, normally
the same as the Com.X address on this interface.
DHCP DNS Server
The primary DNS address to be configured on the

device requesting IP configuration, may be left
blank.

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Figure 30: DHCP and Comma server

configuration options.

Note: When applying network configuration changes, the unit automatically restarts
the affected network interface(s). NB. If the interface used for GUI access is
reconfigured, GUI access may be affected.

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3.2.2.4 Providing Comma iTA boot services
Double-click an ethernet interface and select the Servers tab.
Field
Content
Comma Start Address
The start address for the range of IP

addresses to allocate to new Comma
units. Once a unit has been assigned an
IP, the system will remember this
assignment, across restarts, and will
always reserve the IP for that unit. The IP
will no longer be served to future
commas.
Comma End Address
The end address for the range of IP

addresses to allocate to new Comma
units.

Note: On Com.X 1 and Com.X10 units, the eth3 interface is the internal ethernet
bus serving the Com.X Media processor card and should not changed from its
factory configuration.

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3.2.3. Configuration of PPPoE Interfaces
Figure 31: PPPoE interface configuration options.
Field
Content
Name
A suitable name for the virtual interface.
Description
A description of the interface, marking it as PPPoE.
Automatic IP
If enabled, the interface will attempt to dynamically

receive an IP address via IPCP. If left unchecked, a static
IP address can be configured.
Enabled
By default the interface is enabled. Unchecking this

option will disable the interface for all functionality.
IP Address
If Automatic IP is disabled, a static IP address can be

configured in this field. If Automatic IP is enabled then
this field can not be edited from the Gui.
Network Mask
The network mask to use on the network. Auto-assigned

if DHCP Client is enabled.
Gateway Peer IP
The IP address of the bridge mode router handling

PPPoE traffic. This could be an internal router (on select
Com.X5 units,) or another router on the local network.

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PPP Attached Device
The port through which the router can be reached. This

could be a Lan port on the Com.X or the internal dsl
router on select Com.X5 units.
PPP Username
The username of with which to authenticate with your

service provider.
PPP Password
The password associated with your username, used to

authenticate with your service provider.
PPP Authorization
The authorization method used to authenticate your

account with your service provider.
PPP Service name
This field should be left blank, unless your service

provider has requested that your configure your service
with a specific name.
Default Route
If enabled, the PPPoE interface will be configured as the

default route for any IP traffic to an address matching no
other configured route.
Default Metric
The metric of a network route is a property used by a

routing protocol to determine whether one particular route
should be chosen over another.
In conflicting cases, traffic will be directed towards the
gateway containing the lowest value metric.
3.2.4. X.21 Interface configuration

If the Com.X is equipped with an X.21 serial interface (typically for connection to
leased line or frame relay services), the interface is listed as serial0 (X21-1) in the
Interfaces list.
The Com.X X.21 interface is always DTE on the physical layer (although it may act
as protocol DCE in Frame Relay mode). Therefore, the clock and thus the data rate
of the interface is provided by the DCE equipment connected to the X.21 interface,
typically an NTU.
Figure 32: Interface list including serial0, the X21 interface
As with other network interfaces, the interface configuration may be edited by doubleclicking the list entry. The availability of the different fields will depend on the Serial
Protocol chosen.

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Figure 33: X21 interface configuration
The fields are described in the table below. Note that only PPP mode supports
Automatic IP configuration.
Field
Contents
Name (not editable)
serial0
Description (not
editable)
X21-1
Serial Protocol
IP/HDLC Raw IP in HDLC encapsulation

Cisco HDLC Cisco compatible IP HDLC encapsulation
Frame Relay ANSI/CCITT (ITU) compatible frame relay
encapsulation
PPP Point-to-Point Protocol
Frame relay LMI
(Only enabled when Serial Protocol is Frame Relay) Local


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Management Interface protocol selection. Valid values are:
None
ANSI DTE
ANSI DCE
CCITT DTE
CCITT DCE
Automatic IP
(only valid for PPP mode) obtain IP address automatically

from the PPP peer
Enabled
Interface enabled/disabled at startup
IP Address
(except when Frame Relay or Automatic IP enabled) IP

address of this interface
Network Mask
(except when Frame Relay or PPP enabled) Network

mask of this interface
Gateway/Peer IP
(optional, not enabled when Frame Relay or PPP enabled)

Peer IP for point-to-point links, or gateway for multipoint
PPP attached device

(not editable)
(only valid for PPP mode) /dev/ttyX0
PPP username
(only valid for PPP mode) PAP/CHAP username
PPP password
(only valid for PPP mode) PAP/CHAP password
PPP authorization
Method of PPP authorization:

PPP_NONE don't use PAP/CHAP
PPP_AUTO use PAP or CHAP as required
PPP_PAP use PAP only
PPP_CHAP use CHAP only
PPP service name
(optional) PPP provider's service name, normally left blank
Default metric
The metric of a network route is a property used by a

routing protocol to determine whether one particular route
should be chosen over another. In conflicting cases, traffic
will be directed towards the gateway containing the lowest
value metric.
3.2.4.1 Frame Relay PVC Configuration

If the X.21 interface is configured for Frame Relay protocol, it is possible to add
multiple PVCs (Permanent Virtual Circuits), which themselves are separate network
interfaces with the usual configuration options. The list below shows a PVC interface
entry.

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Figure 34: PVC interface shown in interface list
PVCs are added by selecting Options New PVC... The screenshot below shows
the configuration of a new PVC.
Figure 35: X21 PVC device configuration
The fields are described in the table below.
Field
Contents
PVC physical interface X21-1


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PVC encapsulation
IP/Frame Relay Raw IP is encapsulated in the PVC frame

Ethernet/Frame Relay IP over Ethernet is encapsulated in
the PVC frame
DLCI
Data Link Connection Identifier number from 1 to 1023

(normally in the range 16 to 991) that uniquely identifies this
PVC from others sharing the physical link
IP Address, Network
mask, Gateway,
Default metric
Same as for other interfaces
To edit a PVC, double-click its interface entry as usual. To delete a PVC, right-click
on the interface entry and select the Delete option in the menu.
3.2.5. Creating Virtual ports for VLAN and PVC

Virtual ports can be configured from the GUI, to connect to VLAN and PVC networks.
3.2.5.1 VLAN
To create a new virtual port on a Vlan network, navigate to the Connections page,
under the Network tab of the GUI. Then select New Vlan.
Figure 36: VLAN, VPN and

PVC port creation.

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Figure 37: VLAN Configuration Options
Field
Contents
VLAN Physical
Interface
The actual physical LAN interface through which this virtual port
connects to the rest of the virtual network. This could be any of
the LAN ports, or the internal dsl router on certain Com.X5
models.
VLAN ID
A numerical identifier for the VLAN. Integer values from 1 to

4095 are valid VLAN identifiers.
Use 802.1p priority If checked (the default), the VLAN priority field on the incoming
inward
packet will be mapped to a network queue priority on the
Com.X device
Use 802.1p priority If checked (the default), the QoS priority setting of the traffic
outward
source will be set in the VLAN priority field of the outgoing
packets
DHCP Client
If checked, this virtual port will receive an IP address from a

DHCP server elsewhere on the VLAN.

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Enabled
By default the port is enabled. Disabling the port prevents all

functionality and will delete any associated IP routes. Reenabling the port will recreate associated IP routes.
IP Address
If the interface is not configured as a DHCP client, a static IP

address can be configured in this field. If the interface is a
DHCP client, then this field can be read, but not edited from the
GUI.
Network Mask
A network mask can be manually configured along with a static

IP address. If the interface is configured as a DHCP client, this
field will be populated in the DHCP negotiation process and can
be read, but not edited from the GUI.
Gateway
The routing gateway for this interface. Auto-assigned if DHCP

Client is enabled.
Default Metric
Sets the default metric value of routes utilizing this interface.

In conflicting cases, traffic will be directed through the route
containing the lowest value metric.
VLAN ports can be configured to act as DHCP and Comma servers , as in section
3.2.2.3 . Individual and specific IP routes can be associated to the VLAN interface in
a similar was as described in section 3.2.2.2.

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3.2.5.2 VPN Interface
Figure 38: VPN interface configuration page.
Field
Contents
Name
vpns1 VPN server interface

vpncX VPN client interfaces (X=1,2,3...)
Description
Describes the type of VPN interface, automatically

configured to either VPN Server or VPN Client with a
number index.
DHCP Client
Read only. Not applicable to VPN. A VPN server always has

a fixed IP address, whilst a VPN client always receives its IP
address from the remote server.
Enabled
Optional setting to enabled or diasble the interface.

Disabling the interface will remove all associated ip routes.
IP Address
The IP address of the VPN interface.Configurable only for


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VPN servers. The VPN server IP must end in .1.
Network Mask
The network mask of the interface. Automatically configured

on client interfaces.
Gateway
N/A
Default Metric
The metric of a network route is a property used by a routing

protocol to determine whether one particular route should be
chosen over another.
In conflicting cases, traffic will be directed towards the
gateway containing the lowest value metric.
3.2.5.2.1 VPN Server

By default a disabled VPN server port is configured on the Com.X. To configure,
right-click the interface in the interfaces list and select edit. The VPN interface
configuration as in Figure 38 will load. Configure interface IP settings appropriately,
then select the VPN tab.
Figure 39: VPN Server configuration options
Field
Contents
Port Number
The port to be used by clients to connect to the VPN.

Defaults to 1194.

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Use LZO
Compression
The VPN may be configured to use compression on all

packets. This will save network bandwidth but incur some
performance penalty. Client and server configuration must
match.
Transport
Select between TCP and UDP for VPN transport.
VPN Public IP
The Public IP address of the VPN server, if required. This is

used allow client configuration to be generated automatically,
see next section.
VPN Public port
The associated port on the public IP of the VPN, if required.
3.2.5.2.1.1Routes
Routes are added by selecting the Routes tab once configuring the interface. Routes
on VPN interfaces are added as with any other network interface as described in
Section 3.2.6
3.2.5.2.1.2Generating certificates for clients
To generate authentication certificates for client devices, right-click on the interface in
the interfaces list, and select VPN Clients. Select New and enter the name of the
client. Review and apply your changes to create the client.
Figure 40: Select VPN Clients to generate licenses for client devices.
Once a client is created and the configuration applied, a certificate pack and/or
configuration pack for the client can be generated. Right-click the client in the clients
list, and select Get Configuration.

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Figure 41: Certificates are available as standard .p12 or in tar or zip

format, for particular handset devices.
Select your desired certificate packaging and save the file to a location on your
desktop/laptop computer.
Note: Changes must be applied after the creation of a client before requesting
configuration. No client configuration is created before the changes are applied, and
so no configuration will available for unapplied clients.
3.2.5.2.1.3Revoking Client certificates

To revoke certification for a client device, right-click on the desired device in the VPN
Served Clients list, and select Revoke. Clients authenticating with revoked certifcates
will be denied. Note that a certificate, once revoked, cannot be used again. A new
client certificate will have to be issued.
3.2.5.2.2 VPN Client
To add a new VPN client interface, navigate to the network page of the Comma GUI
and select Options, New VPN as per Figure 36.

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Figure 42: Client VPN configuration.
Field
Contents
Port Number
Port number on associated VPN server. Defaults to 1194.
Transport
Protocol to be used for traffic on the VPN. Select between

UDP and TCP.
Remote Server
Address
The IP address or hostname of the VPN server hosting the

VPN.
HTTP proxy IP
address
Optional: the IP address or hostname of an HTTP proxy

server via which the VPN server can be contacted. Only
applicable if transport is TCP.
HTTP proxy port
Required if HTTP proxy is used: the port of the HTTP proxy
HTTP proxy
username:
An optional username for authentication with the HTTP

proxy
HTTP proxy
password
An optional password for authentication with the HTTP proxy
HTTP proxy retry
If checked, try every 5 seconds to connect to the HTTP

proxy if not yet connected
Valid Certificates and A properly configured client requires a valid CA certificate,

Keys
client certificate and client key. These are uploaded using

Page 49
the button below, and may be supplied separately (e.g.
ca.crt, client.crt, client.key) or in a PKCS12 package (e.g.
client.p12)
CA/Certificate/key/PK Click on the ... button to upload certificates from your local
CS12
file system to your client device.
3.2.6. Adding IP Routes to Interfaces

When an interface is configured with an IP address, an ip route matching that
address and the configured subnet of the interface is autoimatically configured,
associating the port with traffic on the subnet.
Additional routes can be added to an interface, ensuring all traffic to the associated
address(es) is directed through the interface.
Routes can be configured for individual addresses as well as for a range of IP
address, by declaring the subnet range after a '/'.
To configure an IP route, select the desired interface and select the Routes tab and
then the add button.
Figure 43: Adding a route to all addresses in the 10.0.0.X range

by declaring the subnet, /24
Click Accept and then review and apply your changes.
Note: After applying network configuration changes, the unit needs to be rebooted
or the networking service restarted.
3.2.6.1 Removing IP Routes from an Interface
Configured routes can be removed from interfaces by selecting the routes tab of the
associated interface, selecting the route, and then clicking the Remove button.
Accept and apply your changes.

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Figure 44: Managing IP routes on an interface.
Multiple routes can be deleted at once, by using CTRL-click to select all routes
marked for deletion.
3.3 Firewall Configuration

With the on-board firewall option enabled, the Com.X can manage firewall services
on your network.
3.3.1. Zones
The Com.X allows separation of its interfaces into different zones, each of which is
authorized to access only portion s of the available network. The access provided to
devices in each zone is configured through firewall policies (See section 3.3.2)
Figure 45: Ports can be allocated a particular zone into which they fall.
Configuration for each port can be edited by double-clicking on the desired

appropriate and then selecting the desired firewall zone.

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Figure 46: Ports can be allocated into one of four default zones.
Zone
Access
Com.X
The Com.X zone contains interfaces (physical and virtual) on the

Com.X system itself. This includes the ports listed on the interfaces
section of the Network page of the Com.X GUI.
LAN
Interfaces placed in the LAN zone have access only to the internal
network. The internet and any external destination is only reachable
through another device in the DMZ.
DMZ
The De-militarized zone has access both to the internet and a limited
partition of the internal network. Interfaces in this zone are typically
used to provide services to users on the external network.
Internet
This zone holds any devices on the external internet.
3.3.2. Policies
Policies are used to control the access available to each firewall zone (see section
3.3.1) Policies are explicitly configured to provide a standard way of dealing with
traffic moving from one zone to another.
Figure 47: Policies are listed showing source and destination zones.
A policy can be configured to respond in one of three ways:

Policy
Action

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Accept
Allow traffic between the two zones.
Drop
Ignore any packets travelling from the source zone to the

destination zone.
Reject
Reject any packets sent from the source zone to the destination
zone with a rejection message.
3.3.2.1 Applying Default Policies

A comprehensive suite of firewall policies are available for restoration on each
system. These policies include what we at Far South believe should be a good out
of the box policy configuration that would need particular rules and perhaps
additional policies to be configured in order to meet the needs of the client.
To add the default policy suite to your configuration, select the Firewall page of the
Network tab, and then the Load Defaults option from the options list.
Figure 48: Add the default firewall

policies to your configuration.
Default Policies:

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Source Zone
Destination Zone
Action
Com.X
Internet
Accept
Com.X
LAN
Accept
Com.X
DMZ
Accept
LAN
Com.X
Accept
LAN
Internet
Accept
LAN
DMZ
Accept
DMZ
Com.X
Accept
DMZ
Internet
Accept
DMZ
LAN
Drop
Internet
All Zones
Drop
All Zones
All Zones
Reject
Table 3: Com.X Firewall default policies.
3.3.2.2 Managing Policies

Policies are evoked in preferential order based on their position in the list on the GUI.
By right-clicking on a policy, it can be moved up and down, deleted and edited.
Figure 49: Policy Management and preference manipulation.
3.3.2.3 Editing Policies

To edit a policy right-click the policy and select edit. This edit does not allow the
changing of source and destination zones (for that a new policy is required) but
allows the resultant action to be changed (choosing between Accept, Reject and
Drop) as well as select the type of logging required for traffic evoking this policy.
Further, a policy can be disabled, and a disabled policy re-enabled.

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Figure 50: Editing an existing Firewall policy.
Figure 51: Logging Options for firewall policies.
3.3.2.4 Creating a New Policy

To create a new firewall policy, select Options, then new policy.
Figure 52: Options

for new rules and
policies.

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Figure 53: Configuring a new firewall policy.
3.3.3. Rules
Rules are very specific exceptions to the configured policies. Rules allow the
forwarding of ports. Rules allow for traffic to move from particular ports or addresses
in a manner that would normally be prevented by the policies.
To configure a new rule, select Rule Wizard from the options menu of the Firewall
page and select the type of rule you wish to configure.
Figure 54: Rule Wizard . Types of

configurable rules.
Each Rule type makes use of the same rules parameter sheet, with certain fields preconfigured depending on the type of rule.
3.3.3.1 Port Forward
To forward a single port or a range of ports to the firewall, you can use the port
forwarding rule wizard.

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Illustration 1: Port Forward rule wizard

options.
Field
Description
Source Zone
Since this is the port forward rule wizard, the source

zone is automatically configured to Internet and can
not be edited.
Source IP address
If required, access to the port forward rule can be

restricted to a particular source IP address or address
range. To configure an IP range, specify the non-zero
bits of the netmask numerically after a / character.
<ip_address>/<numerical_netmask>
Source start port
The start port of the source which you would like to be

able to access this port forward rule.
Source end point
The end port of the source which you would like to be

able to access this port forward rule.
Destination Zone
The allocated zone of your destination device. For port

forward rules this option is limited to LAN and DMZ.
Destination IP address
The ip address of the destination device to be accessed

by this port forward rule.
Destination start port
The start port to be accessed on the destination device.
Destination end port
The final port in the range to be accessed on the

destination device.
Protocol
The communication protocol that will be available on

this rule. Select between tcp and udp.
Loglevel
The logging tag you would like attatched to any logging

on this route.

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Enable
An Enable toggle. If unchecked, the rule will not be

active.
Description
An optional description for the firewall rule.
3.3.3.2 Inbound Redirect
Figure 55: Inbound Redirect rule wizard.
Field
Description
Source Zone
In a redirect rule, the source will always be in the

internet zone, so this field has been preconfigured and
can not be edited using the rule wizard.
Source IP address
Access to this rule can be limited to a single IP address

or a range of IP addresses on the same subnet.
Source start port

able to access this rule.
Source end point
The final port of the range of ports on the source which

you would like to be able to access this rule
Destination Zone
The allocated zone of your destination device. As this is

an inbound redirect rule, the zone is pre-configured to
Com.X and can not be editted from the rule wizard.
Destination start port
The port to be accessed on the destination device.
Protocol

Loglevel
The logging tag you would like attached to any logging

on this route.

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Enable

active.
Description
3.3.3.3 Inbound Access

An inbound access route can be configured to allow access to the Com.X device
from particular devices on the external internet.
Figure 56: Inbound access rule wizard.
Field
Description
Source Zone
In a redirect rule, the source will always be in the

Source IP address

Source start port

able to access this rule.
Source end point
The final port of the range of ports on the source which

you would like to be able to access this rule
Destination Zone

an inbound access rule, the zone is pre-configured to
Com.X and can not be edited from the rule wizard.
Protocol

Loglevel


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on this route.
Enable

active.
Description
3.3.3.4 Ping
A ping rule allows the Com.X device to receive and respond to pings sent from
particular devices in the internet zone (i.e. not on the local network, but rather on the
external internet).
Figure 57: Ping firewall rule wizard.
Field
Description
Source Zone
In a ping rule, the source will always be in the internet

zone, so this field has been preconfigured and can not
be edited using the rule wizard.
Source IP address

Destination Zone

an inbound access rule, the zone is pre-configured to
Com.X and can not be edited from the rule wizard.
Protocol
As this is a wizard for a ping rule, only icmp traffic is

permitted. This field can not be edited on this type of
rule.
Loglevel

on this route.
Enable

active.
Description


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3.3.3.5 Blacklist
If required, particular ip addresses or ranges of ip addresses can be black listed by
the administrator. This results in any traffic from the source being rejected.
Figure 58: Blacklist rule configuration wizard.
Field
Description
Source Zone
In a blacklist rule, the source will always be in the

Source IP address
The IP address (addresses) you would like to blacklist.
Destination Zone
All zones are included in a blacklist rule
Loglevel

on this route.
Enable

active.
Description
3.3.4. Advanced
Should you require some behaviour that can not be achieved with the rule-types
already listed, configuring an advanced rule allows you to manually configure each
field of the rules sheet. This allows for advanced port-by-port traffic routing of your
choice of protocols. We suggest consulting
the shorewall documentation before configuring a rule of this type.
http://www.shorewall.net/
3.3.5. Failed Rules

When using the advanced rule wizard, it is possible to configure a rule that is for
some reason forbidden. This could be a result of a logical flaw, or an unknown
interface.

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If such a rule is present, the checking firewall services step will fail in an apply, but
the apply will succeed. The GUI elects to disable the individual rule, but enable the
firewall service and all other configured rules or policies.
Rules that have been disabled due to logical flaws or ambiguities are highlighted in
red and displayed in the rules list.
Figure 59: This rule features an unknown interface and so has been disabled and
marked in red.
3.3.6. Network Quality of Service

Network Quality of Service provides a mechanism to differentiate between and
classify various network traffic sources. Each traffic class can be assigned a priority
to allowing preferential queueing and transport treatment of particular streams of
traffic to meet specific needs. By default, three special classes of traffic are
supported: SIP signalling, SIP audio and SIP video. Other, unclassified traffic is left
as is by Com.X.
The special traffic classes support the setting of Layer 3 class of service (DSCP) and
Layer 2 class of service (CoS Priority) for traffic generated by the Com.X. DSCP is
generally visible right through the network path, whilst CoS Priority is only used for
the local network queuing priority and on VLAN if configured.
Figure 60: Network traffic quality of service list
3.3.6.1 Loading network QoS defaults

To load default classification and priority settings, select options, Load Defaults.
Figure 61: Load default QoS

settings.
Accept that all user changes will be over written, and Review/Apply your
configuration.

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3.3.6.2 Editing classification and priority
To edit the classification and priority of a particular type of traffic, right click the entry
in the classes list and select edit.
Figure 62: Editting network traffic classes
Figure 63: QoS Class configuration options
Field
Contents
Class Name
sip_sig, sip_audio or sip_video
Class Description
Description of above
DSCP Value
Differentiated Services Code Point as per RFC 5865.

Unclassified traffic has DF code point.
802.1p priority
VLAN priority as per IEEE 802.1Q, also internal network queue

priority. P0 is normal priority (unclassified traffic), P7 is highest.
3.4 Hardware Device configuration

To list, detect and configure hardware devices (including the Com.X, external iTAs,
IP phones and other network devices), select the Start button, then Configuration and
open the Hardware panel.
3.4.1. Configured devices

The list of configured devices is displayed on the hardware panel. Devices are
classified as follows:

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Generic devices the IP address and/or MAC address of the device is

known, and the Com.X GUI provides proxy web access to the device. This
class supports generic IP phones, ADSL routers, etc.
Comma devices built in telephony hardware and Comma iTA devices
Managed SIP devices supported IP phone types that are fully managed by
the Com.X
For each device, the device name, type, model, MAC address and Auto IP values are
listed. Where available, the device IP address and current status are also listed.
3.4.2. Detecting and scanning for devices

The Com.X unit can detect network devices present on all networks attached to it. By
default, devices requesting either DHCP IP information, internal telephony devices
and external Comma iTA devices are automatically added to the list of detected
devices. A scan of all network interfaces can also be initiated by selecting the Scan
button. This process may take a couple of minutes.
Figure 64: The network can bge scanned for devices from the Hardware panel.
The number of devices detected is displayed in the top right corner of the Hardware
panel. To add detected devices to the configuration, select the Add menu item
followed by the Detected menu item.
Figure 65: The number of detected devices is displayed on the Hardware panel.
Scanning the network increases the number of detected devices.
This results in a list of detected devices from which one can be selected for
configuration. Where available, the existing MAC address, device type, model, IP
address and vendor information is listed.
3.4.3. Adding Detected Devices

Once a device has been detected by the Com.X, navigate to Add then Detected
and select the device from the list. If the network was scanned to detect the handset,
then the list of detected devices will include several devices and the correct handset
must be selected. Select the appropriate handset and click OK.

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Figure 66: Selecting the appropriate detected device

after a scan of the network.
3.4.4. Manually adding devices

If a device is not present on the network or does not appear on the detected list, it
can be added manually by providing its MAC address and IP information. Comma
iTAs, Generic devices and Managed SIP devices can be added by selecting the Add
button, followed by the New menu and the desired device type.
Figure 67: Selection of device types to be added.
3.4.5. Configuring devices

After a device has been selected for configuration, manually added or edited, a
device configuration form will appear:
3.4.5.1 Configuring Comma devices
Double-click an ethernet interface to configure the following fields:
Field
Content
Device Name
A custom name for the device
Model Type
The card configuration of the device. If the device was detected,

this field will be populated automatically.
Auto IP
On first boot, request an IP address from the Com.X Manager.

Once an IP has been allocated, the Auto IP field becomes
disabled and the IP is statically reserved for the device by the
Comma Manager, always issuing the same IP to the device

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Field
Content
matching the configured MAC address.
IP Address
An IP address unique to the device on the network. Autoassigned if Auto IP Client is enabled.
Network Mask
The network mask the device should be configured with. Autoassigned if Auto IP is enabled.
Gateway
The routing gateway the device should use. Auto-assigned if

Auto IP is enabled.
Attached network The network interface to which the device is attached.

Note: the interface selected here must be configured as
providing Comma services, or the addition of the device will fail
with the message 'Device not supported on this interface'. The
Comma device manager only listens on and provides
configuration to devices on boot-up on interfaces configured for
Comma services.
Timing Mode
The timing source for the device. Options are:

Local: Recover timing from the local Com.X clock (i.e. this
device is a clock master)
System: Recover timing from another clock master (i.e. this
device is a clock slave) This is equivalent to the iTA
Network clocking mode.
Bri/Pri Priority 1: Configure this device to recover clock from
the first active ISDN interface (PRI or BRI)
Bri/Pri Priority 2: Configure this device to recover clock from
the second active ISDN interface (PRI or BRI)
Note: On Com.X5 systems, the primary clock source is
always the system, unless an iTA is configured as clock
master (Line 1 or Line 2) on a PRI or BRI port. When using
an X5 , it is an invalid configuration to configure any iTA as
'local'.
Table 4: Comma device configuration

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Figure 68: Comma device configuration
3.4.5.2 Configuring Generic devices

Field
Content
Device Name
A custom name for the device
Model Type
The type of the device. In most cases this will remain blank for
manually added generic devices.
Auto IP
Indicates whether the Com.X DHCP server should allocate an IP

to the device on DHCP request. Once the DHCP lease on the IP
expires, the Com.X DHCP server might issue a different IP to the
device. To ensure a device is always allocated the same IP
(static), disable Auto IP and enter the IP manually. If the static IP
falls within the DHCP pool, the DHCP server will always allocate
it to the device (i.e. the DHCP client on the device should still be
enabled). If the static IP falls outside the DHCP server IP pool,
the device should be manually configured with the static IP and
its DHCP client disabled.
Please see section 3.2.2.3 for more information on configuring
DHCP services on an interface.
MAC
The device's unique hardware address. May be left empty if an IP

address is provided.
IP Address
An IP address unique to the device on the network. Autoassigned (leave empty) if Auto IP Client is enabled.
Network Mask
The network mask the device should be configured with. Autoassigned if Auto IP is enabled.
Table 5: Generic device configurationSip Handset Provisioning


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3.5 Managed Sip phone provisioning

SIP handsets can be configured and manipulated in various ways directly from the
Com.X GUI.
For each phone added to the hardware list, a corresponding .cfg file, marked as
unique to the handset with a matching MAC address, is created and stored within
/etc/commagui/autoprovision/, where each phone manufacturer and then model, is
allocated its own folder containing the .cfg files of all similar handsets. Changes
made on the GUI are reflected in these files, which are requested by their
corresponding handsets on start up.
A configuration backup of the system will back up these files to the backup archive.
Should an archive be restored, in which configuration files are present, they will be
restored to the correct folder within /etc/commagui/autoprovision/ after the restore
has been applied. Restoring the factory default .tar.gz archive on the Com.X unit will
result in these files being deleted and the Com.X losing all association to the phone.
Each handset will have to be re-added to the hardware panel to allow configuration of
any of the settings mentioned.
Note: If the phone connects to a network with a DHCP server other than the Com.X
present, the DHCP server should configure option 66 to refer to the Com.X when
responding to phone DHCP requests.
Much like a firewall is able to block unwanted traffic by forwarding selected ports on
the Com.X to be visible from the internet, the Com.X is able to port-forward traffic to
the handsets so that they sit behind the Com.X, but can still be safely accessible
from the internet. This allows for the installation of unconfigured phones on site, to be
configured remotely at a later stage.
Figure 69: The Com.X forwards and proxies the necessary

traffic to the handsets so that they can sit safely behind a
firewall and still be configured remotely.
This means that an array of phones can sit behind a firewall, and be configured
remotely.

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3.5.1. Adding Hardware (Managed SIP Device)

In order to configure a sip handset, it needs to be added as a hardware device on the
system.Once a phone is added as a hardware device, the ports of the handset
become visible on the CommaGUI ports tree and are available for configuration.
Note: If a phone is featured on the hardware list, either as a managed SIP phone, or
a generic device, the GUI of the phone can be accessed through the GUI of the
Com.X. This is done by forwarding ports, ensuring that all hardware is available
remotely, even if it is behind a firewall.
3.5.1.1 Manually add Managed SIP Phone

It is possible to manually add a device, without the device even being present on the
network, however the MAC address of the device must be known. This allows for
configuration of units prior to installation, or for the autoprovisioning of handsets that
are receiving DHCP ip addresses from remote servers.
Figure 70: Adding Hardware.
Select Options New and then select either the manufacturer of your handset.
Figure 71: Configuration options for a Managed SIP

Phone.

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Note: If a Managed SIP Phone is receiving DHCP from a remote server (i.e. Not
from the Com.X), that dhcp server should be configured to set option 66 to contain
the location /etc/commagui/autoprovision on the Com.X server.
Field
Description
Device Name
Choose a suitable name for the device.
Model Type
Use the drop down list to select the appropriate model
Auto IP
On first boot, request an IP address from the Com.X

Manager. Once an IP has been allocated, the Auto IP
field becomes disabled and the IP is statically reserved
for the device by the Comma Manager, always issuing
the same IP to the device matching the configured MAC
address.
Mac Address
The Mac address of the device being added. This field is used
to identify the device when it is physically connected to the
network.
IP Address
An IP address unique to the device on the network. Autoassigned if Auto IP Client is enabled.
Network Mask
The network mask the device should be configured with.

Auto-assigned if Auto IP is enabled.
Attached Network
The network interface through which the device will connect

to the Com.X.
Configuration Path
The location on the com.x at which the configuration is stored.

This read only variable can be copied from the GUI if it is
required elsewhere.
Autoprovision
Firmware
If a .rom firmware file has been added to the Com.X via the
file manager, selecting the .rom will install the selected
firmware on the handset.
3.5.1.2 Adding an Auto-detected Managed Sip phone

If a compatible phone is receiving an IP from a DHCP server on one of the ethernet
ports on the Com.X, it should identify themselves as Managed Sip phones. To scan
for a list of devices obtaining IP addresses from the Com.X, select scan and enable
the auto detect option.

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Figure 72: Enable Auto Detect to see a

list of devices receiving an IP address
from the Com.X.
You will notice the number of detected devices (displayed in the top right hand corner
of the GUI) increase.
Figure 73: Detected

Devices increases.
Select Options, and then Detected to see a list of detected phones. Select a
Managed SIP Device.
When configuring an auto-detected device the Com.X automatically detects and
populates all fields with appropriate settings. The name of the device is defaulted to
the Mac address of the unit, and this is the only field that should require (optional)
editing.
Figure 74: Adding Managed devices.

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Figure 75: Auto-detected Managed SIP Device configuration

options.
Field
Description
Device Name
Choose a suitable name for the device. By default, this field is

set to contain the MAC address or model number of the
device.
Model Type
On a detected Managed SIP phone, the device should

automatically populate this field with its model type.
Auto IP
This option is enabled by default for a detected device.
Mac Address
This field contains the MAC address of the device, and is

populated automatically for a detected device.
IP Address
An auto-detected device will have received its IP address

from a DHCP server on the Com.X, so this field will
automatically be populated by the GUI.
Network Mask
The network mask of an auto-detected device will be

automatically configured.
Attached Network
This field will be automatically set to the interface serving

DHCP to the auto-detected phone.
Configuration Path
The location at which the configuration file for autoprovision is

stored. This read-only variable is available to be copied and
pasted from the GUI.
Autoprovision
If any compatible .rom firmware files have been uploaded to


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Firmware
the Com.X, they will be available as choices here. Selecting

any .rom file in this field will install the selected firmware on
the device.
Accept and Review/Apply your configuration changes.

3.5.1.3 Provisioning via Spreadsheet based configuration
For larger installations, featuring several sip endpoints,spread sheet provisioning is a
very powerfull tool. Various fields are configurable on the spreadsheet and will be
automatically added to the configuration of the phones, once the Com.X detects
them.
3.5.1.3.1 Adding hardware via spreadsheet
To add hardware via spreadsheet, first prepare a spreadsheet containing your
handsets, extension numbers and settings (voicemail, DNS, NAT etc).
Navigate to Options and select Spreadsheet the import. Browse for your spreadsheet
file and select ok.
A sample spreadsheet for you to work from is available at:
http://www.farsouthnet.com/wp-content/uploads/2013/08/Com.X-IP-phone-configsample.xls
Note: The fields listed in the spreadhseet can vary and not all options need be
configured. It is recommended that anyone using spreadhseets to configure phones
first manually add a few managed sip phones to a unit and then export a
spreadhseet of the configuration, which can be used as a template for all future
configurations. See Section 3.5.1.3.2 for details on exporting a spreadsheet.
Figure 76: Options to import or

export hardware spreadsheet files.

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Figure 77: Browse for the appropriate spreadsheet file.
Note: Provisioning from a spreadsheet will configure the extension automatically.

Spreadsheets should not contain extension numbers for any existing extensions. If
an existing extension number is present on the spreadsheet, it will be deleted and
replaced by a new extension of the same number, featruing the configurations of the
spreadsheet entry.
Note: If a handset in the spreadsheet already exists on the Com.X (identified by

MAC address) the handset will be updated with the settings from the spreadsheet.
3.5.1.3.2 Exporting spreadsheet of current hardware and extensions

The GUI can produce a spreadsheet export of your current hardware and extension
configuration. This export can be useful as a template, where only mac addresses
and other optional changes need to be made from system to system.
To export your hardware and extensions select Options, Spreadsheet and then
Export.
3.5.2. Adding a Managed SIP phone from a VLAN interface

Note: At this point only compatible Yealink handsets are supported for autoprovision
on VLAN interfaces. Other handsets can be manually configured to receive DHCP
on the VLAN and then manually configured to register as an extension.
For a variety of security and performance reasons, it may be desireable to host SIP
handsets on a virtual LAN.
In order to configure a managed SIP device to reside on a VLAN, a VLAN interface
needs be configured as in Section 3.2.5.1.
In order for the DHCP server on the VLAN interface to provide DHCP to the handset,
the phone needs first to be configured to connect to the VLAN. This can be easily
achieved by configuring DHCP server on both the physical interface and the VLAN,
as per Section 3.2.2.3.
Connect your handset to the physical LAN port (most likely through a switch) and
power up the phone.
Ensure that auto-detection of hardware is enabled, and add the handset from
deteced devices, as in Section 3.4.3.

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On booting, the phone will initially receive a DHCP adddress from the DHCP server
on the physical LAN interface. Click accept, but do not apply.
Before applying the configuration of the handset, edit the associated LAN interface
and select the VLAN interface from the drop down list. Accept and Review/Apply.
Note:The drop down list is not visible when adding the handset. You will need to first
accept the handset configuration, then right-click and edit the hardware entry before
the other interfaces become available for selection
Figure 78: Select the VLAN interface as the attached network

field
The phone should then detect the changes to it's network settings and reboot itself, if
this does not happen automatically, reset the handset to factory default settings from
the GUI. On reboot, the phone will retrieve it's updated configuration.
Note: Upon the phone booting, the review/apply button on the GUI may become
bold. If this occurs, REVERT your configuration, to update the IP address field on
the managed SIP phone. Only apply from the review/apply button once the
managed SIP phone shows both the correct VLAN interface and IP address.
Figure 79: This shows one handset residing on a network on LAN2, whilst another
phone, connected to the same physical LAN port, resides on the virtual network hosted
by VLAN2.50.
Note: With certain handsets, the DHCP settings remain set until the interface goes
down, so it may be necessary to physically remove and replace the network cable,
or induce a power break cycle before the phone reconfigures itself to the VLAN.

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If it is not feasible to enable DHCP server on the physical LAN (e.g. because of
another DHCP server on the LAN), the phone must be manually pre-configured to
attach to the VLAN so that it can receive its IP address and auto provisioning
information from the DHCP server on the VLAN.
3.5.3. Removing a hardware device

To remove a device that has been added as hardware, simply right-click the device in
the hardware list, and select remove. Once you apply the device and its ports will be
removed from the GUI. Any extensions configured on the ports become free
extensions, but remain in any inbound routes, ring groups or queues.
3.5.4. Restore Factory Defaults of a Managed handset

To restore factory default configuration of a managed phone, right-click on the phone
in the device list and select the Factory Defaults option. This will delete any nondefault configuration on the phone.
If you have the phone any extensions configured on the phones ports, the phone will
receive this configuration when it starts up. To have the phone remain in a factory
default configuration, first delete or unassign extensions on the phones ports and
then factory default the phone.
Note: It is recommended that any phones that are added be factory defaulted as a
part of their initial configuration. It is not possible to reassign lines on devices if they
are already assigned to extensions, the association to the extension must first be
deleted. Factory defaulting phones as you install them deletes any extensions
configured in a previous installation and ensures that the Com.X is able to configure
all lines seamlessly.
Note: Due to changes in more recent Yealink firmware, It is not possible to factory
default of restart Yealink handsets featuring firmware of version 72.25 and more
recent. For these handsets you can connect to the phone's GUI through the Com.X
GUI and manually restart or default the handset.
3.5.5. Configuring Extensions on a Managed Sip phone

By default, Line1 of any Managed Sip Phone will be enabled. Any further lines will
first have to be enabled before they can be configured to support and extension.
3.5.5.1 Enabling Lines
To enable a line on a Managed Sip Device. Select the device in the hardware list.
Find the desired line from the ports tree on the right hand side of the GUI. Right-click
the line and tick the Enabled box.

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Figure 80: Enable the line on the

device.
3.5.5.2 Assigning Extensions to Managed SIP Devices

Extensions can be assigned to the enabled lines of Managed SIP Devices using one
of two methods:
3.5.5.3 Creating New Extensions
Extensions that have yet to be created can be created and assigned to the Managed
SIP Device simultaneously.
To create new extensions that are assigned to the Managed SIP Device, navigate to
the Extensions page of the GUI and right-click the particular line on the device in the
ports list. Then select Assign.
Figure 81: Right click the appropriate line in the

ports tree to assign it a new extension.
Multiple extensions, with consecutive DID numbers can be assigned to multiple lines
on the device by holding 'Ctrl' while selecting individual lines, then right-clicking and
selecting assign.
Figure 82: Consecutive extensions can be created and assigned to multiple

lines on the same device.
Enter the DID of the starting extension, this extension will be assigned to the first
selected line on the handset. The other selected lines on the device will be assigned
extensions with consecutive numbers.

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Figure 83: Four extensions will be created

and assigned to the lines of the same
handset.
3.5.5.4 Assigning free extensions to available lines on Managed SIP Device

A free extension configured on the system can be assigned to an available line on a
SIP handset. Left-click the appropriate line in the port tree of the Extensions page of
the Com.X GUI and (while holding left click) drag the line over the entry of the
desired free extension and then release.
A green tick icon indicates that the selected extension is available for assigning,
whilst a red cross icon indicates that an extension is not available to be assigned.
Figure 84: Click and drag the desired line over the desired free extension. Note the
green tick icon on the line being dragged.
3.5.5.5 Configuring a port on the Phone to register as an extension on a

remote system
If there is an unused port(line) on the handset, you can configure it to register as an
extension on a remote PBX, from the Com.X GUI.
Navigate to the hardware page and select the port from the port tree. Right-click and
select Edit.
Figure 85: Editing a line on a Managed Sip

Phone.

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FIELD
DESCRIPTION
Enable
If not ticked, the line will not be available for configuration.
Local Extension If enabled, SIP Server, Register Login and Register Password
Port
fields will be automatically populated. Disable this option to
manually enter configuration details and register an extension on a
remote system.
Label on Phone Enter the label you would like to see on the phones display.
Register Login
If Local Extension Port is disabled, this field can be edited to

contain the authorisation name for an extension on a remote
system.
Register
Password

contain the password for an extension registering on a remote
system.
SIP Server

contain the IP address of the remote SIP server on which the
extension is registering.
SIP Server Port If Local Extension Port is disabled, this field can be edited to
contain the port on which the extension should register on the
remote system.
Use Outbound
Proxy
If enabled, an outbound proxy can be configured through which to

register the extension.
Outbound proxy The address of the outbound proxy through which the extension
registers with the remote system.
Outbound
Proxy port
The port of the outbound proxy through which the extension

registers on the remote system.
NAT Traversal
If enabled, the phone will use NAT to locate its remote host.
Note: In order for a line on a managed Sip phone to register as an extension on a

remote system, correct routing needs to be configured on both systems. This would
require a route to allow the phone to connect to the external network through the
correct interface, and a corresponding route on the remote system, directing traffic
to the phone to the correct interface.
3.5.6. Codecs
By default extensions created in the above manner, will be configured to support an
alaw codec. To add or remove supported codecs to the extension, right-click the
extension and select Edit. Select the Codecs tab. This will present a list of
available Codecs. To select multiple Codecs, hold Ctrl and click on the desired
Codecs.

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Figure 86: Extensions can be configured to support multiple

Codecs.
3.5.7. Removing an association to an extension

Two possibilities exist for removing extensions from device lines.
3.5.7.1 Unassigning Extensions
To unassign an extension, right-click on the extension in the extensions list and
select Unassign. This will result in the phones line becoming a free port and the
extension becoming a free extension. The extension will remain in any inbound
routes,destinations (queues and ring groups) and flex-path contexts and can be
logged in to from any free port by dialling the *11 feature code (See section 3.7.5 )
3.5.7.2 Deleting Extensions
To delete an extension, right-click on the extension in the extensions list and select
Delete. This will result in the line on the phone becoming a free port, and the
extension being deleted. The extension will be removed from any inbound routes,
queues, ring groups, flexpaths and other internal destinations.
Illustration 2: Deleting or Unassigning an

extension will result in a free port on the phone.
3.5.8. Advanced Configuration options for Managed SIP Devices

The advanced configuration options can be configured by right-clicking the Managed
SIP Device entry on the hardware panel and selecting 'Advanced'.

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3.5.8.1 Configuring Soft Keys
The Com.X GUI offers a mechanism to provision the soft keys keys of compatible
handsets. These keys can be set to transfer to, and monitor the status (through BLF)
of : other extensions, conference extensions, queues, ring groups, parking lots and
particular parking bays within parking lots.
3.5.8.2 Configuring keys individually
Right click the device and navigate to Advanced ->Switchboard.
Using this interface, the keys of the handset, as well as any extension modules to
which it is connected,can be configured directly from the GUI.
Figure 87: Available destinations for soft keys include

extensions,conferences,queues,ring groups, parking
lots and individual parking bays.
Right-click the particular key you wish to edit, and select a destination.
If the user of the managed SIP Device is on a call, pushing the key will transfer the
call to that destination. If no call is in progress, pressing the key will initiate a call to
the destination.
By default, the DSS key will also show the status of the destination through the Busy
Lamp Field. If the destination is available, the lamp next to the DSS key will be green.
If the destination is busy, the lamp will be red and if the extension is ringing, the light
will flash red.
3.5.8.3 Configuring Multiple DSS keys to map to consecutive extensions.
It is possible to simultaneously configure multiple keys to map to extensions. Rightclick the Managed SIP Device in the hardware list. Navigate to Advanced>Switchboard and then select multiple soft keys by left clicking the first key you
would like to edit and then holding shift while left clicking the last key in the range. To
select non-consecutive keys, hold Ctrl while selecting the desired keys.

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Figure 88: Multiple DSS keys can be configured

to transfer to consecutive extensions
Select the Extension option and choose the first extension in the range. The first
highlighted key will then be set to the first extension selected and the subsequent
keys set to the next extensions featured in the list.
3.5.8.4 Adding an Extension Module
To add an extension module, simply click Add. Navigate to the correct EXT tab to
configure the soft keys of the extension module. Multiple extension modules can be
added to handsets that support them.
The EXT keys are configured as per section 3.5.8.1.
Note: Applying any changes to Managed SIP Devices from the Com.X GUI will
cause the handset to reboot. The device will reboot,and then update configuration
upon startup. This process results in the phone being unavailable for around one
minute after a Review/Apply.
3.5.9. Supported Devices

At present Yealink, Snom and Polycom handsets are supported for configuration as
Managed SIP Phones.
Yealink: The entire Yealink IP Phone range is supported
Snom: The entire Snom Efficient Line and Advanced Line range of devices is
supported.
Polycom: The Entire VVX range apart from the VVX1500, as well as the IP
soundpoint range of handsets is supprted for autoprovision.
Note: If you are uncertain whether a device is a supported as a candidate for
Managed SIP Phone configuration, please enquire with your dealer/distributor.

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3.6 Configuring Comma device ports

When a device is selected in the device list, the available ports on the device are
categorized and listed in the ports tree on the right of the Hardware panel. Comma
ports are enumerated using the device's name, the card index in the device and the
port number. I.e. device:a2-5 would indicate the fifth port on the second analogue
card (a analogue, d bri, p pri) in a device called 'device'.
3.6.1. Configuring Analogue Trunk (FXO) ports

To edit the configuration of an FXO port, double-click on the desired port from the
ports tree on the right of the hardware panel.
Field
Content
Input gain
Indicates how much, if any, gain adjustment should be performed

in the input direction. Values are in decibels.
Output gain

in the output direction. Values are in decibels.
Echo canceler
The echo cancellation delay and algorithm to be employed on the

port. The delay up to which the echo canceler can compensate is
specified in milli-seconds. An optional non-linear processing
echo suppression algorithm (NLP) can also be employed during
echo cancellation. If you expect the devices connected to the port
to be in a noisy environment how-ever, you might prefer not to
enable NLP, as it might result in the far-end noise being cut off
when speech ceases. Some local users may find this
uncomfortable. NLP is recommended for quiet remote
environments. Please see section 3.6.8.1 for a detailed
discussion around echo cancellation.
Detect Fax
When enabled, the Com .X will differentiate between an incoming

call and an incoming fax. Any echo cancellation configurations
will be bypassed and the jitter buffer for the call adjusted for a
higher tolerance, in the event of an incoming fax.
DID number
A direct inward dialing number to associate with this port. A call

incoming on this port may be routed using the DID.
DID description
A custom description for the DID.
Detect caller ID
If this is enabled, the incoming caller ID, if available, will be

detected and may be used in inbound routing.
Hangup
detection
Devices such as Least Cost Routers connected to FXO ports, or

the Telco equipment in various parts of various countries may
signal hangup differently. Change this setting to match the tone
set for the country of device deployment, or the duty cycle of the
hangup signal you are receiving. Options include 250, 500 or
2500 ms of tone followed by the same period of silence and any
regular pattern.

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Field
Content
Note: if a country tone-set is not available for the country of
deployment and as a result on of the available duty cycles is
configured, it is possible that different exchanges routing
inbound calls to the system might have different end-of-call
tones.
This may result in FXO port lockup if the varying tones cannot
be detected using the configured duty cycle. If this is the case,
use the 'Any regular pattern' setting to detect such tones.
For more information on the options available to detect end-ofcall under such conditions, please see the Com.X Troubleshooting guide.
Table 6: FXO port configuration
3.6.2. Configuring Analogue Extension (FXS) ports

Field
Content
Input gain

in the input direction. Values are in decibels.
Output gain

in the output direction. Values are in decibels.
Echo canceller

enable NLP, as it might result in the near-end noise being cut off
when speech ceases. Some remote users may find this
uncomfortable. NLP is recommended for quiet local
Allow transfer
If enabled, the phone using this port is allowed to transfer calls.
Immediate mode When enabled, ring tone is not provided when the phone goes off
hook and the Com.X is required to take immediate action, such
as dialing an emergency number. Please see section 3.12.11.7
for more detail on using FlexPath with his feature.
Fax channel
Indicates that this port is to be used for fax. Enabling this setting
disables echo cancellation on the port. Please see section 3.6.8
Table 7: FXS port configuration

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3.6.3. Configuring BRI ports

Field
Content
Link Type
Indicates whether this device is part of a point-to-point (i.e. DIDs

may be issued by the telco) or point-to-multipoint (i.e. one or
more multi-subscriber numbers may be issued, or multiple BRI
devices share the BRI bus.) The link type needs to match the
mode of the service provided by the telco.
Act as NT
When disabled, the device is an ISDN TE, connected to the telco

NT. When enabled, this port acts as though it is an ISDN NT
device. This is useful when the Comma device is placed between
the PBX and the telco NT, e.g. for least-cost routing and BRI
gateway routing.
Bus termination
Comma BRI cards have terminating resistors on-board, removing

the need for external terminators. If this device should terminate
the BRI bus (this is the usual case), enable this setting.
If multiple devices are attached to the BRI cable (point-tomultipoint) and another device is providing termination, the
Com.X can be configured not to terminate.
A BRI link should always be terminated by a device on at least
one side of the BRI connection, or electrical reflection on the link
could interfere with BRI signaling and result in loss of layer 1 link
establishment, or CRC errors.
Echo canceler

enable NLP, as it might result in the far-end noise being cut off
when speech ceases. Some local users may find this
uncomfortable. NLP is recommended for quiet remote
Note: This setting applies to all BRI ports in the same BRI trunk
group as this BRI port. This option is not available if the port is
not part of at least one BRI trunk group.
Detect Fax
When enabled, all BRI ports in all trunk groups to which this BRI
port belongs will detect fax tones and disable echo cancellation
for the duration of a fax call. We recommend enabling this feature
on all BRI ports that are likely to support fax calls.
Note: This option is not available if the port is not part of at least
one BRI trunk group.
Table 8: BRI port configuration


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3.6.4. Configuring PRI

Field
Content
Physical interface The service offering, either 23B+D T1 or 30B+D E1. For E1,
double frame or CRC4 multi-frame can be enabled, and for T1
framing can be either Super Frame (D4) or Extended Super
Frame (ESF, using CRC-6) The physical interface configuration
must match that of the PRI peer.
Switch type
National switch type options are:

US National ISDN phase 1 (NI-1)
US National ISDN phase 2 (NI-2)
ETSI Euro ISDN: Europe ISDN standard
QSIG: protocol for switching between PBXs
Signaling role
When configured as Terminal, the device is an ISDN TE,

connected to the telco NT. When configured as Network, this
port acts as though it is an ISDN NT device. This is useful when
the Comma device is placed between the PBX and the telco NT,
e.g. for least-cost routing.
Overlap dialing
Enabling this setting will result in digits being transmitted across

the PRI link as they are dialed, as opposed to the full number
being collected before dialing. Overlap dialing allows a
destination number to be specified partially resulting in the first
stage of a routing path being established before the remaining
digits arrive.
Table 9: PRI port configuration
3.6.5. Removing devices

To remove a device configuration, select Remove from the device's context menu.
Note: Any of the device's ports that are included in trunk groups will be removed
from the groups, outbound routes that make use of the ports will no longer be
associated with the ports and any extensions associated with device ports or
channels will become unassigned.
3.6.6. Echo cancellation

When two phones, B and C are in a call, and the person's speech at phone C is
reproduced at phone B, some of the audio exiting B's earpiece re-enters B's
microphone and is returned as echo. Analog lines exacerbate this problem due to
additional reflection of audio in the electronic circuitry. The result of audio feedback at
B is that person C hears his/her own voice after a short delay i.e. echo occurs.
The person hearing the echo should look to the remote party for the cause of the
echo and not to their own phone and system. If both parties experience echo, both
systems should be investigated.

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In analog phones, both the 2-wire to 4-wire converter (hybrid) and the handset can
introduce echo. The Com.X impedance parameters are aligned with South African
telecommunications standards. Non-compliant handsets with a large impedance
mismatch are likely to introduce noticeable echo. The delay introduced by VoIP
based systems can result in the returned echo becoming noticeable to the person
who is speaking.
The degree of echo and the delay between the original audio and the echo is a
function of the handset, gain settings, potential wiring problems and the audio
transmission path.
The Com.X products incorporate a sophisticated Echo Canceller for each telephony
channel (BRI, PRI or analog). The algorithm adapts to the detected audio feedback
within a configurable timeframe of 16, 32 or 64 milliseconds, and attempts to create
an identical image of the echo, which is then removed from the audio signal prior to
sending it on to the other party. Generally, a shorter timeframe is preferable since it
reduces the time the algorithm takes to learn the feedback environment. Typically,
echo is reduced to below detectable levels within 1 2 seconds. In extreme cases
convergence might take 3 12 seconds, during which call quality would increase to
an optimum.
After primary echo has been canceled in this way, some residual echo may remain
(echo present beyond the time-frame the canceler buffered or non-linear
reflections). Such echo is then removed by a non-linear echo cancellation algorithm,
if enabled. This may result in one of the parties' audio being clipped when both
parties are talking. Some users may find this uncomfortable. If this is the case it
might be preferable to disable the NLP.
Although the default (16ms + NLP) is normally a good compromise, the Echo
Cancellation settings may need to be adjusted on a per-port basis, as the audio
environment might differ from one port to the next (e.g. an office environment for one
range of extensions vs a factory floor for another and an open plan arrangement for
another range of extensions.)
Wiring problems, such as power cables crossing telephone cables, or insufficient
earthing of the Com.X or iTAs may add noise to the audio path that may reduce
performance of the algorithm.
In diagnosing echo cancellation problems, establish good earth (~0.5 ohm or less, if
resistance approaches 1 ohm earthing might be a concern), and proceed to eliminate
each of the potential contributing elements from the equation by testing with different
phones, connecting phones directly to the Com.X FXS ports, turning echo
cancellation off completely to establish a quality baseline and then trying different
combinations of echo cancellation, with and without NLP, on the different extensions.
Often the first measure to attempt, often yielding the greatest improvement is to try
adjusting the gain on problematic extensions.
Note: In exceptional circumstances, if the echo cancelers take a long time to
converge (e.g. > 10 seconds), the acoustics of the environment might be introducing
too much echo, long loops may be present in the telco network and / or the handset
might be too sensitive.
Adjusting the gain on the port down might help, or fixing the FIR coefficients in the
echo canceler for that port might help. After restart, the first call on ports configured
thus would take a long time to converge, and there-after the coefficients are re-used
for subsequent calls. Due to the dynamic nature of echo, fixing the coefficients is not

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recommended for most cases.
To fix the coefficients select the 'Keep' option on the FXO/FXS port.
Following this, restart the iTA for the new settings to take effect with:
comma-ls -r
Please note that echo heard by users of the Com.X (as opposed to external parties)
is typically caused by the remote party. The Com.X does attempt to cancel the echo
from the remote party on BRI, PRI and FXO connections to the telco, but the
effectiveness of this cancellation is highly dependent on external conditions which
are impossible to predict or control.
Note also that SIP connections (including local SIP phones) are never echo
cancelled (it is not technically feasible) therfore, this is the responsibility of the SIP
phone itself or of the upstream service provider.
Figure 89: Echo cancellation
3.6.7. Earthing the Com.X

Proper earth with < 0.5 ohm measured from the Com.X stud to the earthing point,
preferably to telco or failing that, building earth should be present on all installations.
The Com.X provides an earth connection in its power brick and cable. If a good
mains earth is present, a separate earth strap to the Com.X earth stud could be
unnecessary.
The Comma iTA (if present) does not have an earth connection in its power brick and
must be earthed using an earth strap and its earth stud.

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It is important that Com.X and iTA devices are earthed in a star pattern to a single
earthing point. Do not wire an iTA to a Com.X's earth stud or vice versa.
Be careful not to create earth loops which may induce noise into the system. Ensure
high quality multi-adapters (multi-plugs) are employed, and do not place the Com.X
at the end of a chain of devices chained on the same multi-adapter.
3.6.8. Configuring devices for Fax / modem support

3.6.8.1 Echo cancellation
Echo cancellation plays an important part in ensuring voice quality through the
telecommunications network.
The Com.X product range incorporates state of the art, innovative, embedded onDSP-chip echo cancellation that off-loads resources from the system server.
Echo cancellation can how-ever affect fax and modem signaling and data. It is
therefore important to understand the impact of echo cancellation on fax and modem
negotiation and configure devices correctly for fax use.
Echo cancellation is performed at the source device, where 'source' here indicates a
network endpoint that introduces a signal into the PBX system. In the Com.X
systems echo cancellation takes place on FXO, BRI and PRI ports/channels
terminating traffic from the PSTN for routing in the system, as well as FXS ports
originating calls via the PBX.
If not configured properly for fax support, faxes and modems may fail or higher baud
rate operation might not be possible.
At these sources, if a fax start CNG tone is detected and the port / channels are
correctly configured for fax, the echo cancelers are automatically disabled, and a
clear channel is offered for fax service.
In the case of PRI and BRI channels, channels may serve both voice and fax traffic,
and detection of the fax tone is required before echo cancellation can be switched
off. This detection is configurable from the Hardware panel for BRI trunk groups and
PRI ports.
Since fax devices are usually affixed to individual ports, and specific incoming DIDs
reserved for fax use, in the case of FXS and FXO, ports need to be deliberately
reserved for fax use by disabling echo cancellation. This is done for FXO ports by
enabling the Detect Fax option, and for FXS ports by enabling the Fax channel
option in the hardware panel, which disables echo cancellation and configures
additional jitter buffering for the port.
3.6.8.2 Telco issues
Telco FXO connectivity, cable distance and environment, etc. introduces additional
signal to noise ratio and impedance elements into the equation and may not support
high baud rates.
3.6.8.3 Timing
Timing in the system is very important to ensure successful fax / modem
communication. The iTA device (e.g. the internal BRI / PRI enabled iTA MPC card in
the Com.X1 case or the BRI / PRI iTA in the Com.X2 case) connected to the telco
interface should have the highest timing priority (line N), and other iTAs connected
should derive timing from this device (system). Please see section 3.4.5.1 for detail
on configuring the timing source on an iTA device.

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In scenarios where multiple PRI links are configured to the same telco, setting the
first PRI as PRI Line 1 is recommended (i.e. deriving clocking from the telco line) and
setting all other PRIs to System (i.e. sharing the clocking of the first PRI in the
transmit direction and honoring the telco's clocking on each PRI in the receive
direction.
In gateway scenarios, where one or more PRIs face the telco and one or more PRIs
face a 3rd party device, the telco-facing PRIs should be configured as above, and the
PRIs facing the 3rd party device should be configured to use System timing. In this
scenario the transmit clocking setting should be enabled the relevant PRI ports in
the GUI hardware panel.
This indicates to the Com.X that the receive direction clocking (from the 3rd party
device) should not be honored (as in the telco case), but that the Com.X should
propagate its system-derived timing to the 3rd party device. In such configurations the
3rd party device should be configured as a clock slave.
This additional configuration is not required for BRI ports, as the NT/TE mode setting
on BRI pre-selects the correct timing driver on the gateway leg facing the 3rd party
device. BRI timing configuration follows the same Line Priority and System
configuration pattern as described above for PRI.
If an iTA is connected to a switch located between the iTA and the Com.X host
server, network packet delays could occur which could also affect timing. If this is the
case, from the GUI, Monitor Telephony dahdi statistics should show slips and
skips increasing on the span to the iTA. The Com.X can be instructed to buffer in the
receive direction from the iTA in order to alleviate such network delays. This is done
by enabling the Network slip tolerant setting on the GUI under Setup Global
Settings.
Please also refer to the Com.X troubleshooting guide for more information on fax and
modem troubleshooting.
3.6.8.4 Additional configuration for faxing over PRI
In order to enable clear-channel communication for fax over PRI, the PRI needs to be
configured to allow fax detection and the disabling of echo cancellation on fax calls.
This is performed by enabling the Fax detect setting on the PRI port in the
hardware panel.
3.6.8.5 Fax over IP
Com.X systems support t.38 faxing and g711 bypass on SIP trunks. With G.711
bypass, the trunk is configured for a compressed codec (e.g. G.729) as well as
G.711 A-law. The upstream gateway re-invites on detecting the fax tone, excluding
the G.729 from the allowed codecs. The Com.X switches to the A-law codec,
allowing clear channel fax transmission.
3.6.8.5.1 G.711 voice band data (ITU-T V.152)
Com.X features that support voice band data for fax over IP are:
Automatic G.711 bypass for fax (see above)
The default configuration of Com.X SIP is constant latency (i.e. fixed jitter
buffer)
Fax detection with automatic echo canceller disable is supported on

analog/ISDN channels where echo cancellation may impair the fax signal

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The A-law codec does not use DC removal filters
Com.X VoIP does not use Comfort Noise Generation or Voice Activity
Detection
The SDP negotiation extensions (V.152 clause 7.1) are not supported in
Com.X. However, a switch to G.711 A-law on re-invite accomplishes the clear
channel requirement for voice band data, as described above.
3.7 Extension configuration

3.7.1. Extensions and Extension Ports
The Com.X treats extensions and extension ports differently.
Extensions represent dialable numbers (users) in the system, e.g. extension

1000.
Extension Ports represent hardware devices available for making and

receiving calls, e.g. FXS lines (analog phone) and SIP phone lines.
Note that a SIP phone may have more than one line and hence more than one
extension port.
Extension Ports are created when Comma or Managed SIP devices are added in the
Hardware Panel, and when Generic SIP extensions are added in the Extension
Panel.
An Extension and an Extension Port can be associated, either by configuration as
described below or by hot-desking user login.
3.7.2. Listing existing extensions and ports

To manage the list of configured extensions, select Start, followed by Configuration
and the Extensions panel.
Configured extensions are categorized as those having been assigned to extension
ports, and those that are unassigned (free extensions). Free extension ports are also
listed. These ports are available for either fixed extension assignment or for dynamic
extension login (hot-desking).
For each extension, the number, friendly name, pickup group, port assignment, port
secret, extension type (analogue or SIP), class of service and voicemail indicator are
displayed.
At the right of the extensions panel, available FXS and SIP extension ports are
shown, with current extension number assignments to these ports appended in
square brackets.
3.7.3. Adding Extensions

Extensions can be added either by selecting extension ports in the ports list and
right-clicking to assign extensions to the ports (select multiple ports for assignment
using CTRL-click), by dragging a port from the tree on the right onto a free extension,
or by using the Add button on the Extension panel to add free or generic SIP
extensions. Individual extensions, or a range of extensions can be added.

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Figure 90: Assigning extensions
3.7.3.1 Adding extensions from ports

Single extensions or a group of extensions can be assigned to available extension
ports by selecting the ports in the ports list and selecting Assign from the context
menu.
A starting extension number can be specified, and upon acceptance extensions are
numbered incrementally from the starting number for the number of extensions
selected.
Note: The end of the range is not specified, instead the number of extensions
desired is specified. If a large range is entered, the GUI will present a warning to the
user.
3.7.3.2 Adding Unassigned extensions

When a range of extensions is added, the friendly name is set equal to the extension
number.
3.7.3.3 Adding Generic SIP extensions
A Generic SIP extension allows local or remote login of any standards based SIP
phone or softphone.
When a Generic SIP extension is created, both an Extension and an Extension Port
are created with the same name, and automatically associated together. In this case,
there is no device element related to the Extension Port.
The following password generation schemes are supported for SIP extensions:
Fixed: The same password is set for all the newly created extensions. A new
strong, random password of length 12 is suggested by default.

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Same as Extension: The password for each newly created extension is the same
as the extension number.
Note: setting the extension password to the extension number is strongly
discouraged and may lead to weak system security!
Random: A strong unique, random password of length 12 is generated , for each
newly created extension.
3.7.4. Configuring extensions

3.7.4.1 Basic configuration
Field
Content
Number
The extension number selected during creation. This cannot be

changed after extension creation.
Friendly name
A custom name for the extension
Port
The device and port the extension has been assigned to, if any.
This can only be changed by unassigning and re-assigning the
extension (please see 3.7.9 and 3.7.8)
Table 10: Extension configuration

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3.7.4.2 Advanced configuration
Field
Content
Voicemail
Enable this to enable voicemail for the extension.

Note: Disabling voicemail will reset any call recording settings
that may have been configured on the extension!
Pin
The user pin used to access voicemail and features
Email
An email address to which voicemail will be emailed
Call waiting
Enables the call waiting feature, allowing an indication to be

provided to the user when an incoming call arrives while the user
is on the phone
Campon Callback If enabled, an automatic callback feature will be offered to local

users who call to this extension whilst it is unavailable,
Ring time
The number of seconds to ring before activating voicemail. This

setting is ignored if voicemail is not configured, or follow-me is
active
Pickup group
Assign this extension to a pick-up group, allowing other

members of the pick-up group to pick up this extension when it
rings using a specified feature code (*8). Specify multiple pickup
groups using , E.g. 1,2,3 In order to allow an extension to be
picked up using directed pickup (**) from any group, enter a * in
this field.
Account code
The account code to be recorded in the CDR record for calls

placed by the extension.
Note: FlexPath account code or outbound trunk CID override

configuration may supersede the value configured here.
Secret
Password used to authenticate the extension. Only visible and

configurable on SIP extensions.
Use NAT
When enabled, the system ignores the SIP and SDP headers'
address and port and replies to the sender's address and port.
Outboud CID
Optional field to configure outbound caller identification
Table 11: Advanced extension configuration
3.7.4.3 Campon Callback

The campon callback feature is an optional setting, configurable for each individual
extension. The feature affects only calls between two internal users. If an extension,
with campon callback enabled, receives a call whilst it is engaged, the caller is
presented with a keypress option for callback when available. Once the initial
destination extension has been placed back on hook, a call between the initial caller

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and target is automatically generated. First the caller's extension will ring, and then
the target extension, once the caller has picked up.
If the calling party does not press '1' to accept the callback offer, the call proceeds as
usual (e.g. to voicemail, or normal busy signal).
If the target extension has follow-me enabled, with a ringall strategy, the call will first
be passed through the follow-me list, with campon callback offered only if no
extensions in the follow-me list answer the call, and at least one of the extensions is
busy.
Should the initial target extension be unanswered rather than engaged, the campon
callback will only be offered if an extension attempted in the follow-me list is
engaged, and also has campon callback enabled. In such a scenario, the campon
callback will be attempted once the busy extension has hung up, with the resulting
call attempting to connect the intial caller and their initial target, failing over to the
now-available follow-me list if the user does not answer.
Follow-me configured with any other ring strategy will ignore follow-me on internal
calls, if campon callback is enabled.
3.7.4.4 Configuring Follow-me / Divert
Extensions can be configured to forward calls to other extensions in a follow-me list.
From the Extensions panel, right-click the extension and select 'Follow me'.
Configure the follow-me settings:

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Field
Content
Disable
When checked, the follow-me configuration is ignored and the

extension rings as usual.
Note: when follow-me is disabled here, directed pick-up of the
extension is no longer available.
Initial Ring Time
The number of seconds to ring the extension before processing

the follow-me list. If set to 0, the follow-me list is processed
immediately (e.g. divert). The extension may be in its own
follow-me list, in which case it will also ring.
Ring strategy
Determines the strategy in which extensions in the follow-me list

rings. One of:
Ring all (v2): Ring the primary extension for the initial ring time,
then rings all the extensions in the follow-me list simultaneously
until one answers
Note: In certain configurations Ring all (v2) results in an
extension ringing only once. If this is problematic, change the
ring strategy to Ring all
Ring all: Rings all the extensions in the follow-me list

simultaneously until one answers
Hunt: Rings available extensions, one at a time, in turn.
Memory Hunt: Rings the first extension in the follow-me list,
then the first and the second, then the first, second and third,
etc.
First available: Rings only the first available extension (if that
extension is engaged, but has call waiting, it will ring call
waiting) I.e. 'available' includes the concept of call waiting.
First not on phone: Rings the first available extension for which
the phone is not off-hook (i.e. ignores call waiting settings)
(strategy)-prim: This setting behaves as the selected strategy
described above, however, if the primary extension is engaged
or set to do not disturb, the extensions in the follow-me list are
not processed.
Ring time
The maximum time to ring phones. If the selected strategy is a

hunting strategy, this is the duration of each ring iteration.
The maximum value for this field is 60 seconds.
Follow-me list
The list of extensions to process according to the ring strategy.

This list may include the primary extension itself and may also
include external numbers, suffixed with a #, in which case the
number would be called using the applicable outbound route.
E.g. including 0115551234# in the follow-me list would call the
external number 0115551234

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Field
Content
Announcement
An announcement to be played to the calling party before

processing the follow-me list. Please see section 4.17 for more
detail on system recordings.
Play Music on
Hold?
If set to Ring, will play ring-tone to the calling party while

processing the follow-me list.
If set to a music on hold class (e.g. Default), will play music to
the calling party while processing the follow-me list.
CID Name Prefix
An optional CID prefix to display to the ringing extensions.
Alert info
Optional SIP alert info to enable SIP extensions in the follow-me

list to ring differently on receiving forwarded calls from the
primary extension
Confirm calls
When enabled, extensions in the follow-me list are required to

press 1 when picking up the call to confirm that the call has
been answered. This is useful for preventing voicemail on
external numbers from picking up the call when another
extension down the follow-me list could still be serviced up by a
person. This feature is only available for ring-all strategies.
Remote announce A message to be played to the person answering the call in the
case where call confirmation is required. Please see section
4.17 for more detail on system recordings.
Too-late announce A message to be played to a person answering the call in the
case where they have picked up the call, have not pressed 1 to
confirm yet, and in the interim someone else has picked up and
confirmed. Please see section 4.17 for more detail on system
recordings.
Note: It is recommended that a too-late announcement is
configured on all follow-me's. Without such an announcement
the person answering the extension too late might interpret the
engaged tone as a dropped call.
Destination if no
answer
The destination to route the call to if the ring strategy is

unsuccessful in delivering the call (i.e. no-one answers). Please
see section 3.11.4 for more detail on call destinations.
Table 12: Follow-me configuration
3.7.5. Logging into phones with an extension (hot-desking)

Users can log into ports that are categorized as free ports on the Extensions panel by
using the associated feature codes (the default is *11 to log in and *12 to log out)
followed by their extension number and pin. Calls placed from a device connected to
such a port will be flagged in CDRs as having originated from the extension number
that is logged in to the device.

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To make a port that currently has an extension assigned to it available for dynamic
login, first unassign the extension from the port, resulting in a free extension and a
free port.
Note: In order to log in dynamically to a free device using a free extension, the
extension must have a user pin configured.
3.7.6. Configuring SIP extensions

In addition to the basic and advanced configurations (please see 3.7.6), SIP
extensions also allow the configuration of various voice codecs in order to provide
compatibility with SIP providers and phones.
Codecs supported by a SIP extension can be selected by using the Codecs button.
Use Ctrl-Left-Click (holding Ctrl down) to select multiple codecs. Individual codecs
can be moved up and down the codec priority list by selecting the codec entry, and
selecting the Up or Down buttons. Codecs may specify the integration time in
milliseconds after the codec name in the form codec:integrationtime, e.g. 'g729:40'
When negotiating a SIP call, the codec with the highest priority(at the top of the list)
will be attempted first, then the second and so on.
3.7.7. Remote SIP extensions

When configuring SIP extensions as remote extensions at a remote site (e.g. home
ADSL), if the Com.X is located behind a NAT/firewall on the network, configure the
NAT/firewall to forward UDP port 5060 to the Com.X internal IP address. If required
(no NAT available), also forwards RTP ports 10000-20000 to the Com.X internal IP
address.
You may also need to configure the external address on the Com.X by selecting Start
-> Configuration -> Setup and then setting the External address. Configure this field
to the dynamic DNS name or the FQDN that resolves to the external (public) address
of the Com.X (i.e. the external address should be set to the same as will be
programmed into the remote phone).
If you are using dynamic DNS, make sure that the External Is Dynamic field is set to
true.
3.7.8. Assigning extensions

Free extensions can be assigned to a port by editing the extension, and selecting the
Port button. From there a menu of available extension ports can be accessed. Only
available ports are listed in the menus.
Alternatively, drag a free port onto a free extension in the Extensions panel.
3.7.9. Unassigning extensions

Extensions' associations with ports can be removed by selecting the extension in the
extensions list, and selecting Unassign from its context menu (right-click.)

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3.7.10. Configuring multiple extensions

Multiple extensions can be configured simultaneously. When the list of extensions for
configuration include both analogue and SIP extensions, only those fields that are not
unique to any individual extension can be configured:
When only SIP extensions are included in the list, only those fields not unique to any
of the SIP extensions can be configured. This includes the SIP codec configuration.
Note that, since multiple extensions may have different values for shared fields,
these fields default to disabled during a multiple-edit, i.e. the form is not a reflection
of the underlying configuration when first opened.
Simply accepting this form will set all the extensions' values to disabled. Canceling
the form will leave all extensions' configurations unchanged. Enabling one of the
fields will enable the field for all of the extension being edited.
3.7.11. Configuring Multiple Sip Extensions from a Spread Sheet.

Certain Sip handsets can be added to the GUI and provsioned using a spreadsheet.
This also assigns and configures the extensions as per the spreadsheet, for details
on configuring handsets and extensions from a spreadsheet, please see the Adding
hardware devicesfrom a spreadsheet technical note, available from
http://farsouthnet.com/support/technical-notes/
3.7.12. Deleting extensions

Extensions may be deleted individually, or as a group, by selecting Delete from the
extension context menu.
Note: Deleting an extension that is in use in an inbound route will result in an update
to the route to terminate with hangup.
3.8 Trunk groups and VOIP providers

3.8.1. Listing existing trunk groups and VOIP providers
To view the list of configured trunks, select Start, followed by Configuration and then
the Trunks Panel. The list is populated with legacy telco trunks in the top section and
VOIP providers in the lower section of the panel.
3.8.2. Analogue trunk groups

Analogue and BRI trunks can be grouped to facilitate simplified routing. PRI channels
are always grouped into a trunk per PRI. Each analogue and BRI trunk may only
belong to one group, containing only members of the same the trunk type (analogue
or BRI).
3.8.2.1 Adding trunk groups
A new trunk group may be created by selecting Add, then the Trunk Group option
from the Trunks page of the GUI..
A new group may also be created by selecting trunks of the same type, and selecting
New Group from the context menu.

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Field
Content
Name
Unique identifier for this group. Analogue groups (FXO) may only
be numbers from 1 62, and BRI groups may be any unique
combination of alpha-numeric characters.
Type
The group type. Only trunks of this type may be included in the
group. Allowed types are:
Analogue trunk (FXO)
ISDN BRA (BRI)
When the group was created from a selection of trunks, the type
will automatically be selected
Table 13: Trunk group configuration (addition)
After creation, the new group and any ports belonging to it will be listed under the
Trunk Groups node.
3.8.2.2 Editing trunk groups
Trunk groups can be edited by right-clicking on the trunk group in the group list and
selecting edit.
Figure 91: Trunk group configuration (editing)

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Field
Content
Enable trunk
When disabled, though part of an outbound route, the trunk

group will not be attempted for outbound or inbound calls.
Instead, outbound calls will fail-over to the next trunk or group
in the outbound route. Individual trunks in the group will still
be used if they are explicitly added to and outbound route.
Outbound caller ID
The caller ID to present for call placed via this trunk group.
Outbound dial prefix
A prefix to be prepended to all dialed numbers on all trunks in

this trunk. This can be used in conjunction with the | symbol
in an outbound route dial pattern to accomplish dialed
number transformations across all the trunks in the group.
Round robin
Place calls using the round-robin scheme for trunk selection

when placing calls using this trunk group. If not enabled,
trunk selection always starts with the first available trunk in
the trunk group.
Table 14: Trunk group configuration (editing)
3.8.2.3 Adding trunks to a group

Trunks may be added to a group either by selecting the trunks and creating a group
using them, as detailed in 3.8.2.1, or by selecting the trunks and then selecting Add
to group from the context menu and selecting an existing group of the same type.
Only groups of the same type will appear as options for selection.
After being added to a group, the trunks under the Trunk Ports node will show the
group name they belong to in parentheses.
3.8.2.4 Removing trunks from a group
To remove a selection of trunks from a trunk group, find the group under the Trunk
Groups node and select the trunks (using Ctrl-left-click to select multiple). Select the
Remove from Group option from the context menu. The trunks will be removed from
the group and the reference to the group removed from the trunks under the Trunk
Ports node.
Note: You will not be prompted to confirm this action.
3.8.2.5 Removing a trunk group
To completely remove a trunk group and remove the group association from all the
trunks included in the group, select the group in the Trunk Groups list and select the
Remove Group option from the context menu. The trunks will be removed from the
group and the reference to the group removed from the trunks under the Trunk Ports
node, and the group will be removed from under the Trunk Groups node.
Note: Deletion a group may affect routing. The group will be removed from routes
that made use of it, potentially resulting in routes with no trunks associated (empty
routes). We suggest reviewing outbound routes after any trunk configuration has
been applied.

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3.8.2.6 Viewing group members
Membership of trunks to their respective groups may be observed either by
expanding individual devices listed under the Trunk Ports node, where group
membership is denoted in parenthesis after trunks, or by selecting the desired group
in the group list under the Trunk Groups node, where the devices and trunks included
in the group are listed.
3.8.3. SIP trunks

Multiple SIP trunks can be configured for call routing. SIP trunks can also act as SIP
providers, providing the capability to configure back-to-back SIP trunks for inter-office
connections.
3.8.3.1 SIP call setup
SIP uses a peer to peer mechanism for call setup. In order for a call to be
established, two peers need to be defined and be able to communicate with each
other. This necessitates a SIP trunk entry for each peer of a given system.
Communication between two peers implies that the peer initiating the call knows the
IP address of the recipient, and that the recipient can match the caller's credentials in
order to allow and configure the call.
The peer initiating the call can either have the recipient's IP address configured, or
may know the recipient's IP address as the result of the recipient registering with the
it (i.e. the calling peer is acting as a provider.) When acting as a provider, a peer
allows registration of a remote peer by matching the incoming registration request's
username with the provider's SIP trunk name. At this point the provider's SIP trunk
moves from having an UNKNOWN peer to being associated with the registered IP
address.
An incoming call is matched by the recipient by looking at the incoming call's IP
address and FROM field, which defaults to the caller ID of the caller, or the calling
peer's from-user field if configured. The recipient first looks at the caller's FROM field,
and if this matches the username field of a peer that has Identify by user enabled or
a configured extension, a successful match is made.
If no such match can be made, the recipient matches the first SIP trunk entry that is
associated with the IP address of the caller. This implies that, if multiple SIP trunks
are associated with peers with the same IP address, they have to have Identify by
username enabled and a username specified, and the remote peers have to have
their from-user fields set to the appropriate username.
If a successful match is made and the password and codecs that are configured for
that matched peer are compatible with those in the call setup request, the call is
established.
This mechanism is used for call setup in both directions.
3.8.3.2 Adding a SIP trunk
To add a SIP provider, activate the Add menu on the Trunks panel. Select SIP
Provider.
3.8.3.3 Sip trunk configuration
SIP trunk configuration is divided into basic, advanced and codec configuration,
accessible from their accompanying tabs on the SIP Provider form.

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Field
Content
Name
The unique name of this SIP trunk. If this trunk is a SIP

provider, the name of the trunk must match the username.
Description
A custom description of this SIP trunk
SIP Server
The address of the SIP provider peer
Port
The port on which the SIP service resides (defaults to 5060)
Username
The registration user name provided in the registration string to

the SIP provider when registering. When registering, the
username must be the same as the username configured in the
SIP provider.
When not registering, if Identify by username is enabled, this
username is used to match incoming SIP calls. I.e. the fromuser field for the remote peer must be set to this value.
Password
The registration password required by the SIP provider
Act as Provider
When enabled, this trunk allows other SIP users to register with
it. An UNKNOWN peer will be created and the IP address of
the peer will be associated with this trunk on registration.
Note: When configuring a Com.X to Com.X SIP trunk, if the
registering trunk is configured and applied on the registering
Com.X before the SIP provider configuration is applied on the
Provider Com.X, the registering box will need a 'sip reload' to
initiate registration. We therefore recommend configuring and
applying the provider trunk before the registering trunk. SIP
can be reloaded from the Monitoring Reload menu
Register with
Provider
When enabled, the SIP trunk will attempt to register its IP

address with the SIP host at the provider address configured.
The registration string will use the configured username.
Identify by
username
When enabled, will use the username field to identify a SIP

client. This is necessary when multiple SIP trunks are
registered with the same SIP provider from the same IP
address and allows the provider to determine the incoming
trunk from the SIP FROM field provided. The remote SIP peer's
from-user field needs to match this trunk's name and username
field if enabled.
Table 15: SIP trunk basic configuration

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Figure 92: SIP trunk basic configuration options.

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Field
Content
Proxy
The SIP proxy to use when establishing outgoing connections in

peer to peer setups. This will in most cases be left blank when
registering with a service provider, although some service providers
may require registration via a SIP proxy.
Proxy port
The port on which the SIP proxy service resides (defaults to 5060)
From user
Defaults to the extension number of the initiator of the SIP call. If

the provider rejects the registration, the from user might need to be
set to the provider's registration / account / username. The fromuser field only affects SIP calls, and not SIP trunk registration.
This fields overrides the caller ID number sent in the SIP call setup
headers.
From domain
A domain to append to the from-user during call setup. I.e. fromuser@from-domain.
Auth user
An authentication username to use during SIP registration and call

setup.
NAT
When enabled, the system ignores the SIP and SDP headers'
address and port and replies to the sender's address and port. This
should be enabled when the Com.X is located behind a NAT router.
Secure
If this is setting is enabled, after registration with the SIP provider,

all invites will require successful authentication as well as port
number matching.
Disabling this settings allows invites to proceed without further
security requirements after successful registration.
Skype
Indicates that the trunk is to connect to a Skype Connect SIP server

and configures Skype-specific trunk parameters appropriately.
Retry
registration
If network problems result in trunk intermittently becoming

unavailable, enable this option in order to keep retrying registration
every 30 seconds, indefinitely. If this option is disabled, registration
attempts will stop after 10 failed attempts.
Qualify
connection
When enabled, checks the reachability of the peer every 60

seconds.
Note: SIP trunk qualification is an additional (non-standardsbased) feature which allows for monitoring of SIP trunks. Not all
SIP providers support this additional feature. For providers that do
not support this add-on feature, the standard SIP trunking
configuration is recommended (Qualify turned off)
Contact # for
incoming
A DID number to assign to incoming calls on this SIP trunk. This

facilitates inbound routing where the DID is not provided by the SIP
provider.

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Field
Content
Maximum #
channels
The maximum number of simultaneous SIP channels supported on

this trunk. Leave blank for no limit.
T.38 fax mode Configures the trunk as a T.38 end-point, allowing faxes to be sent
and received using UDP over IP using the T.38 fax protocol. Also
configures UDPTL and datagram settings accordingly.
Max datagram Fine-tuning for T.38
size
Table 16: SIP trunk advanced configuration
Codecs supported by a SIP trunk can be selected by using the Codecs button. Use
Ctrl-Left-Click (holding Ctrl down) to select multiple codecs. Individual codecs can be
moved up and down the codec priority list by selecting the codec entry, and selecting
the Up or Down buttons.
When negotiating a SIP call, the codec with the highest priority (at the top of the list)
will be attempted first, then the second and so on. Codecs may specify the
integration time in milliseconds after the codec name in the form codec:integration
time, e.g. 'g729:40'
Note: Com.X systems support dynamic RTP payloads in the range 96-127. These
are set up per-call by the peer using the SDP a:rtpmap parameter.

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3.8.3.4 Editing a SIP trunk
To update the configuration of a SIP trunk, select the Edit option from the trunk's
context menu.
3.8.3.5 Removing a SIP trunk
To remove a SIP trunk, select Remove from the Sip Provider node's context menu.
Note: Removing a provider will remove the SIP trunk from all routes that are using it,
possibly resulting in routes with no trunks associated.
Note that if the trunk was a provider, with active registrations from clients, the clients
need to re-register, or their registrations will remain invalid until they time out.
Figure 93: SIP provider deletion confirmation
3.8.4. Inter-Asterisk Exchange (IAX) trunks

IAX is the native Asterisk PBX server to server communications protocol designed to
work well across firewalls and NAT.
Multiple IAX trunks can be configured for call routing. IAX trunks can be configured
as providers, or as peers registering with providers.
IAX supports multiplexing of channels over a single link. Data from multiple calls are
merged into a single stream between two endpoints, reducing the IP overhead
without creating additional latency.
If the Com.X is located behind a NAT/firewall on the network, configure the
NAT/firewall to forward UDP port 4569 to the Com.X internal IP address.
3.8.4.1 Adding an IAX trunk
To add an IAX provider, activate the Add menu on the Trunks panel. Select IAX
Provider.
IAX trunk configuration
IAX trunk configuration is divided into basic, advanced and codec configuration,
accessible from their accompanying tabs on the IAX Provider form.

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Field
Content
Name
The unique name of this IAX trunk. If this trunk is an IAX

provider, the name of the trunk must match the username.
Description
A custom description of this IAX trunk
Registrar
The address of the IAX provider's registration server
Username
The registration user name provided in the registration string to

the IAX provider when registering. When registering, the
username must be the same as the username configured in the
IAX provider.
Password
The registration password required by the IAX provider
Act as Provider
When enabled, this trunk allows other IAX users to register with
it. An UNKNOWN peer will be created and the IP address of
the peer associated with this trunk on registration.
Note: When configuring a Com.X to Com.X IAX trunk, if the
registering trunk is configured and applied on the registering
Com.X before the IAX provider configuration is applied on the
Provider Com.X, the registering box will need a 'iax2 reload' to
initiate registration. We therefore recommend configuring and
applying the provider trunk before the registering trunk.
Register with
Provider
When enabled, the IAX trunk will attempt to register its IP

address with the IAX host at the provider address configured.
The registration string will use the configured username.
Table 17: IAX trunk basic configuration
Field
Content
Qualify
connection
When enabled, checks the reachability of the peer every 60

seconds.
Contact # for
incoming
A DID number to assign to incoming calls on this IAX trunk. This

facilitates inbound routing where the DID is not provided by the IAX
provider.
Maximum #
channels
The maximum number of simultaneous IAX channels supported on

this trunk. Leave blank for no limit.
Table 18: IAX trunk advanced configuration
Codecs supported by an IAX trunk can be selected by using the Codecs button. Use
Ctrl-Left-Click (holding Ctrl down) to select multiple codecs. Individual codecs can be
moved up and down the codec priority list by selecting the codec entry, and selecting
the Up or Down buttons.

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When negotiating an IAX call, the codec with the highest priority (at the top of the list)
will be attempted first, then the second and so on. Codecs may specify the
integration time in milliseconds after the codec name in the form codec:integration
time, e.g. 'g729:40'
3.8.4.2 Editing an IAX trunk
To update the configuration of an IAX trunk, select the Edit Provider option from the
trunk's context menu.
3.8.4.3 Removing a SIP trunk
To remove an IAX trunk, select Remove from the IAX Provider node's context menu.
Note: Removing a provider will remove the IAX trunk from all routes that are using it,
possibly resulting in routes with no trunks associated.
Note that if the trunk was a provider, with active registrations from clients, the clients
need to re-register, or their registrations will remain invalid until they time out.
3.9 Budget Trunks

Each Voip trunk and trunk group can be configured as a budget trunk. These trunks
are associated with a variable minutes value that depletes as outbound calls are
completed. These trunks are skipped in outbound routes once the balance of
remaining minutes reaches zero. This minutes value is reloaded on a userconfigurable schedule, either monthly or weekly.
3.9.1. Configuring Budget Trunks

To configure budget settings on any VOIP trunk, or any trunk group, select the trunk
in the trunks panel of the GUI, right-click and select the Budget Settings.
Figure 94: Budget configuration settings


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Field
Content
Budget Mode
Select whether the trunk has an associated budget will

be skipped in outbound calling routes when the budget is
reduced to zero. Budget trunking is enabled when this is
set to Skip When Depleted.
Reload Period
Selects the frequency at which the monthly budget is

loaded on to the trunk. Monthly and Weekly patterns are
available.
Reload day/date
The day of the week/month on which the budget will be

reloaded. If the reload period is weekly, the reload day
can vary between 1 and 7, corresponding to Monday and
Sunday repsctively. If the reload period is monthly, days
1 to 28 are available and indicate the date on which the
budget is to be reloaded.
Initial Charge Period

(Sec)
Normal Charge
Period(sec)
Reload Minutes
The number of minutes to reload to the system on the

selected reload date.
Carry Over (%)
This indicates the percentage of the remaining unused

minutes at the end of the reload period to be carried over
into the next period.
Current Minutes
The initital minutes to be added to the trunk as soon as

budget options are configured.
Last Reload Date
Automtacially populated: This field serves to indicate the

last date on which minutes were reloaded to the trunk
group.
Next Reload date
Automatically generated: This field serves to indicate the

date on which the next reload of minutes is scheduled.
Note: Budget trunk configurations are backed up as a part of the configuration back
up. If a backup is made whilst a trunk has minutes remaining, these minutes will be
set as the Current Minutes field when the backup is restored. For accurate budget
trunking, this field needs to be manually reset to zero after a configuration back up
has been restored.

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3.9.2. Budget Trunk Behaviour

Budget trunking allows for outbound calls to be completed only whilst the budget
remaining on the trunk exceeds zero seconds. In the event that the remaining budget
on the trunk depletes to zero or beneath, no outbound calls over that trunk will be
permitted. Trunkgroups with zero budget will be bypassed in outbound routes, and
routes containing only trunks with zero budget will be disabled.
Note: If the remaining budget exceeds zero seconds, an outbound call can be
placed. If the budget declines to zero during a call,no further outbound calls will be
allowed, but any current calls will be allowed to be completed, which may result in a
negative remaining budget.
3.10 Outbound route configuration

A variety of trunk types can be combined and prioritized to form outbound routes,
routing calls based on dial-patterns. The Com.X supports routing to trunk groups, as
well as individual FXO trunks and BRI channels.
3.10.1. Viewing configures outbound routes

The list of existing configured outbound routes can be accessed by selecting Start,
followed by Configuration and the Outbound panel. For each route listed, the route's
name, the pin code and an indication of whether the route is an emergency and / or
intra-company route is displayed. Routes can be managed from their context menu.
To prioritize one route above another, select the Up and Down options in the context
menu. Routes appearing first in the list have the highest priority. Dial pattern
matching tries each route in turn for a match, starting with the highest priority route
and progressing to the last route in the list.
3.10.2. Adding outbound routes

New routes can be added by selecting the Add button on the Outbound panel. For
the route configuration, please see 3.10.3.

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3.10.3. Configuring outbound routes

Field
Content
Name
A unique name for the route.
Pin
If set, after a user has dialed a number that matches the dial
pattern of this route, the user will be prompted to enter the pin
code in order for the call to proceed.
Pin sets
If more than one pin should allow access to this route, leave the
Route password field empty and specify a pin set here. The pin
set needs to exist before referring to it here. Please see section
4.14 for details on configuring pin sets.
Time Limit
Optional hard timeout setting. If set, calls placed through this

outbound route will be terminated after the configured number
of settings. Leave blank for no time limit.
Emergency
When enabled, emergency dialing indicates that the emergency

CID set on a device or port should override all other CID
settings and be sent with the call information.
Intra-company
When enabled, the CID sent with the call information is the
internal, local CID, and not the outbound CID of the extension
or trunk.
Table 19: Outbound route configuration
Note: If changes are applied while an outbound route has no associated trunk, the
dial patterns for that route will be lost after the apply.
3.10.4. Dial Patterns

3.10.4.1 Adding dial patterns to an outbound route
The Dial Patterns for outbound routes are analyzed, and the highest priority route for
which a pattern match is made is tasked with routing the call. New dial patterns can
be added to a route by selecting New from the context menu of the Dial Patterns
section of the route. The route has to be edited first before this option becomes
available. If no existing patterns are included in the route, the context menu can be
accessed by right-clicking the Dial Pattern header.
3.10.4.2 Dial pattern syntax
Dial patterns consist of the following characters only:
0123456789XZN|.![],-+
X matches any digit from 0 9
Z matches any digit from 1 9
N matches any digit from 2 9
| separates a dialing prefix from numbers dialed, e.g. 123|4567 would match the
dialed number 1234567, but would only send 4567 on to the trunks.

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. matches one or more digits, e.g. 0. would match a 0 followed by one or more digits
! provides legacy overlap dialing support and matches as soon as it can
unambiguously, without waiting for the dialing to complete, allowing subsequent
digits to be transferred directly to the trunk (only supported on PRI channels).
[ ] allows a collection of digits to be matched, e.g. [7-9] would match 7, 8 and 9
- indicates a range of sequential numbers to match
+ in conjunction with | allows insertion of digits into the dial string, e.g. to change
0114567890 to 27114567890 use dial rule '27+0|X.' in the Dial Rules section on the
trunk from the Advanced menu
Removing Dial Patterns from a route
A Dial Pattern can be removed from a route by selecting Remove from the pattern's
context menu.
3.10.5. Adding trunks to an outbound route

Trunks can be added to a route by selecting Add from the context menu of the Trunk
Priorities section of the route. The route has to be edited first before this option
becomes available. If no existing trunks are included in the route, the context menu
can be accessed by right-clicking the Trunk Priorities header.
A list of available trunks, including analogue trunks (FXO), BRI trunks, PRI, SIP
providers and Trunk Groups will be displayed. Multiple trunks and groups can be
selected for addition by using Ctrl-Left-Click.
Once added, trunks can be prioritized by selecting a trunk and selecting Up or Down
from the trunk's context menu. Calls will be attempted on the trunks in the order that
they appear in this list.
3.10.6. Removing Trunks from a route

A Trunk can be removed from a route by selecting Remove from the trunk's context
menu.
3.10.7. Deleting outbound routes

To delete an outbound route, select Delete from the route's context menu.
3.10.8. Outbound route priority

Outbound routes can be moved up and down the overall list by right-clicking and
selecting Up or Down. An outbound number that matches a pattern in more than one
route will be routed via the highest priority route.
3.11 Inbound route configuration

Inbound routes allows incoming calls to be routed to a particular destination based on
the incoming call's DID and CID number pair. The Com.X supports a rich set of
destination types, greatly simplifying incoming call routing.
Note: The Com.X will automatically try and find extension numbers that match
incoming DIDs. E.g. if the system receives a call from a BRI link with DID 9031, and
an extension 9031 exists, no inbound route is necessary. The call will automatically
be routed to extension 9031.

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3.11.1. Viewing existing inbound routes

This list of configured inbound routes can be viewed by selecting Start, followed by
Configuration and the Inbound panel. For each configured inbound route, the DID
and CID pair used to match the route, the route description, destination type and
destination are displayed. To manage inbound routes, select an option from the
desired route's context menu.
3.11.2. Adding a new inbound route

To add a new inbound route, select the Add button on the Inbound panel.

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3.11.3. Configuring inbound routes

Field
Content
DID
Direct Inward Dialing allows a number of incoming numbers to

be serviced across a limited number of physical connections
from the telco to the PBX. This field may contain a pattern to be
matched against an incoming number. Please see 3.10.4.2 for
more on the dialed pattern syntax. If left blank, this route will
match any dialed number or if no DID information is present.
Be careful of using a single '.' as the pattern for an inbound
route, as this might also match dial plan constructs such as fax
and hangup and may result in unexpected behaviour. Rather
use 'X.'
The DID and CID together form a pattern matching pair which
must be unique amongst the list of inbound routes. If this rule
matches the DID and CID best, the route is used to route the
call.
FXO ports do not provide DID information from the telco.
However, the FXO port configuration on the Hardware Panel
allows a DID to be assigned to calls originating on the FXO for
routing purposes.
MSN note: DID numbers typically are sequential, and so the
telco might only signal a subset of the DID numbers, e.g. the
last 4 digits. MSN numbers are not guaranteed to be
consecutive. As a result the MSNs signaled by the telco might
be longer than the DIDs. The full MSN number arriving at the
PBX needs to be matched in order to process the call.
CID
Caller ID number is the caller's number sent to the PBX by the

telco. This field may contain a pattern to be matched against an
incoming caller ID. Please see 3.10.4.2 for more on the dialed
pattern syntax. If left blank, this route will match any caller ID or
if no CID information is present.
Note: The DID and CID together form a pattern matching pair
which must be unique amongst the list of inbound routes. If
this rule matches the DID and CID best, the route is used to
route the call
Description
A custom description describing the inbound route
Destination Type
Extension, Day/Night, Conference, Terminate, Ring Group,

Voicemail (busy), Voicemail (no message), Voicemail
(unavailable)
Destination
The destination extension, conference extension, termination

destination (busy, congestion, hangup, on hold forever, ring, sit
ring), ring group number, voicemail extension. The list of
available destinations reflects the current configuration of
features in the system.

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Table 20: Inbound route configuration
3.11.4. Configuring call destinations

3.11.4.1 Conferences
To configure conferences, select Advanced from the main menu bar and
Conferences. Once configured, the destinations will become available selections in
the Inbound Routes destination menus.
3.11.4.2 Day/Night
To configure day/night inbound routing modes, select Advanced from the main menu
bar and Day/Night Control. Once configured, the destinations will become available
selections in the Inbound Routes destination menus.
3.11.4.3 Extensions
To configure extensions, select Start from the main menu bar, followed by
Configuration and the Extensions panel. Please see 3.7 for more detail. Once
configured, the destinations will become available selections in the Inbound Routes
destination menus.
3.11.4.4 Ring Groups
To configure ring groups, select Advanced from the main menu bar and Ring Groups.
Once configured, the destinations will become available selections in the Inbound
Routes destination menus.
3.11.4.5 Termination
The following termination options are available for inbound routing:
busy: a busy tone is provided to the caller
congestion: an 'All circuits are busy now' message is provided to the caller
hangup: hang up the call
on hold forever: places the call on hold without termination
ring: provides the caller with a ring tone
sit ring: provides the caller with a specific ring tone.
3.11.4.6 Voicemail
To choose Voicemail for an extension as a destination, select Start from the main
menu bar, followed by Configuration and the Extensions panel. Select the desired
extension and edit the extension configuration. Please see 3.7 for more details. Only
extensions that have voicemail enabled will be available for voicemail termination.
3.11.4.7 IVR
Interactive Voice Response systems allow the playback of a set of audio messages
and the transfer of calls based on user interaction via the phone keypad. To select an
IVR, first configure an IVR entry and detailed in 4.7 and then select it here.

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3.11.4.8 Phonebook
To choose an entry in the phonebook as a destination, first configure the phonebook
entries and then select the appropriate entry from the options menu for the
phonebook destination.
3.11.5. Deleting an inbound route

To delete an inbound route, select Delete from the route's context menu.
3.12 FlexPath configuration

The innovative Com.X FlexPath feature is extremely powerful and allows flexible
routing, trunk to trunk routing, gateway and class of service capabilities. FlexPaths
allow grouping of call sources into contexts (paths) which can be associated with
outbound routes, inbound routing tables, transformations, destinations, etc.
FlexPaths are not required for many typical PBX configurations, but are extremely
useful for special routing needs, gateway configurations, partitioning and other
applications.
Fundamentally, the FlexPath feature provides customisable call handling through
hand crafted routing paths.
To access FlexPath, select Configuration from the Start menu, and then the
FlexPath panel.
3.12.1. FlexPath Fundamentals

Every SIP or FXS Extension Port, FXO, BRI or PRI Trunk and VoIP Provider Trunk is
a candidate for FlexPath routing.
In the FlexPath configuration panel, all these ports are displayed in a tree structure in
the right hand column of the screen, along with the selected routing path for each
port. By default, no FlexPaths are defined and every port is displayed with the
default path.
A port configured as default will route as follows:
FXS, SIP phone port routed as internal extensions, i.e. access to internal
destinations and Outbound routes.
FXO, BRI, PRI, VoIP trunks routed as inbound, i.e. access to Inbound
routes and DID mapped extensions only
In order to change the default routing, one or more FlexPaths are created, and one or
more extension ports or trunks are placed into the FlexPath. A port that is mapped
to a flexpath is indicated in the tree on the right as <port name> -> <flexpath name>
instead of <port name> -> default. Right clicking on the port text allows selection of
default or any of the configured FlexPaths.
An extensions port or trunk assigned to a particular FlexPath follows the routing rules
of that FlexPath instead of the default routing as described above.
FlexPaths are also available as destinations for Inbound routes (on the Inbound
panel) i.e. custom FlexPath routing can be applied after initial Inbound routing. In
this case, the trunk port will typically be assigned to the default routing context to
enable normal Inbound handling.
Each FlexPath has the following:
A name and description


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A number of configuration parameters
One or more FlexPath Routes
Types of FlexPath Routes available are as follows:
Include the selected routing set is simply included (routing sets include
Inbound, Internal, other FlexPaths and Outbound routes)
Match a pattern for matching dialled number/DID and/or CID is configured,

along with a destination route set as above and an optional transform to the
dialled number. In this case, only matching numbers are routed.
Start a route set is selected for calls with no dialled number information
3.12.2. FlexPath Route Types

3.12.2.1 Include
Include routes allow the inclusion of all, or part of existing routing contexts configured
in the system, including Inbound, Outbound and other FlexPaths. Calls originating
from ports in a FlexPath that has includes will, if no other route entries in the
FlexPath match the call source, be forwarded to the route destinations specified in
the order of includes.
Inclusion type
Description
Inbound DID
Includes inbound rules that match incoming DIDs
Inbound Direct
Includes configured extensions only (including services that

have extensions, such as queues and groups). Excludes
feature codes. This type forms a bridge between DIDs and
internal extensions. If a call is inbound with a DID that matches
an extension, having Inbound Direct enabled allows the call to
be routed directly to the extension without the need for an
inbound route.
Internal
Includes configured extensions and feature codes, as well as

the phonebook and other internal applications.
Table 21: FlexPath route types
Outbound destinations can also be included, as well as other FlexPaths. A FlexPath

can include only one additional FlexPath. If multiple FlexPaths need to be in the
include chain, chain them together by including the second in the first, the third in the
second and so forth. Though multiple FlexPaths includes in a FlexPath are allowed,
only the first will be processed.
3.12.2.2 Match
Matching routes allows calls to be routed based on DID and CID matches. This
functionality differs from normal inbound routing in that, upon match, the DID can be
transformed, routed to a subset of inbound routes, or routed to an outbound route
(i.e. trunk to trunk routing.)

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3.12.2.3 Start
Start routes are special cases of Match routes and match when no DID are available.
These routes are used to assign DIDs to originating calls when no DID information is
present. Start routes will only match if no DID are available. If it is possible that a DID
will be present, include a Match rule as well.
Note: An Include route will take preference over a Start route.
3.12.3. FlexPath Destinations

Calls originating from ports that have been included in a FlexPath may be routed to
the following destinations:
Destination
Description
FlexPath
Existing FlexPaths. Beware of circular references. Once a call

has been routed to another FlexPath, the matching process
starts with the first route in the destination FlexPath.
Other
Internal extensions, feature codes and existing inbound routes.

Please see 3.12.2.1 for more detail.
Outbound
Existing outbound destinations.
Table 22: FlexPath destinations
3.12.4. Viewing existing FlexPaths

To view existing FlexPaths, select Start Configuration FlexPath.
For each FlexPath, the path name and the current routes included in the path are
displayed. For each route, the route name, DID pattern match, CID match,
description, transform, destination type and destination are shown if available.
Routes are processed in the order in which they are listed in a FlexPath. Paths are
sorted in alphabetical order.
3.12.5. Adding FlexPaths

To add a new FlexPath, select the Add Path button on the FlexPath panel.

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Field
Content
Name
Unique name for the FlexPath. Spaces are not allowed in the
name.
Description
Custom description for the FlexPath.
Account Code
If configured, adds an account code to calls within a FlexPath

which can be used to group calls for billing purposes.
Lookup Caller ID
If checked, the normal caller ID lookup will be performed prior

to routing. If unchecked, the raw port ID will be used for CID
routing.
Allow Directed
Pickup
Other users of the system will be able to answer ringing calls to

extensions in the flexpath using the directed pickup. (by dialing
**<exten> from any handset)
Table 23: FlexPath configuration
3.12.6. Deleting FlexPaths

To delete a FlexPFlexPath, select Delete from the path's context menu. Confirm
deletion to delete.
Note: Ports that have beenassigned to a deleted FlexPath will become invalid,
resulting in no service for calls originating on such ports. Service to these ports can
be restored by assigning the ports either to the default FlexPath or to another
FlexPath.
3.12.7. Adding routes to a FlexPath

To add routes to a FlexPath, select Add Route from the path's context menu.
Field
Content
Routing mode
One of Include, Match or Start. Please see 3.12.2 for more

detail on FlexPath route types
Pattern
The DID pattern that this route will match
CID
The CID pattern that this route will match
Description
Custom description for this route
Transform
If specified, the originating call's DID will be transformed as

configured. Please see 3.12.11.8 for more detail on
transformations.
Destination
The routing destination. Please see 3.12.3 for more detail on

FlexPath destinations.
Table 24: FlexPath route configuration


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3.12.8. Changing route priority in a FlexPath

Routes that have been added to a FlexPathFlexPath are processed in the order they
appear in the FlexPath. Once a route has matched in a FlexPath, other routes are
ignored. If no Match or Start routes match, Include routes match in the order they are
listed.
The priority of a route in a FlexPath's match list can be changed by selecting Move
up or Move down from the route's context menu.
3.12.9. Editing routes in a FlexPath

To edit a route in a FlexPath, select Edit from the route's context menu. Please see
3.12.7 for more detail on route configuration.
3.12.10. Deleting routes from a FlexPath

To delete a route from a FlexPath, select Remove from the route's context menu.
Note: Empty FlexPaths are valid. A port assigned to an empty FlexPath will not be
able to route calls.
3.12.11.
FlexPath use cases
3.12.11.1 Class of service

Any source of a call in the system can be pre-processed using a FlexPath. This
allows class of service control on both outgoing and incoming calls.
Note: To block an extension from making calls, place it in an empty FlexPath (i.e. a
FlexPath with no routes.)
Note: To allow phones to only make internal calls and limit outbound calls to the
phonebook, create a FlexPath with only one route. The route should contain one
include mode with destination 'Internal'.
Note: Assigning analogue phones to class of service FlexPaths results in service
loss while apply is in progress. Assignment of SIP phones to FlexPaths can be
applied with no break in service.
Scenario
Extension 301 is required to have access to internal and national dialing only.
Extension 302 is required to have access to internal, national and

international dialing.
Configuration:
Extensions:
Two SIP extensions are configured:
Two outbound routes:
national access
international access
Two FlexPaths:

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pathNational includes the national outbound route and the Internal include.
pathInternational includes both the national and international outbound routes

as well as the Internal include.
Port assignments:
301 is assigned to pathNational
302 is assigned to pathInternational
3.12.11.2 Partitioning / multi-tenant

Scenario:
Company A and Company B share a PBX. Each company has reserved a set of BRI
channels for their exclusive use.
Company A: 4 BRI channels on BRI ports d1 and d2
Company B: 4 BRI channels on BRI ports d3 and d4
Extensions 8xxx belong to Company A with reception at 8000.
Extensions 9xxx belong to Company B with reception at 9000
Outgoing calls must be routed to the companies' BRI channels.
Incoming calls on the companies' BRI lines must be routed to their reception.
Configuration:
Extensions:
Extensions ports (both FXS and SIP) are assigned the appropriate extension
numbers for each company.
Two outbound routes:
companyAout routes over Company A's BRI channels
companyBout routes over Company B's BRI channels
Two Inbound routes:
Fixed incoming DID 8000 is routed to Company A's reception. This DID will
be assigned to BRI calls originating from Company A's BRI channels in a
FlexPath
Fixed incoming DID 9000 is routed to Company B's reception. This DID will
be assigned to BRI calls originating from Company B's BRI channels in a
FlexPath
FlexPaths:
companyA-in: Add a Start route that transforms incoming BRI calls without
DIDs to the reception extension and forward these to the Inbound routes.
companyA-in: Add a Match route that transforms incoming BRI calls with any
DID to the reception extension and forward these to the Inbound routes.
Assign Company A's BRI trunk groups to the companyA-in FlexPath
companyA-out: Include an internal route that facilitates dialing internal

extensions

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companyA-out: Add a rule that matches outgoing numbers other than

extensions and routes them to the companyAout outbound route.
Assign Company A's extension ports to the companyA-out FlexPath
companyB-in: Add a Start route that transforms incoming BRI calls without
DIDs to the reception extension and forward these to the Inbound routes.
companyB-in: Add a Match route that transforms incoming BRI calls with any
DID to the reception extension and forward these to the Inbound routes.
Assign Company B's BRI trunk groups to the companyB-in FlexPath
companyB-out: Include an internal route that facilitates dialing internal

extensions
companyB-out: Add a rule that matches outgoing numbers other than

extensions and routes them to the companyBout outbound route.
Assign Company B's extension ports to the companyB-out FlexPath
Note: Inter-company calls are facilitated by default and can be accessed by directly
dialing a company's extensions (e.g. the receptionist for Company B could dial 8000
to reach the receptionist for Company A.) To disable this, create empty FlexPaths
that terminate service for cross-company numbers:
Company A: Create an empty FlexPath that Matches dial pattern 9XXX
Company B: Create an empty FlexPath that Matches dial pattern 8XXX
Note: This is a simplified example. FlexPath could also be used to match incoming
DID ranges and route these to separate inbound routes for different partitions.
Note: The incoming CID can also be prefixed with a custom name for inbound
routes. To configure this, access the FreePBX Inbound Route configuration (please
see section 6.5) and configure the 'CID name prefix' field.
3.12.11.3 Simple partitioned DID routing:
If DIDs are provided by the telco, the use of inbound FlexPaths are unnecessary and,
provided the extension numbers created for each company matches the DIDs
provided over by the Telco, calls will automatically route to the correct extensions.
Scenario:
DID ranges routed by the Telco as follows:
Company A: 5559000 5559049 -> ports d1 and d2
Company B: 5559050 - 5559099 -> ports d3 and d4
Telco communicates the last 4 digits of DID across channels during call setup.
Configuration:
Creating extensions 9000 9049 for Company A and 9050 9099 for Company B
assigned to the appropriate FXS and SIP ports will result in automatic call routing to
the correct extensions for incoming calls.
3.12.11.4 Trunk to Trunk gateway
Scenario:

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A company has a head office in a major city, and a remote office in another city. The
company has an agreement with an internet connectivity provider for a high quality IP
link between the two offices. The company requires a PBX system at the head office,
with internal extensions at both office locations, the two offices SIP trunks and PSTN
trunks provided for call routing at the head office location.
Configuration:
Both the head office and the branch office are provided with Com.X units.
Extensions:
Head office extensions configured as 101 105
Branch office extensions configured as 201 205
Sip trunk configuration:
Head office SIP trunk configured as provider
Branch office SIP trunk configured to register with head office
Ensure that Proxy, NAT, connection qualification and codec selection options
are configured appropriately.
Outbound route configuration:
Head office outbound route routes calls to branch extensions across the SIP
trunk
Head office outbound route routes all calls to PSTN
Branch office outbound route routes all calls (including calls to head office
extensions) to head office across the SIP trunk
FlexPath:
Head office configured with a FlexPath that places calls originating from the
SIP trunk either internally or routes them trunk-to-trunk to the PSTN outbound
route.
3.12.11.5 Legacy PBX Remote extension gateway

Scenario:
A company has a head-office with 4 FXO trunks to the telco and 10 analogue
extensions on a legacy PBX. The legacy PBX has a number of unused extension and
trunk ports available. The company is opening a second branch, geographically
separated from the first, and desires a stand-alone PBX at the branch, with its own
BRI telco interfaces, but the ability for head-office and branch to phone one another
using extension numbers across a SIP trunk. There will be four extensions required
at the branch, with potential for some growth.
Configuration:
Install a Com.X1 at both sites. For the branch, the Com.X1 serves as a PBX, with the
BRI outbound routes routing across the BRI interface.
In the head-office, place the Com.X1 behind the legacy PBX, connecting its FXO
ports to free FXS ports on the legacy PBX and configuring the SIP trunk to the
branch.
Extensions:

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In this configuration, the branch extensions look like additional extensions on the
legacy PBX, but the Com.X1 gateways these across the SIP trunk to the branch.
Calls from the branch to head-office are routed across the SIP trunk, prepended by
caller ID, and the Com.X1 at the head-office then routes based on the branch CID to
the appropriate FXO port, in effect using the correct legacy PBX extension for activity
from the associated branch extension.
FlexPath:
At the branch, create a FlexPath with a Match rule that matches head-office
extensions with an outbound route as destination and the following transform:
CD.
At the head-office, add FlexPaths with Match rules that match the DIDs
transformed by the branch and routes them based on the received CID
number in the DIDs to the correct FXO port.
At the head office, assign DIDs to the Com.X1 FXO ports and add a FlexPath
that pre-pends this DID to the dialed number before routing to the branch
across the SIP trunk.
At the branch, add FlexPaths with Match rules that routes to the appropriate
extensions based on the DIDs received from the head-office.
3.12.11.6 Legacy PBX feature set extension

Scenario:
A company has a significant investment in a legacy analogue PBX. This investment
includes some features, e.g. voicemail, routing, groups, feature codes and IVRs that
the company would like to preserve.
They would how-ever like to extend the legacy PBX with new SIP extensions and
additional features, including call recording, least cost routing and additional SIPbased features for the new SIP phones, and possibly enable the PBX to support a
remote office as described in section 3.12.11.5. The existing PBX cannot be
extended, or the cost associated with such extension is prohibitive.
As a result, the company want existing analogue phones to be served dial-tone
directly from the existing PBX (preserving the existing feature set) and the new
features and LCR to be available to new SIP extensions. Call recording should be
available on all extensions, and it should be possible to block extensions from dialing.
Configuration:
Install a Com.X solution with sufficient analogue FXS and FXO ports for the
company. The Com.X solution provides SIP extensions, LCR and the other features
mentioned to the business.
Connect the analogue phones to the Com.X solution's FXS ports and the Com.X
solution's FXO ports to the existing PBX's FXS ports.
For each FXO port, assign a DID matching the legacy PBX's extension. Create an
extension with the same extension number for each of the legacy PBX's extensions.
For each of these FXS ports, create an outbound route with dial pattern '!' (please
see section 3.10.4.2) and a FlexPath with the extension FXS port as the source, and
destination the associated outbound route. Enable immediate dialing on the
extension port (please see section 3.6.2)

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Using this configuration, when an analogue phone goes off-hook, the legacy PBX
would be providing dial-tone directly. If the legacy PBX's extension rings, the
associated Com.X extension also rings. This enables calls to be recorded on the
legacy analogue extensions. Though dialed numbers cannot be recorded in CDR
records for the legacy extensions (since the number is not collected by the Com.X
solution but rather passed straight through to the legacy PBX), the duration of calls
originating from legacy extensions can be recorded. Extensions can also be blocked.
Call recording would be fully featured for the new SIP extensions.
3.12.11.7 Immediate dialing
Emergency and single-purpose devices can be configured to dial immediately on
pick-up.
Scenario:
An emergency phone is required to dial 911 when picked up.
Configuration:
Extension:
From the Hardware panel, edit the port to which the device is connected and
enable Immediate mode.
FlexPath:
Create a FlexPath that includes a Start route, configured to transform the DID
to 911
Assign the emergency port to the FlexPath.
3.12.11.8 Transformations
The Com.X allows both outgoing and incoming dialing numbers to be transformed.
To transform, create a FlexPath and add a Start or Match route to the path, with the
required transform specified.
Transforms may contain digits 0 9 and special placeholders D, which is replaced
with the current DID and C, which is replaced with the current CID.
Scenario:
All outgoing calls must be prepended with 08 and followed by 55 and the CID, if any.
Configuration:
Add a FlexPath with a Match rule with an outbound route as destination and
the following transform: 08D55C
Assign extension ports that need this transform to the FlexPath.
Scenario:
All calls incoming on a SIP trunk needs to be prepended with 77.
Configuration:
Add a FlexPath with a match rule that includes inbound DID routing
transforms as follows: 77D
Assign the SIP trunk to the FlexPath.

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3.12.11.9
Assigning DIDs to BRI ports
When DIDs are not provided over BRI channels, DIDs can be associated directly with
BRI ports in the Com.X to facilitate call routing.
Scenario:
DIDs 9001 9004 need to be associated with 4 BRI ports.
Configuration:
Create a Trunk Group for each BRI port. BRI ports cannot be assigned to
FlexPaths directly.
Create FlexPath for each BRI port
Add Start routes to the FlexPaths that transform to the required DIDs
Set the destinations of the Start routes to Inbound DIDs for inbound call
routing.
Assign the BRI ports to their specific FlexPaths.
3.12.11.10 Load-balancing calls

Outbound calls can be load-balanced across telco interfaces or network routes using
FlexPath.
Scenario:
Company A has 8 extension, and wishes to load-balance outgoing calls across two
DSL links, each supporting up to a maximum of 4 simultaneous calls. The DL routers
have IPs 192.168.0.1 and 192.168.0.2. The company's VOIP service provider has
provided the company with the IP addresses of two SIP proxies: 10.0.0.1 and
10.0.0.2
Configuration:
Add two SIP trunks, with distinct IP address end-points (in this scenario
10.0.0.1 and 10.0.0.2) These may be two proxies with the same provider, but
for IP routing capability, the IP addresses must be different. When routing
across telco interfaces, one would replace the SIP trunks here with trunk
groups that contain the trunks to be used.
Create two outbound routes, one using the first SIP trunk and the other the
second SIP trunk (or the appropriate telco trunk groups).
Create two FlexPaths, and add an Include rule that includes the appropriate
outbound route for each. Point the first FlexPath to the first outbound route
and the second to the second outbound route.
For network load-balanced scenarios, log into the Com.X using ssh and
configure two routes for routing the traffic across the two network paths in
/etc/network/interfaces:
auto eth0
iface eth0
post-up ip
post-up ip
post-up ip
inet dhcp
route flush dev eth0
route add 10.0.0.1 via 192.168.0.1
route add 10.0.0.2 via 192.168.0.2

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3.12.11.11 FXO Trunk pick-up to test for trunk service
When installing a Com.X, at times it might be useful to determine whether a specific
trunk port has been service provisioned by a device connected to the port. For
example, the installer might want to check whether a legacy PBX is providing dialtone in the trunk, or a premi-cell is available. This might also be of use if the wiring to
the Com.X1 is not clearly marked, to determine which trunks have service.
Configuration:
create an outbound route called 'trunk-tester' with dial pattern '99!' and the
trunk of interest included in the trunks list
Dial 99
Configure for automatic test on pickup:
Replace the dial pattern above with '!'
create a FlexPath called 'trunk-tester' and use the Include mode to include the
'trunk-tester' outbound route
on the FlexPath panel, assign an extension port to the trunk-tester FlexPath

and plug a phone into the extension port
select the extension port in the Hardware panel and enable Immediate mode
Pick up the handset. If the service is active you'll hear dial-tone or the service
prompt.
Note: If immediate mode is not set for the automatic test configuration, the Com.X
will still provide dial-tone on the extension as usual, which might be incorrectly
perceived as an external trunk service, so be sure to enable Immediate mode on the
extension port.
3.13 Call recording

Please note that the call recording features provided by Far South Networks
products are best-effort services and do not meet the requirements for
recording to be used in legal cases, nor are they certified in any way by any
regulatory bodies or authorities. Use of call recording may be subject to
regulations specific to your country as well as international law.
Note: The ATMOS Call Logger solution provides a secure, scalable, cloud-based
voide logging solution. Recording files processed by the ATMOS call logger are
encrypted at source and securely uploaded, stored and backed-up on in cloud
servers. These recordings are fully compliant (FICA, FAIS, POPPI, CPA) and
suitable for presentation withing legal proceedings. See Section 6.12.1 for further
details.
Calls can be configured to be recorded on an individual extension, on all extensions

and in the incoming, outgoing or both directions, for internal and external calls.
Depending on the size of the hard disk in the Com.X configuration, capacity for call
recording may vary (E.g. a Com.X with a 250GB hard drive can record ~400,000
minutes of voice in A-law WAV format and a Com.X with a 60GB SSD drive
~500,000 minutes of voice in GSM format or 100,000 minutes in A-law WAV).

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3.13.1. Configuring individual extensions

To configure call recording setting on an individual user basis, log into the User
Portal from the Com.X's main web page, and select Settings.
To configure call recording setting for all users, log into the User Portal as 'admin'
from the Com.X's main web page, and select Settings. All users will be affected!
When set to On-Demand, calls will only be recorded when the user starts recording
using the in-call call recording feature code (*1 by default). Please see section 4.5 for
more detail on feature codes.
Note: In order for the caller or the recipient of the call to initiate on-demand call
recording, the general dial command options need to include w and / or W.
Please see 4.6.1 for more detail.
Note: Disabling voicemail on an extension will also disable call recording.
3.13.2. Recording all calls

Log into the user portal as 'admin' with password 'ari_password' and set their Call
Monitor settings for both incoming and outgoing calls to Always. Please see 3.13.1
for more detail.
Enable the Extension Recording Override setting in the system's General Settings.
Please see 4.6 for more detail. If this setting is Disabled, calls will not automatically
be recorded, even if Always is selected in the user call monitor settings.
If a user's call monitor is configured to Never record calls, calls received by that
user will not be recorded, even if Extension Recording Override is enabled.
3.13.3. On-demand recording

If so configured, users can record calls on-demand by pressing *1
On-demand recording will only work if the Advanced General settings dial options
include the correct recording permissions.
SIP phones need to translate DTMF properly (RFC2833)
Some analogue phones may not properly relay the DTMF to the PBX, in which case
on-demand recording may not work. For such phones configuring recording to
always be on is a work-around.
Pressing *1 activates recording if not active, and de-activates recording if active. The
last *1 to activate on-demand recording in a call will over-write previous recordings
for that call. I.e. if recording was active, then de-activated and re-activated, only the
re-activation recording will be present for that call.
3.13.4. Call recording conditions

Due to privacy issues in countries that enforce these, the Com.X recording feature
only records calls as follows:
Outbound calls are recorded if the caller has recording enabled in the outbound
direction or uses on-demand-recording on the call.
Inbound calls are recorded if the caller has recording enabled in the inbound direction
or uses on-demand-recording on the call.
Internal calls are recorded in both users' call monitors if the called user has inbound
recording configured, regardless of the calling user's configuration.

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Internal calls are not recorded in either of the users' call monitors if the called user
does not have inbound recording configured, or has on-demand inbound call
recording configured and has not used the on-demand feature code, regardless of
the calling user's configuration.
If on-demand recording is activated on an internal call by either party (provided that
party has on-demand recording configured), the call will be recorded, regardless of
the other party's call recording configuration.
In summary then, on-demand recording over-rides all call recording configurations.
Otherwise, the called user's call recording setting dictates the call recording behavior
for internal calls. Calls placed over or received via trunks are subject to the user's
outbound and inbound call settings.
Note: It is the legal obligation of the recording party to inform callers when recording
is activate on a system. Similarly, when on-demand recording is used to record
calls, the DTMF feature code activation might be audible to the remote party, but is
not guaranteed. In such scenarios it is the user activating on-demand recording's
responsibility to inform the other party that call recording has been activated.
3.13.5. Gateway Recording

Gateway recording can be enabled via the Global Settings menu on the settings
page of the Comma Gui (see section 3.1 for further details on global settings). This
will record all outbound calls placed over any trunk, including those with no
associated extension, for instance in a trunk to trunk gateway scenario.
Gateway call recordings are stored in /var/spool/asterisk/monitor/ and are named
with the prefix gw-rec.
Note: Outbound calls from extensions will feature two recordings if gateway
recording is enabled. One will be prefixed OUT<extension number> whilst the
gateway recording will be prefixed ge-rec
3.13.5.1 Gateway Recording Encryption

This option is configurabled from the Global Settings menu. If enabled, all gateway
call recordings will be encrypted and stored in /var/spool/asterisk/monitor/ .
3.13.5.2 Gateway recording in a-law
To have gateway recordings saved in 8-bit a-law format set the Call Recording
Format variable in the General Settings page of the Advanced menu (see section 4
for further information on advanced configuration options) to .wav as opposed to
.WAV
3.13.6. Viewing call recordings

Individual users can view and play call recordings from their user's Call Monitor by
selecting the Call Monitor option.
The system administrator can view and play all call recordings for all users by logging
into the User portal with username 'admin' and password 'ari_password'.

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The admin user's password can be changed by editing the
/usr/share/freepbx/recordings/includes/main.conf.php file and specifying the desired
password in the $ARI_ADMIN_PASSWORD field.
The system administrator can also view all call recordings (including gateway
recordings,) by selecting Call log from the Reports menu. This view allows one to
search for call recordings by month, day, destination, source, channel and call
duration. Presented for each call is the call date, the call channel, source, caller ID,
destination and call disposition (NO ANSWER, ANSWER, FAILED, BUSY). If a
recording of the call is available, a link to the recording is provided, or alternatively
the unique call identifier, which can be used to look up the call recording at an
alternative storage location. Recording filenames include the unique call id.
Figure 95: CDR Call logs
3.14 Powering the Com.X via an Uninterrupted Power Supply

Com.X products ship with a UPS service that supports standard USB-connected
UPS products. When enabled the service will do the following:
Monitor the UPS Report in /var/log/syslog when mains power goes down or is
restored.
Cleanly shut down the system when battery level is low.
Power off the UPS at the end of shut down
The UPS will automatically switch on again (after some short delay) when
power is restored. If present, it is recommended to connect applicable ADSL
routers and related network switches to the same UPS as the Com.X.
3.14.1. Configuring Comma-Ups

To enable the comma-ups service, first navigate to /etc/nut/ edit /etc/nut/nut.conf
to contain:

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MODE=standalone
Then edit /etc/nut/ups.conf to include:
driver=blazer_usb
Then restart the nut service :
sudo /etc/init.d/nut restart
Note: If the restart fails, as a result of failing to create the /var/run/nut/ directory, you
can manually create the directory, and then restart the service again.
3.14.2. Supported UPS Devices

The following UPS devices are confirmed to work with the comma-ups service. Other
devices may also function correctly:
Proline B1000
Proline B600
3.15 Securing the system

It is of utmost importance to ensure that the Com.X system is secure. If a Com.X unit
is public facing, large costs can be incurred if the VoIP configuration security is weak,
or downtime and degraded performance may result from the Operating System being
compromised. The following steps are recommended to ensure a secure Com.X
system:
3.15.1. General
Ensure that access to the unit is restricted. Install the Com.X in a secure
server room or locked cabinet. This prevents unauthorized access to the unit
using the serial port or USB keyboard, as well as physical damage to the unit
or its power supply.
Change the default password for the comma user.
Regularly check the system log files for unauthorized activity (/var/log/auth* and
/var/log/syslog*)
Use strong passwords.
Regularly backup the system configuration and important data.
3.15.2. Network
Preferably allow remote (external) access (VoIP and data) to the Com.X only
by means of a Virtual Private Network (VPN)
Configure the Com.X and telephony devices (e.g. iTA or VoIP phones) on a
separate network partition on a managed switched, or a physically separate
network.
Assign IP addresses and services on the VoIP network based on a white list
of approved MAC addresses.
Place the Com.X behind a firewall.


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Perform port forwarding on a non-standard port to port 22 on the Com.X for

SSH access. Tunnel HTTP access through the SSH connection.
Limit administrator access (SSH and web GUI) to a white list of MAC
addresses, IP addresses or networks.
Activate fail2ban (please see section 6.17)
3.15.3. VoIP
If the Com.X is accessing external VoIP trunks via a NAT router/firewall, no

special port forwarding is required. If external VoIP agents (e.g. SIP phones)
need to access the Com.X, forward only ports 5060 and 10000-20000 to the
Com.X for VoIP control and voice communication.
Configure the firewall to block all incoming VoIP traffic except that from a
white-list of remote extensions.
Ensure that all VoIP extensions and trunks have strong (generated)
passwords, different from the extension number.
Activate fail2ban (please see section 6.17)
3.15.4. PBX
Change the Com.X GUI password from the default.
Log the PBX console messages to /var/log/asterisk/messages and monitor

this periodically. A large number of rejected VoIP registrations may indicate
an attack on the system.
Use pins and pin sets on international and high-cost outbound routes,
requiring the caller to know the pin in order to place a call.
Disable feature codes for administrative features, or change the feature codes
to access those features to something other than the default. (E.g. ChanSpy,
ZapBarge).
Require all users to set their voicemail pin and user portal login passwords to
something other than the default.
Regularly backup the PBX configuration and valuable recordings.
If direct database access is configured for CDR access, limit the access to a
white list of MAC addresses, IP addresses or networks.
3.16 Call behavior on No answer / Busy

This section describes the Com.X behavior and call routing for incoming calls that go
unanswered or encounter a busy extension.
3.16.1. Unattended transfer

On unattended transfer, the person transferring the call is taken out of the loop, i.e.
the call is treated as though the caller called the final party directly. Voicemail or
follow-me for the recipient would determine where the call goes on busy and no
answer.
If both voicemail and follow-me are not configured, the extension would ring
indefinitely.

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3.16.2. Attended transfer

The person wanting to connect the caller with a third party may place the caller on
hold by picking up the second line, pressing a hold button or initiating an attended
transfer.
The caller hears music, and the intermediary has one extension channel available to
call the intended destination with and confirm availability. The intermediary can
interact with the destination and connect the caller with the destination by using a
transfer button or by putting the phone / channel down.
3.16.3. Call parking

The person wanting to connect the caller with a third party may place the caller in a
parking bay.
The caller hears music, and the intermediary has both extension channels available
for use. Once the necessary calls have been made, conferences set up, availability
determined, etc., the intermediary can pick up the call in the parking bay and transfer
or respond as required.
Should the time the caller spends in the parking lot exceed the specified timeout
period, the action configured for orphaned callers is initiated. This could include
playing an announcement to the caller and keeping the caller in the bay, or
transferring the caller to reception.
3.16.4. Ring-back
In the event that a call is transferred to a destination extension that is busy, or
unanswered, the ringback mechanism will send the call back to the transferring party,
with a modification to the CID. The new CID presented to the transferring party will
be prefixed by Ret: to indicate that the call is returning, as well as a the extension
number of the destination extension to which the original call was transferred. The
Ringback caller id will have the form :
Ret: [original intended destination extension]CID
Note that Ringback functionality is only activated on destination extensions with
follow me or voicemail enabled. Transferred calls to extensions with neither
configured will ring indefinitely if unanswered, or fail with an engaged tone should the
extension be unavailable.
Note that the follow me of the destination extension will be evoked before the
ringback occurs, so a transfer to an extension with follow me enabled, will only ring
back to the intermediary extension if the destination extension and all extensions in
its follow me list, are unavailable or unanswered.
If the ringback mechanism sends the call back to the transferring extension and this
extension is no longer available, or the call is unanswered, the destination will be as
in the table below:
Intermediary Intermediary Destination

Voicemail
Follow me
Voicemail
Enabled
Enabled
Enabled
Destination
Follow me
Enabled
Ringback
Occurs
Ringback
Failover
Destination
N/A

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Destination
extension Follow
me failover
Destination
extension
voicemail
Destination
extension Follow
me failover.
N/A
Destination
extension Follow
me failover.
Destination
extension
voicemail
Destination
extension Follow
me failover.
N/A
Destination
extension Follow
me failover.
Destination
extension Follow
me failover.
N/A

Voicemail
Follow me
Voicemail
Enabled
Enabled
Enabled
Destination
extension
voicemail
Destination
Follow me
Enabled
Destination
extension Follow
me failover.
Destination
extension
voicemail
Destination
extension Follow
me failover.
Ringback
Occurs
Destination
After
Ringback
Destination

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Voicemail
Follow me
Voicemail
Enabled
Enabled
Enabled
Destination
Follow me
Enabled
Ringback
Occurs
Destination
After
Ringback
extension
voicemail
Destination
extension
Follow me
failover.
N/A
Destination
extension
Follow me
failover.
Destination
extension
voicemail
Destination
extension
Follow me
failover.
Table 25: Ring-back behavior
Note: Users need to be informed as to what to expect. With ring-back enabled their
phones should 'ring by itself' once available. If the hand-set it picked up, they should
hear ringing while the initial calling party is being contacted.
If the other party is now on the phone and also has neither follow-me nor voicemail
enabled, when the handset of the phone on which ring-back was initiated is picked
up, the user would hear a busy tone. A second round of ring-back will then be
initiated. I.e. once the original caller (now on the phone) ends the call, his phone will
ring, and picking it up, the initial called user would be attempted again.
3.16.4.1 Disabling Ringback Functionality

From the global settings menu, ringback functionality can be customised to be
enabled or disabled for all transfers to busy extensions or unanswered extensions.
(See section 3.1 for details on configuring ringback on busy and ringback on
unanswered). Both settings are enabled by default.
If ringback on transfer busy is disabled, transfered calls to busy extensions will go
directly to the voicemail or other configured failover destination for the extension and
will not ring back to the transferring extension.

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Similarly, disabling ringback on transfer no answer will direct unanswered transfered
calls directly to the voicemail or failover destination of the unanswered extension,
rather than ring back to the transferring extension.

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4 Advanced configuration
Access to a variety of advanced features is provided via the Advanced menu.
Selecting a function from this menu opens a new window, which can be maximized
with detailed configuration options for the feature. Once configured, select Submit at
the bottom of the configuration page. Also remember to review and apply the
updated configuration to the underlying system when ready.
4.1 Announcements
Announcements added as destinations allow messages to be played on inbound
calls before transferring the calls to appropriate destinations. Announcements can
play recordings created or uploaded as detailed in 4.17.
Field
Content
Description
The description of the announcement that will appear in

selection lists (e.g. when selecting announcements for IVR)
Recording
The audio recording to play for this announcement
Repeat
The key a user may press to repeat the announcement. If this

is set, a delay will be inserted after the announcement has
been played to allow the user the option of repeating. Allowed
keys are: 0-9, *, #
Allow skip
If this is enabled, should the user press a key during an

announcement, the announcement will be skipped.
Return to IVR
When enabled, after the announcement has been played, the

call is transferred to the last IVR.
Note: Even if the call was transferred to other destinations
since the last IVR, this will still transfer the call the the last
IVR.
Don't answer
channel
If enabled, the channel will not be answered and the call

transferred to the destination. If the channel supports early
media, the announcement will be played as early media when
the next destination answers the channel.
Destination after
playback
Destination to transfer this call to. Please see 3.11.4 and 4 for
destination options and configuration.
Table 26: Announcement configuration
4.2 Callback
This feature allows users to dial into the system and hang up while ringing, after
which the system will call the user back.

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Field
Content
Callback description
A custom description for this callback.
Callback number
The number to call back. If this is left blank, the caller ID

will be used to callback.
Note: the device to be called back must support incoming
calls directly(e.g. some hotel and pay phones do not).
Note: If a callback number is not specified here, the
calling device must presents a proper CID.
Delay before callback
The amount of seconds to delay before calling back.
Destination
Once connected via callback, the destination to which to

transfer the called-back party.
Table 27: Callback configuration

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4.3 Conferences
This feature allows multiple users to join a conference with two-way audio on all
devices. Conference features include join and leave announcements, passwords,
conference administration, in-conference menus, etc.
Field
Content
Conference number
The number to dial to access the conference
Conference name
A custom name for this conference
User PIN
A pin required to be entered by users in order to join the

conference. This field is optional. The user must press # after
entering the pin.
Admin PIN
A pin that identifies the conference leader (admin). This field

is optional, unless Leader wait has been enabled. The user
must press # after entering the pin.
Join message
If set, this audio recording will be played before a user joins

the conference.
Leader wait
If enabled, the conference will not start until the leader has
joined by entering the Admin pin.
Quiet mode
When enabled, audio recordings associated with joining and

leaving the conference will not be played.
User count
If enabled, the number of users already in the conference will

be announced to a new user joining the conference.
User join/leave
If enabled, an announcement will be made when-ever a user

joins or leaves the conference.
Music on hold
If enabled, music will be played when there is only one user

in the conference.
Allow menu
When enabled, users and the admin user can access their
menus by pressing *
Record conference
If enabled, conferences using this conference number will be

recorded.
Table 28: Conference configuration

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4.4 Day / Night control

A number of day / night modes can be programmed on the Com.X. Each has a
feature index and can route calls according to its own set of rules. These allow
configuration of day / night to support multiple receptionists, tea or lunch breaks, after
hours, etc.

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Field
Content
Feature code index The index of this day/night mode. The index is the day-night
feature code (please see section 4.5 for more detail on feature
codes), with the last digit of the feature code (e.g. 280)
replaced by this day/night's index (e.g.283, 284, 288, etc.) Up
to 10 day/night entries are available, 0-indexed.
Description
Custom description for this day/night control. This is the value

that will be listed in selection lists that allow day/night
selection.
Current mode
This sets the initial and current mode for this day/night control.
I.e. if Day is currently active, and Night is selected here, night
will become active, and on system restart night will be the
initial value for this day/night control.
Optional password
If not blank, a user will have to enter this password in order to

toggle this day/night control.
Day destination
The destination to use when this control is in Day mode.

Another day/night destination can be selected here, chaining
together these conditions to form a logical AND tree. I.e. a
number of decision junctions can be joined and manipulated
by setting the modes for the day/night controls that have been
chained together. An example:
280 has Day -> 281
280 has Night -> Support ring group extension
281 has Day -> 282
281 has Night -> Alternative receptionist
282 has Day -> Reception
282 has Night -> Tea / lunch room
A receptionist can enable night mode in the evenings and day
mode in the mornings by using *280. When tea-time arrives,
the receptionist enables night mode for 281 using *281. After
tea-time she re-enables day mode for 281 using *281.
Please see 3.11.4 for detail on destinations.
Night destination
The destination to use when this control is in Night mode.

Please see 3.11.4 for detail on destinations.
Table 29: Day/night configuration
4.5 Feature Codes

An extensive list of feature codes is available to users and administrators to facilitate
administration of the system and user call preferences using a phone. For each
function, the administrator may change the feature code associated with the function,

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enable or disable the function, or revert to the default feature code associated with
the function.
Field
Content
Name
The name of the function category or the individual function.
Code
The feature code currently associated with the function
Use default
If enabled, the existing feature code will be replaced with the

default feature code for this function, if different from the
existing one.
Enabled
Allows the disabling and enabling of individual functions. If

disabled, the feature will not be available on the system. The
code associated with the feature will remain configured even
when disabled.
Table 30: Feature code configuration

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4.6 General Settings

General settings apply system-wide on the Com.X platform. Users can typically
override general settings with their own preferences, but some general settings
override all user preferences.
4.6.1. Dial command options

t: Allow the called user to transfer the call with the in-call blind transfer feature code
(default ##).
T: Allow the calling user to transfer the call with the in-call blind transfer feature
code (default ##).
r: Generate a ringing tone for the calling party
w: Allow the called user to start recording after pressing the in-call recording feature
code (default *1).
W: Allow the calling user to start recording after pressing in-call recording feature
code (default *1).
L: Limit trunk calls to a specified duration. Parameters are separated using ':' and
only the first parameter is mandatory. The first parameter indicates the maximum
duration in milliseconds of the call. The second parameter indicates when a warning
message should first be played when the specified number of milliseconds remain on
the call. The third parameter indicates the frequency in milliseconds with which the
warning message should be repeated. To limit calls to 1 hour and warn the user
every minute after only 5 minutes are left on the call, use 'L(3600000:300000:60000)'

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Field
Content
Internal Dial command

options
Options specified here applies to internal calls. Please

see 4.6.1 for more details.
Outbound dial command

options
Options specified here applies to outbound calls.

Please see 4.6.1 for more details.
Extension recording
override
If disabled, all user recording settings other than the

on-demand settings will be ignored, and no recording
other than on-demand will take place.
If enabled, the recording format setting specified here
will override user-specific format settings.
Recording settings for other modules, such as queues
and conferences remain unaffected. Disabling this
override has a positive impact on system performance,
as the checks that are performed for each call re:
recording become unnecessary.
Call recording format
Recordings will be saved on the Com.X hard drive in

the format selected here. Options are: wav, law,
alaw, GSM, SLN, G.729.
We recommend using the wav format, since it results
in higher quality recordings and does not use
additional codec translation resources, which may
affect performance on systems with high-usage.
Recording location
If this field is blank, the default location

(/var/spool/asterisk/monitor/) will be used to store
recordings.
If this field is populated, ensure that the trailing / is
included and that permissions on the directory are
correct. This is required in order for the system to
store recordings and the user's call monitor function to
download the recordings:
chown asterisk:asterisk -R <dir>/recordings/
chmod g+rwx <dir>/recordings
Also configure the ASTERISK_CALLMONITOR_PATH

entry in /usr/share/freepbx/recordings/includes/main.conf.php to point
to the new location.
Users can access their recordings via the User login
and Call Monitor feature.
If there is a requirement for an administratot to get
access to all recordings, set this location to
/var/www/recordings and secure the directory (please see
the Apache web server administration manuals). Then
access the list using http://<server-ip>/recordings/

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Field
Content
Run after record
If not empty, this field specifies a custom script to be

run after each recording has completed. The script
should be executable and the location to the script the
absolute path.
Ringtime default
This specifies the default time an extension should

ring before activating voicemail.
Direct dial voicemail prefix
Dialing the prefix specified here followed by an

extension number will take the user directly to the the
mailbox for that extension.
Note: Please ensure the prefix specified here does
not clash with any feature codes. Please see 4.5 for
more detail on feature codes.
Direct dial to voicemail

message type
The message (or no message) the user will hear

before the voicemail prompt when dialing directly into
an extension's voicemail. Can be one of:
Default - Please leave your message after the tone.
When done, hang up or press the pound key.
No message
Unavailable - The person at extension <ext> is
unavailable. followed by the default message.
Busy - The person at extension <ext> is on the
phone. followed by the default message.
Optional voicemail
recording gain
If present, applies additional gain on the recording.The

number entered is interpreted as whole-numbered
dBs.
Do Not Play "please leave

message after tone" to
caller
When checked, this option removes the default

message to be played after the selected (if any)
voicemail announcement has been made, but instead
plays the tone directly after the voicemail
announcement.
Default Context & Pri
The internal dial-plan context to use if the user dials an

extension or a short number that does not match
outbound routes. This should remain the default
unless requested otherwise for test purposes by Far
South Networks.
Timeout/#-press default
The internal destination to route a user after timeout.

This should remain the default unless requested
otherwise for test purposes by Far South Networks.
Loop Exceed default
The destination that a user is transferred to when

exceeding the maximum number of invalid selection

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Field
Content
options during a voicemail call. This should remain the
default unless requested otherwise for test purposes
by Far South Networks.
Timeout VM Msg
Indicates whether the standard voicemail instructions

should be played on timeout, or whether the tone
should be played directly.
Max Loop VM Msg
Indicates whether the standard voicemail instructions

should be played, or whether the tone should be
played directly once the maximum number of loops
has been exceded.
Direct VM
Msg Timeout
Time to wait after message has played to timeout

and/or repeat the message if no entry pressed.
Msg Play
Number of times to play the recorded message if the

caller does not press any options and it times out.
Error Re-tries
Number of times to play invalid options and repeat the

message upon receiving an undefined option.
Find users in the Company

Directory by
Specifies whether the first three letters entered by a

user using the directory will look up by first name, last
name or both.
Announce Extension
When transferring a call as a results of a directory

lookup, announces the transfer extension if this option
is enabled. Please hold while I transfer you to
extension xxx
Operator Extension
When a user selects a 0 while engaging the directory,

the user is transferred to this number. The number
may transfer to an extension, ring group, external
number, etc.
Extension of fax machine

for receiving faxes
The extension number of the device to receive faxes

when faxes are detected and explicit routing does not
provide for fax termination.
Email address to have

faxes emailed to
The email address to which faxes are sent if the

Extension of fax machine for receiving faxes is set to
system
Email address that faxes

appear to come from
The email address from which faxes appear to have

been sent, when Extension of fax machine for
receiving faxes is set to system
Country Indications
Configures the Com.X PBX country


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Field
Content
24-hour format
Configures whether 12-hour AM/PM or 24-hour format

should be employed by the Com.X PBX
Allow Anonymous Inbound

SIP Calls?
Enabling this setting disables authentication checks on

incoming SIP requests. I.e. anyone can place a SIP
call to this PBX.
Check for updates
Allow the system to automatically check the internet

for FreePBX and Asterisk version updates. Updates of
the FreePBX and Asterisk modules are managed by
Far South Networks for the Comma and Com.X
product ranges. This field should be left blank.
Update email
An email address to send notifications of available

updates to. Updates of the FreePBX and Asterisk
modules are managed by Far South Networks for the
Comma and Com.X product ranges. This field should
be left blank.
Table 31: General PBX configuration settings

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4.7 IVR
The Interactive Voice Response feature allows the creation and chaining together of
interactive voice announcements and corresponding menu selections to facilitate
user call transfer in the Com.X system.

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Field
Content
Change Name
Changes the name used to identify this IVR from the default
(Unnamed). This entry will be displayed in selection lists that
support IVR.
Announcement
The audio message to be played to the user when this IVR is

accessed. The message typically would present the user with a
list of options and corresponding digit(s) to activate the option.
Timeout
The number of seconds without user response before this IVR

times out. When timed out, if a destination with option 't' has
been defined, this destination would be transferred to on
timeout.
Enable directory
If enabled, this allows callers currently being serviced by this

IVR to dial the Directory feature code (default #) to access the
directory. Please see 4.5 for more information on feature
codes.
VM Return to IVR
When enabled, should one of the IVR options result in the

user's call being transferred to a voicemail box, after leaving a
message or exiting the voicemail menu, the user is returned to
this IVR.
Directory Context
If directory access is enabled, access the directory specified

here if the user uses the Directory feature code.
Enable Direct Dial
If enabled, users can dial extensions directly from within the

IVR.
Note: Having direct dialing enabled introduces a delay before
the IVR responds to menu options while the system gives the
user a chance to dial more digits re: dialing an extension.
Note:If enabled and some of the configured extensions start
with the same number as one of the configured IVR options,
the option might be inaccessible as the system will be
collecting digits in order to dial an extension, as opposed to
servicing IVR options. Rather disable direct dial in your main
IVR and include an option for direct dialing (e.g. 'If you know
the number of the extension you wish to dial, please press 8)
which leads to a second IVR with a return option, and direct
dial enabled, from which the caller can then dial the desired
extension.
Loop before t-dest If enabled, the IVR loops as described in the 'Repeat loops
field' when user selection times out.
Timeout message
If specified, this audio message is played when user selection

times out.

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Field
Content
Loop before i-dest If enabled, the IVR loops as described for the 'Repeat loops
field' when user a user selects an invalid option.
Invalid message
If specified, this audio message is played when the user

selection is invalid.
Repeat loops
The number of times the IVR should be played if the user does
not provide an option before timeout, or if the option is invalid.
The IVR will only loop on invalid input if the 'Loop before i-dest'
option is enabled. The IVR will only loop on timeout if the 'Loop
before t-dest' option is enabled.
If the number of repeats has been reached and another timeout
or invalid selection occurs, the relevant timeout 't' or invalid 'i'
destination is used. If these destinations are not configured, the
call terminates.
Destinations
For each valid selection in this IVR, enter the digit(s) to activate
the selection and the call destination to transfer the caller to.
A special 't' destination is used to configure a destination on
user selection timeout.
A special 'i' destination is used to configure a destination on
invalid user selection.
For each destination, if the Return to IVR option is enabled, the
destination will be ignored and the user returned to the parent
IVR of this IVR.
Note: It is possible that the last IVR was not the transfer directly
before this IVR, but some steps removed. The Return to IVR option
will cause the call to return to the last previous IVR, regardless of
how many non-IVR transfers may have taken place in the interim.
Table 32: Interactive voice response configuration

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4.8 Music on hold

The Music on hold feature allows for the upload of audio files (e.g. music,
announcements, advertisements, informationals, etc.) in either WAV or MP3 format
for playback to users when calls are on hold.
Note: When mp3 files are uploaded, the files are internally converted to wav format.
This process can take as long as the total length of the mp3, and utilize a high
amount of CPU resources. We recommend that mp3 upload is performed during
scheduled maintenance.
Music may be categorized and music from a specific category played for specific
contexts, e.b. when a user is waiting in a support queue.
4.8.1. Configuring streaming music

To configure music on hold to stream from a media server, select Music on hold from
the Advanced menu, and then add a streaming category.
Please see section 6.15 for details on configuring a streaming music server.
Enter a name for the stream, and configure the music player command as follows,
using the URL to the streaming server:
/usr/bin/mpg123 -q -s --mono -r 8000 -f 16384 -b 100 <url>
The -s switch redirects the music stream for PBX use

The --mono switch transforms the stream to 1 channel audio as required
The -f switch adjusts the audio volume. You can also try -f 8192 (softer) or -f 32768
(louder)
The -b switch configures an audio buffer of the specified amount of Kbytes. If your
connection to the streaming server suffers from delays, consider adjusting this value.
Setting this to a high value might result in a delay before streaming commences.
<url> represents the URL to the streaming server, including any protocol and port
indications.
Use the optional -@ switch if the streaming server is publishing a playlist (e.g. .m3u)
instead of directly streaming the audio (e.g. as in the case with the GNU MP3
daemon).
Submit and apply the configuration. This streaming category can now be used where
music on hold selection is possible (e.g. queues).
For inbound calls you can set the music on hold class from the inbound route's
advanced context menu.
Warning: If you incorrectly configure the application entry above, applying might
result in the apply freezing up. To fix this, restart the Com.X, correct the application
entry, or remove the streaming music on hold category and apply.
4.8.2. Playback configuration

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Field
Content
Upload
The audio file to upload. Supported formats are: WAV, MP3
Volume
This value is linear, but human sound perception is logarithmic.

As a result we recommend that you test the volume configured
here for comfort and quality levels. This feature will convert
MP3 files to WAV files. If you do not have mpg123 installed,
you can set the parameter: AMPMPG123=false in your
/etc/amportal.conf file.
Enable random
play
Enabling random play will select audio files to play in a random

order.
If this is disabled, files are played in the order listed here.
Table 33: Playback configuration
4.8.3. Adding a music category

Music categories categorize the audio files available on the Com.X PBX server
storage.
Field
Content
Category name
A unique name for this audio category
Table 34: Music category configuration
4.8.4. Adding a streaming category

Streaming categories allow the inclusion of internet audio streams for features that
support Music on hold.

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Field
Content
Category name
A unique name for this streaming

category.
Application
This is the "application=" line used to

provide the streaming details to Asterisk.
Please see information on the
musiconhold.conf configuration for
different audio and internet streaming
source options.
Optional format
An optional "format=" line that is used to

provide the audio format. This should be
a format understood by Asterisk such as
alaw, and is specific to the streaming
application you are using.
Please see information on the
musiconhold.conf configuration for
different audio and internet streaming
source options.
Table 35: Streaming music category configuration
4.9 Outbound routes

This feature allows advanced access to outbounds routes and should only be used
when the Outbound Routes panel does not provide sufficient levels of detail or
flexibility given the administration need.

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Field
Content
Route name
A unique name for this route.
Route password
An optional password. If present, users

will be prompted to enter this password
before call routing using this route is
allowed.
Note: this is a single password for the
route. Pin sets should be assigned
using the Pin set field.
Pin set
If more than one pin should allow access

to this route, leave the Route password
field empty and specify a pin set here.
Emergency Dialing
When enabled, the CID field of the caller

will be overridden by the calling device's
emergency CID value (if configured).
Intra company route
If enabled, the internal caller ID

information will be retained, and not
replaced by the extension's external CID
nor the route's external CID.
Music on hold
If specified, the audio file will be played

before routing the call. This allows
country-specific or destination-specific
messages or instructions to be played
before placing the outbound call.
Dial patterns
This route will only route calls if it is the

highest priority outbound route that
matches the destination number to one
of the dial patterns specified here.
For more information on dial patterns,
please see 3.10.4.2.
Trunk sequence
This prioritized list specifies the order in

which trunks should be utilized to route
calls. the first trunk in this list is
attempted, failing which the second trunk
is attempted, etc.
Table 36: Outbound route configuration
4.10 Paging
Paging allows groups of extensions to be created that will receive calls (provided the
phones for the specified extensions allow paging).

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Field
Content
Paging extension
The extension to dial to activate the page
Group description
A custom description for this group
Device list
The list of devices and extensions to

include in this paging group
Force if busy
If enabled, the system will not check

whether the extension is busy before
paging it. Active calls may be interrupted.
This is useful for emergency pages.
Duplex
When enabled, all supporting devices will

allow two-way communication to all other
extensions and devices in this list. I.e.
any audio generated by any device will
be audible to all other devices and
extensions in the device list.
Default page group
If enabled, this function removes the

default flag from any previous default
paging group. If set, an extension, when
created, may be specified to join the
default paging group, resulting in it
automatically being added to this group
and automatically being deleted from this
group if such an extension is deleted.
Table 37: Paging configuration
4.11 Intercom
The intercom feature allows for a call to be placed to phones supporting the feature,
and for such phones to automatically answer the call and play the audio on speaker.
To enable intercom (disabled by default), navigate to the GUI, then Start
Configuration Advanced and Feature Codes.
Enable the Intercom Prefix feature code and change the feature code to a code that
will not conflict with existing codes (e.g. *93). Apply.
The list of phones known to support Intercom and Paging include: Aastra,
Grandstream, Linksys/Sipura, Mitel, Polycom, SNOM, and possibly other SIP
phones. ATAs are not supported. If your phone is set to auto-answer, it should also
support intercom.
The phone must be configured to support the feature. On a Snom for example,
configuration would involve turning on Auto Connect Indication, setting the Auto
Connect Type to Handsfree and enabling the Intercom feature.
Please contact your phone manufacturer or consult the phone manuals for specifics
on configuring your phone and model for intercom support.

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4.12 Parking Lot

Callers can be placed in a parking with a time out associated, after which the caller is
transferred to the specified destination. This allows receptionist, for example, to place
a caller on hold while waiting for their desired extension to become available. The
caller in the parking lot would then, after the parked timeout, automatically ring
reception again, by which time the desired extension might have freed up.
To transfer a call to the parking lot, use a transfer feature and dial extension 70. The
allocated slot will be announced. To retrieve the call, dial the slot number.
Field
Content
Enable parking lot

feature
Call parking will only be available if this feature is enabled.
Parking lot
extension
The extension to transfer calls to in order to park them (the

default extension number is 70)
Number of slots
The number of parking bays available. Parking lots are

numbered as follows: Parking lot extension + bay number, e.g.
71, 72, etc.
Parking timeout
The number of seconds after which the call is removed from the
parking bay. The call is then transferred back to the extension
from which the call was transferred into the parking bay.
Parking lot context The dialplan context for parked calls. Changing this setting
might result in parked calls no longer functioning.
Parking Alert-Info
Additional information to be added to the call when the call is

transferred to the destination for orphaned calls. Phones
supporting SIP Alert information could then inform the user
receiving an orphaned call of the call's orphaned status by
displaying a message or playing a different ring-tone.
Caller ID pre-pend Additional information to be pre-pended to the CID when the

call is transferred to the destination for orphaned calls. This
allows a user receiving an orphaned call to be informed of the
call's orphaned status.
Announcement
An announcement to be played to the caller in the parking lot

when the call is flagged as orphaned, before transferring the
call to the orphaned call destination.
Destination
The destination for calls that become orphaned. Please see

section 3.11.4 for more information on call destinations.
Table 38: Parking lot configuration
4.13 Phonebook
The phonebook feature allows names and speed dialing codes to be matched with
extension and outbound numbers. It allows a user to dial numbers by phoning the
phonebook feature code (please see section 4.5 for more detail on feature codes)
and entering the first three letters of a desired phonebook entry's name, after which

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the call is transferred to the desired extension or an outbound call is placed with the
desired number.This feature also enables lookups of users in the phonebook
directory using the directory feature code (#).
Field
Content
Name
The name to look the first three letters up from.
Number
The extension or outbound number to be dialed for this entry.
Speed dial code
A speed dial code to associate with this phonebook entry.
Set speed dial
The speed dial entry will be added to the database if this field is
enabled.
File
An optional CSV format file for uploading multiple phone book

entries. Entries should be in the following format:
Name;number;speeddial
If large phonebook databases (> 2000 entries) are to be
uploaded, additional configuration is required. Please see the
Com.X Troubleshooting guide for more detail large phonebook
imports.
Note: If the delimiter is something other than ';' the import will
fail. Please see your operating system / application help for
configuring ';' as the CSV export delimiter.
Table 39: Phone book entry configuration options
4.14 Pin Sets

Pin Sets are a convenient way to group sets of pins to facilitate multiple users with
individual pins gaining access to features, without using a common pin for the
features.
Field
Content
Description
Custom description for this Pin set. This will be listed in

selection lists that allow Pin sets.
Record in CDR
If enabled, when this Pin set is used in call routing, the Call
Detail Record includes the Pin set details.
Pin list
The list of pins included in this Pin set.
Table 40: Pin set configuration

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4.15 Queues
Queues are a powerful feature of the Com.X system, allowing callers to be placed on
hold and transferred according to a number of strategies and based on priority.
Callers waiting in the queue are provided with their position in the queue, projected
waiting time and the option to use self-service menus to exit the queue and transfer
to other destinations.
Agents servicing queues are allowed to log in by dialing the queue number followed
by *, and log out out of queues by dialing the queue number followed by **, are
provided with information re: callers waiting in the queue, and queues can be
configured in flexible manners to transfer calls to agents.

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Field
Content
Queue number
The extension number of the queue. Any calls or transfers to

this extension will be routed to the queue.
Queue name
A unique name for the queue.
Queue password
An optional password. If set, agents will need to log in with this

password before they can service the queue.
CID name prefix
An optional prefix to be added to the CID of callers in the

queue. This prefix will be displayed to agents servicing the
queue.
Wait time prefix
If set to yes, prefixes the CID name of all callers in the queue
with the total time, in minutes, for which they were in the queue.
Alert info
If configured, additional SIP INFO messages carry the string

configured here to the agent phones, facilitating distinctive ring
and other features.
Static agents
A list of agents who are permanently servicing the queue. Static

agents do not need to log in to the queue and can not log out of
the queue. The list can contain extensions as well as external
numbers, provided there is an outbound route that can support
a call to the external number.
Extension quick
pick
An ease-of-use drop-down list of extensions to allow quick

addition of extensions to the list of static agents in the queue.
Agent
announcement
An announcement to be played to the agent prior to connecting

the agent and the caller.
Join
announcement
An announcement to be played to callers upon their placement

in the queue.
Music on hold
class
The category of music on hold to be played to callers while

waiting in the queue.
Join empty
If set to Yes, callers can be added to the queue even when no

agents are logged in to service the queue.If set to No, callers
may not be added to the queue if there are no agents logged in.
If set to Strict, callers may not be added to the queue if there
are no agents logged in, or if all of the agents are unavailable.
Leave when empty If set to Yes, callers will be removed from the queue if there are
no longer any agents logged in. If set to strict, callers will be
removed from the queue if there are no agents logged in or if all
of the agents are unavailable.
Ring strategy
Ringall: Rings all the available agents simultaneously until one

answers.
Roundrobbin:Takes turns ringing each available agent.

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Leastrecent: Rings the agent who was least recently called by
this queue.
Fewestcalls: Rings the agent who has serviced the fewest calls
from this queue.
Random: Rings an agent at random.
Rrmemory: Round robbin with memory. Rings each agent
sequentially but remembers the position when the call is
answered and rings the next agent in the sequence to service
the next call.
Agent timeout
The number of seconds for which an agents extension will ring

before it is considered a timeout.
Retry
The number of seconds to wait before attempting to dial agents

again. If set to No Retry, then caller will be transferred to the
failover destination after just one round of dialing the agents.
Wrap up time
After a succesful call, this is the number of seconds to wait

before attempting to transfer another call to a potentially free
agent.
Call recording
The format in which calls are to be recorded, if recording of

queue calls is required.
Event when called If set to Yes, a call will trigger the AgentCalled,
AgentDump,AgentConnect and AgentComplete manager (AMI)
events.
Member status
If set to Yes, the QueueMemberStatus manager(AMI)event will

be generated.
Skip busy agents
If set to Yes, agents on occupied extensions will be skipped as

if the line were returning a busy signal. This prevents the call
waiting feature from being presented with a call and the next
agent is attempted instead.
Queue weight
Allows prioritising of queues so that calls waiting in a high

priority queue will deliver it's calls first if there are agents
servicing more than one queue.
Autofill
If not enabled, calls in the queue are processed in

series,requiring all calls to be held until the first caller in the
queue is connected to an agent. If enabled, calls can be
processed in parallel.
Agent regex filter
Provides an optional regex expression that will be applied

against the agent callbacknumber. If the callback number does
not pass the regex filter then it will be treated as invalid. This
can be used to restrict agents to extensions within a range, not
allow callbacks to include keys like *, or any other use that may
be appropriate.

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Frequency
The frequency with which their position and estimated hold time
is announced to the caller.
Announce position If set to Yes, the annoucements will tell the caller their position
in the queue.
Announce hold
time
If set to Yes, the announcements inform the user of the

expected time before their call is connected to an agent.
IVR break out

menu
An option to include an existing IVR as a break out menu. This

allows callers to select options from the IVR, removing them
from the queue and directing their calls to the destination
specified by the IVR option.
Repeat frequency
The frequency with which the optional IVR menu is announced.

(0 seconds to disable announcements)
Destination
The destination for any failover calls (i.e. Calls that are
removed from the queue without being answered by an agent).
Table 41: Queue configuration options
4.16 Ring Groups

Ring groups allow groups of extensions to be notified of incoming calls using a
number of powerful ring strategies. A rich set of ring group parameters allow flexible
configuration of timeout, alerts, music on hold, pick-up confirmation and other
features.

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Field
Content
Ring group
number
The number a caller will dial to ring extensions in this ring

group.
Group description
A descriptive title for this ring group.
Ring strategy
The strategy with which to ring extensions in this ring group.

Options include:
Ringall: rings all available extensions in the ring group
simultaneously.
Hunt: Takes turns ringing each available extension in the ring
group.
Memoryhunt: Rings the first extension in the list. Then rings the
first and the seond extension in the list. Then the first, second
and third extension in the list. And so on...
Firstavailable: Rings only the first available extension in the list.
Firstnotonphone: Rings only the first channel not off hook,
ignoring call waiting features.
*Prim: The patterns with this suffix work as above, except if the
first extension in the list is occupied, then the other extensions
will not be rung. If the first extension has Do Not Disturb (DND)
enabled, it will not be rung. If the first extension has
unconditional call forwarding enabled, then all of the extensions
in the list will ring.
Ring time
The time, in seconds, for which each extension in the list will
ring.
Extension list
The list of extensions to be included in this ring group. External

numbers may be included if suffixed by a #.
If an internal extension is suffixed by a #, then the follow-me of
that extension will be used for the call.
Extension quick
pick
An ease-of-use drop-down list of extensions to allow quick

addition of extensions to the list of static agents in the queue.
Announcement
An optional message to be played to the caller before

attempting to ring the extension in the list.
Play music on
hold?
If a music category is selected, then the caller will hear that

music while the calls is attempted rather than the traditional
ringing.
CID name prefix
An optional feature to prefix the CID of all callers calling this

group. This allows agents to differentiate between calls to the
ring group and calls to their extension directly.
Alert info
If configured, additional SIP INFO messages carry the string

configured here to the agent phones, facilitating distinctive ring

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and other features.
Ignore CF settings When checked, agents who attempt to use call forward will be
ignored.
Skip busy agent
If checked, agents on an occupied line will be skipped as if the

line were returning an unavialble signal. This prevents calls
form being presented to call waiting features.
Confirm calls
If enabled, agents will have to confirm, by pressing the number

1 button on their handset, before answering a call. This
prevents calls from being connected to voicemail or similar
features on external extensions. This feature is only available
when the ringall ring strategy is selected.
Remote announce A message to be played to the agent receiving the call if the
Confirm calls feature is enabled.
Too late announce A message to be played to agents who have answered a call,
but were too slow in confirming the call. i.e. Another agent
answered and confirmed before the initial agent confirmed.
Destination if no
answer
The destination to which to send unanswered calls. This can

include extensions, voicemail boxes, IVR's, announcements,
queues, phonebook entries and terminations.
Table 42: Ring group configuration oprions
4.17 System Recordings

Audio can be recorded and used in features such as announcements, IVR, etc.
Recordings can be uploaded and made from an extension. Follow the instructions on
the system recordings page.
Note: Remember to press the # key to indicate the end of the recording before
giving it a name.
Field
Content
Extension number If an extension number is entered here, this number will be

used to record a recording should the Save Recording feature
code (please see section 4.5 for more detail on feature codes)
be dialed from that extension.
Upload
A recording file in PCM Encoded format, 16 Bits, at 8000Hz,

single channel. Recordings are uploaded to
/var/lib/asterisk/sounds/custom/
Name of this
recording
A custom name for the recording. This name will appear in

selection lists for features that make use of recordings.
Table 43: System recording configuration


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4.18 Time Conditions

Time conditions check the current Com.X system time and compares it with the
specified Time Group to determine whether there is a match. The call is then routed
either to a day-night entry if this over-ride has been enabled, or to the match
destination if the current time falls within the time block specified in the Time Group,
or to the no match destination otherwise.
Field
Content
Name
Name used to identify this time condition in selection lists

that allow time conditions.
Time group
The Time Group that specifies the time range that results
in a match for this time condition. If the current system
time does not match this time group, the call will be routed
to the no match destination. Please see 4.19 for detail on
Time Groups.
Associate with
If a day-night feature code (e.g. 283) is specified here,

and the day/night entry is enabled, this time condition will
route directly to the day/night entry, ignoring the time
condition settings specified here. Disabling the day/night
entry will cause calls to be routed according to the time
condition settings.
Destination if matches
Destination to route the call to if the Time Group

conditions are matched.
Destination it no match
Destination to route the call to if the Time Group

conditions are not matched.
Table 44: Time condition configuration

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4.19 Time Groups

Time groups allows the specification of blocks of time for use in call routing rules,
such as time conditions. Please see 4.18 for more detail on Time Conditions.
Field
Content
Name
A name to identify this time group with. This will appear in

selection lists that allow time groups.
Time to start
The start time for the block of time this time group specifies.
Time to finish
The end time for the block of time this time group specifies.
Week day start
The start day for the block of time this time group specifies.
Week day finish
The end day for the block of time this time group specifies.
Month day start
The start day in the month for the block of time this time group
specifies.
Month day finish
The end day in the month for the block of time this time group
specifies.
Month start
The start month for the block of time this time group specifies.
Month finish
The end month for the block of time this time group specifies.
Table 45: Time group configuration

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4.20 Voicemail Blasting

Voicemail blasting allows messages to be added to a number of voicemail boxes at
once.
Field
Content
Voicemail blast
number
The extension to dial to leave a message to this voicemail

blasting group.
Group description
Custom description for this voicemail blasting group.
Audio label
Select the audio recording or announcement to play before the

voicemail prompt for this voicemail blasting group.A unique
announcement serves to confirm for a user that they have
dialed the correct voicemail blasting group.
Password
An optional password required to authenticate before the

voicemails are delivered.
Voice mailbox list
Select (using Ctrl-click) all the extensions that should receive

the voicemail message in their mailboxes.
Default voicemail
blast group
If enabled, this function removes the default flag from any

previous default voicemail blasting group. If set, an extension,
when created, may be specified to join the default voicemail
blast group, resulting in it automatically being added to this
group and automatically being deleted from this group if such
an extension is deleted.
Table 46: Voicemail blasting configuration
4.21 DISA
A DISA destination allows for an outbound dial tone to be configured as a destination
on the unit. This could be used to allow particular external callers to make outbound
calls through the outbound routes on the Com.X.
Field
Content
DISA Name
An appropriate name for the DISA destination.
Pin
The user will be prompted to enter this pin before a call can
be placed from the DISA tone destination. Mulitiple pins
can be entered, with each separated by a comma.
Response Timeout
The time, in seconds, for which the DISA is available

before hanging up. If an incomplete or invalid number has
been entered, the call will hang up after this duration.
Digit Timeout
The maximum time, in seconds, that the system will wait

between digits being entered. If digits are entered and this
time limit elapses, a call will be attempted using only the

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entered digits.
Require Confirmation
Optional. Requests the user to confirm by pressing '1'

before requesting the password for the DISA destination.
Caller ID
The outbound caller id that the call will appear to have

come from.
Context
The context of the calls generated from this DISA. It is not

advisable to edit this unless you have great confidence in
your Freepbx knowlege.
Allow Hangup
If enabled, a call can be hung up using the feature code **

while in call. This will present the user with dialtone again,
and allow another outbound call.
4.22 Listening in on calls / Spy feature

The Com.X PBX allows a spy feature to be enabled that allows the listener to listen in
on calls in progress. Input from the listener will not be audible to the callers on the
call being listened in to.
To listen in on all calls, enter the ChanSpy feature code. The first active channel will
be listened in on. To cycle through the channels, use the * key. To increase or
decrease the volume of the intercepted audio, use the # key.

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5 Monitoring the system

A rich suite of monitoring functions are available from the Com.X GUI. Monitoring
domains include Call Detail Records, PBX system monitoring and events,
Configuration details, Host system monitoring, Telephony status and events and GUI
web server status.
5.1 GUI Status Page

The status page of the Com.X GUI offers vast monitoring features, as well as
displaying the configuration status of the Com.X.
5.1.1. Status Display

The GUI offers a display to indicate the overall status of the system configuration and
synchronization. This display lists any configurations that are outstanding, as well as
actions required to apply the configuration and set the status to synchronized.
Figure 96: Configuration and synchronization status display
5.1.2. Services menu

A list of services and their status is offered on this page. From this list, services can
be configured to restart and be enabled/disabled and enabled/disabled on start up.
To restart or enable/disable a service, right-click and select the appropriate action.
Figure 97: Services can be enabled/disabled, restarted and enabled/disabled on start

up.
Note: Services that are essential to operation can not be disabled in real time, or at
start up.
Changing the status of services should not normally be necessary. Please ensure
that you understand the consequences of disabling or restarting a service before
using these functions.
5.2 Detailed logging

The Com.X supports a variety of logging levels, as detailed below:

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Log level
Description
debug
Source code level debug reports. Enabling this level results in a large
volume of logging data.
notice
Informative messages that report on system, channel, span and call

state.
warning
Informational messages requiring attention and evaluation. Warnings

may indiacte conditions that may limit or in future potentially impede
functionality.
error
Error reports indicating items that should be addresses should system

functionality be affected.
verbose
Additional informational messages (similar to notices), allowing more

effective call tracing.
Table 47: Logging levels
Logging takes place on the CLI as well as to /var/log/asterisk/messages.

From the Com.X GUI, two levels of detail can be selected, namely brief or
verbose. These may be accessed using Monitor -> Asterisk -> Set log level
<Brief/Verbose>
5.3 Call Detail Records (CDR)

5.3.1. Reviewing Call Detail Records
The Com.X records call detail records for all calls. To access these records, select
Call Log from the Reports menu. Specify the period for which records should be
retrieved and select Search.
Figure 98: Call record search options
For each record, the date, channel, call source, caller ID, destination, the disposition
and duration of the call is logged.An additional UUID field is provided for each call,
containing the unique identifier, or a recording of the call, if one exists.
Note: If the outbound CID on a trunk has been configured with an override (or as
'hidden'), the override CID will be stored in the caller ID field, not the original CID, as
well as in the Source field (empty if the override is 'hidden').
If the extension number or a specific account code should rather be stored in the
CDR when the trunk CID has been over-ridden, configure the Account code field

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on each extension to an appropriate value. The account code is then stored in the
CDR, while the upstream telco is presented with the CID as configured on the trunk.
A results summary is displayed at the bottom of the list, where navigation of the list is
also facilitated. CDR records are stored in a database on the Com.X and can be
backed up. Please see 2.3.1.1.4 for more detail on backing up the CDR database.
Call reports can also be exported to Portable Document Format (PDF) or in Comma
Separated Values (CSV) format for further processing.
Figure 99: CDRs can be exported in CSV or PDF format.
5.3.2. Enabling logging to CSV

To enable logging of all CDRs to /var/log/asterisk/cdr-csv/Master.csv, uncomment the
'enable=yes' line in /etc/asterisk/cdr.conf. Please see section 6.1.2.3 for details re:
editing system files.
5.3.3. Accessing the CDR database

To access the records in the MySQL CDR database remotely, the Com.X needs to
be configured to allow an external user access.
1. Configure the Com.X to allow remote connections to the MySQL server by
editing /etc/mysql/my.cnf and changing the bind-address field to 0.0.0.0
Note: This poses a security risk and remote access should be restricted by a proper
firewall.
2. Restart the MySQL server:
sudo /etc/init.d/mysql restart
3. Restart the comma-gui service:

sudo /etc/init.d/commagui restart
4. Log into the mysql server as root from the Com.X console:
mysql -u root
5. Configure a cdruser with read access to the CDR database:

GNT SELECT ON asteriskcdrdb.* TO cdruser@'%' IDENTIFIED BY 'password';
FLUSH PRIVILEGES;
6. ConfigurX your SQL client to point to the server, with username cdruser and
the password as configured, on the default port 3306, and with database
schema asteriskcdrdb.

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5.4 Monitoring of remaining capacity on the unit

The Com.X provides monitoring of the available space on the hard drive, SSD or SD
card.Space on the device is consumed as new calls are recorded and the recordings
are saved. The Com.X monitors the storage and sends a notification email (see
section 3.1) once the space consumed in exceeds a particular threshold (default
80%). A reminder email is then sent repeatedly until such time as the threshold is no
longer exceeded.
5.4.1. Deleting of call recordings

Once an email has been received informing the administrator of the exceeded
capacity threshold, the administrator can delete any call recordings that are not
needed.
Function to be added in upcoming release.
5.4.2. Changing the memory consumption threshold

Should a consumption threshold of 80% be unsatisfactory, the threshold percentage
can be altered.
Function to be added in upcoming release.
5.4.3. Auto-removal of audio files

If the stored data exceeds a threshold of 97% capacity, an email is sent to notify the
administrator and audio files are automatically deleted to prevent loss of service due
to storage device overflow. The oldest recordings are deleted first and recordings are
deleted until the consumed disk space is less than 96% of capacity.
5.5 Retrieving monitoring information

To access monitoring functions, select Monitoring from the main menu bar, and
select the category and function of interest. Results are displayed in tabular format,
with the ability to page to more results. In some cases the information is color coded
to facilitate easier analysis. The reload button re-issues the monitoring query,
facilitating ad-hoc snapshots of monitoring data.
Note: Due to a known issue with the tabular representation of results, if you are only
seeing one page, manually enter 2 in the page number at the bottom of the table
and press enter. All pages of information should then become visible.
5.5.1. Monitoring functions

Category
Function
Description
Asterisk Events
Asterisk event log (/var/log/asterisk/event_log)
Asterisk Messages
Asterisk messages log (/var/log/messages)
Asterisk Queues
Asterisk queue log (/var/log/queue_log)
Channels
Asterisk CLI (show channels). Lists channels
Asterisk

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Category
Function
Description
in use and active calls.
Database
Displays the PBX configuration database
Set log level brief
Enable logging of notices, warnings and

errors only
Set log level verbose Enable logging of DTMF, verbose messages,

notices, warnings and errors
Configuration
DHCP Configuration
Dahdi system
Dahdi span configuration

(/etc/dahdi/system.conf)
Dahdi channels
Dahdi channel configuration

(/etc/asterisk/chan_dahdi.conf)
Day/Night
Extensions
Extension configuration file

(/etc/asterisk/extensions.conf)
IAX peers
IAX peer configuration file

(/etc/asterisk/iax_additional.conf)
IAX registrations
IAX registrations configuration file

(/etc/asterisk/iax_registrations.conf)
Network interfaces
System nerwork interface configuration file

(/etc/network/interfaces)
SIP peers
SIP peer configuration file

/etc/asterisk/sip_additional.conf
SIP registrations
SIP registrations configuration file

(/etc/asterisk/sip_registrations.conf)
mISDN ports
mISDN configuration file

(/etc/asterisk/misdn.conf)
mISDN configuration mISDN stack configuration file

(/etc/mISDN.conf)
Reload
Reload Asterisk
Reloads the Asterisk PBX. Results in reload

of SIP, IAX, dahdi and mISDN. Complete
service loss during reload.

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Category
Function
Description
Reload IAX
Reloads the IAX module.
Reload SIP
Reloads the SIP module. Causes SIP trunks

and extensions to re-register.
Reload mISDN
Reloads the mISDN module. Service to BRI

channels will be lost while reloading.
Reload Dahdi
Reloads the dahdi module. Service to

analogue and PRI channels will be lost while
reloading.
Apache
Restart the web server
Asterisk and drivers
Restart the PBX software as well as DAHDI

and mISDN drivers
MySQL
Restart the database server
Audit
Performs a system audit and present items

requiring attention to the user
Disk usage
Displays system mount points, their capacity

and usage statistics.
Network status
System network interface status (ifconfig)

Displays interfaces, whether they are Up and
their Rx, Tx and error statistics. Also shows IP
configuration for each interface.
Performance
Displays system performance snapshot,

including CPU performance, active processes,
memory usage and page file usage.
System log
Displays the system-wide log. This includes

information, warning and error messages from
system-wide applications and services
(/var/log/syslog)
Dahdi channels
Asterisk CLI (dahdi show channels) Displays

the list of channels visible to the telephony
driver.
Dahdi spans status
Asterisk CLI (dahdi show status) Displays the

current dahdi span status. If spans are in error
here, there may be a configuration problem,
Restart
System
Telephony

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Category
Function
Description
or a Comma device might not be reachable.
Dahdi statistics
Displays statistics related to the dahdi

dynamic spans, including slips and skips. If
these numbers do not remain constant, it
might indicate network performance issues
that could result in a decrease in call quality.
Device discovery
Displays the list of Comma devices visible to

the Comma Manager on all networks,
including their ethernet and IP configuration
status.
IAX peers
Asterisk CLI (iax2 show peers). Displays IAX

peers that have registered with the PBX and
their status.
IAX registry
Asterisk CLI (iax2 show registry). Displays the

PBX IAX registry. IAX trunks should be listed
here when registered with IAX providers.
Manager log
Displays the Comma Manager log. This is a

useful resource for finding problems with
Comma devices booting and channel events.
SIP peers
Asterisk CLI (sip show peers). Displays SIP

peers that have registered with the PBX and
their status.
SIP registry
Asterisk CLI (sip show registry). Displays the

PBX SIP registry. SIP trunks should be listed
here when registered with SIP providers.
misdn channels
Asterisk CLI (misdh show channels) Displays

available BRI channels
misdn stack status
Asterisk CLI (misdn show stacks) Displays the

ISDN BRI software stack status, including link
enumeration, TE/NT configuration, point to
point and multipoint configuration and status.
Access
Log of web server access events

(/var/log/apache2/access.log)
Errors
Log of web server error events

(/var/log/apache2/error.log)
Client Log
Debugging information, warnings and errors

generated by the GWT-based GUI running in
the browser.
Web Server
GUI

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Table 48: Monitoring functions

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6 Advanced Administration
6.1 Console access
Note: Any customisation or third party integration not explicitly endorsed and
supported by Far South Networks will result in the software warranty becoming void.
6.1.1. Authentication
Default username: comma
Default password: farsouth
It is advisable to change the authentication information from the default if the unit is
accessible over the network via secure shell.
6.1.2. Accessing the unit

On power-on the unit can be accessed either via serial cable or via a secure shell
using ssh if the IP address is known from a DHCP server.
A serial connection can also be used to obtain the IP address allocated by a DHCP
server.
6.1.2.1 Serial access
Connect a serial cable to the unit's serial port and configure a terminal emulator such
as putty with the following settings: 115200,8,N,1
The cable should be a null modem serial cable, with pin 2 (RxD) and pin 3 (TxD)
crossed and pin 5 (ground signal) connected to pin 5 (ground signal).
Note: Com.X10 and Com.X5 units feature serial ports with Rj45 connections.
The DHCP-assigned IP address can be obtained from the serial console using the
ifconfig command, or from the external DHCP server's administration application.
Figure 100: "ifconfig eth0" shows network configurations of eth0 (Lan1)
6.1.2.2 Secure shell access

Secure shell login is supported on the default SSH port 22:
The unit can be accessed using ssh on any of the external LAN ports using their
default or configured IP addresses).

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6.1.2.3 Editing system files
To edit a file, use the 'jed <filename>' file editor. If superuser privileges are required
to edit the file, use 'sudo jed <filename>'.
6.1.2.4 Changing the shell login password
Log in as the comma user. To change the password, type: 'passwd'. Enter and
confirm the new password.
6.2 Rebooting the system

The system can safely be rebooted using the sudo reboot command. Stopping
Asterisk using the 'sudo asterisk -r stop when convenient ' command, is recommended.
This stops the Asterisk service as soon as no calls are in progress and ensures that
rebooting the system does not affect any active calls.:
Figure 101: 'sudo asterisk -rx "stop when convenient" ' stops the asterisk service, so
that no calls are cut off, then 'sudo reboot' reboots the Com.X system.
6.3 Configuring email services

6.3.1. Configuring voicemail to email
To enable the voicemail to email service on an extension, add an email address and
a pin to the email and pin fields in the advanced configuration options of the
extension. Voicemail recordings will be sent to the address when they are created,

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Figure 102: If an email address is listed in this field,

voicemail recordings and notifications will be sent to the
address.
6.3.2. Configuring sendmail for mail relay

Note: Mail relay configuration is not backed up by the Com.X backup utility.
The Com.X product range makes use of sendmail for outbound SMTP mail delivery.
The sendmail server will attempt to directly resolve destination mail server addresses
and contact these servers directly for mail delivery.
If your network design requires the use of a mail relay server for mail delivery,
sendmail can be configured to deliver outbound mail into the relay server's queue. An
authentication username and password can also be configured.
The mail relay server may be configured to only allow emails from a specific domain,
in which case the sendmail server could need to be configured to masquerade as
originating email from that domain.
From the Start menu, select Configure and General settings. Enter the mail relay
server host, username and password and masquerading domain in the appropriate
fields and apply.
Note: The relay server's hostname must be entered, not its IP. If the hostname is
not known, create an entry called 'mailrelay' in /etc/hosts with the IP and use
'mailrelay' as the hostname:
xxx.xxx.xxx.xxx
mailrelay
mailrelay.local

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Figure 103: Sendmail configuration
6.3.3. Sending a test email

You can send a test email by using the 'mail' command. The first parameter of the
mail command is the destination email address. After entering a subject, enter the
message body. A '.' on an empty line signals the end of the message.
For example:
comma@comx1:~$ mail info@farsouthnet.com
Subject: Test email
This is a test email.
This is line 2.
.
Cc: test-user@farsouthnet.com
6.3.4. Monitoring outgoing email

Each outgoing email attempt and the result of the delivery attempt is recorded in the
sendmail log files. To view, from the command line, use one of the following
commands:
less /var/log/mail.info
tail /var/log/mail.info
(navigate through the log)
(list the last few lines)
tail -f /var/log/mail.info
(real-time monitoring)
6.4 Asterisk PBX software management

6.4.1. Accessing the Asterisk PBX console
Log in using ssh as described in 6.1.2.2. From the console, type:

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sudo asterisk -vvvr
'sudo' executes the command with root privileges.

'asterisk' launches the CLI (command line interface) for the PBX software subsystem.
'-vvvr' indicates three levels of verbosity (v) enabled, and remote console (r) access.
6.4.2. Stopping and restarting the Asterisk PBX software

For certain maintenance tasks it may be necessary to temporarily stop or restart the
AsterisPBX software.
6.4.2.1 Restarting Asterisk
To restart the Asterisk PBX software:
sudo /etc/init.d/asterisk restart
6.4.2.2 Stopping Asterisk

To stop the Asterisk PBX software, waiting until there are no longer any active calls,
but allowing new calls in the interim:
sudo asterisk -rx stop when convenient
To stop the Asterisk PBX software, waiting until there are no longer any active calls,
without allowing new calls in the interim:
sudo asterisk -rx stop gracefully
To stop the Asterisk PBX software immediately:

sudo /etc/init.d/asterisk stop
6.4.2.3 Starting Asterisk

To start the Asterisk PBX software:
sudo /etc/init.d/asterisk start
6.4.2.4 Notify when no calls

In some situations you may want to wait until there are no calls on the sytem before
performing some action. The script below will print the message 0 active channels
when no calls are active:
#!/bin/bash
while [ 1 -lt 2 ]; do
printf "."
sudo asterisk -rx "core show channels" | grep -x "0 active calls"
sleep 1
done
6.5 FreePBX access

Though we strongly recommend using the Com.X GUI for all configuration tasks,
occasional advanced configuration might be required that necessitate the use of
FreePBX.

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To access the FreePBX configuration GUI, point your browser at:
http://<ip-of-pbx>/freepbx
The login username is 'admin' and the password is the same as the password
configured for the Com.X GUI.
6.6 Manually backing up and restoring the configuration

Though we strongly recommend using the Com.X GUI for backup and restore,
conditions may arise where the GUI is unavailable and a backup or restore is
required to be manually executed. To do so:
sudo /usr/share/freepbx/modules/backup/bin/ampbackup.pl "backupname" yes yes yes yes no
The five flags indicate whether the following components should be backed up, listed
in order from left to right as presented on the command line:
voicemail, system recordings, system configuration, CDR, operator panel
To confirm successful backup or to restore a backup manually, you can view the
backups list on the FreePBX restore page:
http://<host>/freepbx/config.php?type=tool&display=backup&action=restore
Alternatively you can view the list of backups from the command shell:
sudo ls /var/lib/asterisk/backups
6.7 Setting the system's date, time and timezone

The Com.X unit tries to obtain date and time from an NTP server on the internet.
Failing this, or in the case where the date / time discrepancy on a unit is large, the
date and time can be configured as follows:
1. Log into the system as described in 6.1.
2. If the system is unable to contact an internet NTP server to derive time from,
date and time might be presented using UTC (GMT+0) which may not be
desirable.
3. To reconfigure the system's timezone, issue the following command and
select the correct settings:
sudo dpkg-reconfigure tzdata
4. Set the date / time using the 'sudo hwclock' command and the format
mm/dd/ccyy hh:mm:ss
5. For example, to set the date to 10 Mar 2011 and the time to 18:12:
sudo hwclock --set --date="3/10/2011 18:12"
6. Propagate the hardware clock setting to the running operating system:

sudo hwclock --hctosys
7. Confirm that both the hardware clock and the operating system time matches:

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sudo hwclock
sudo date
8. Ensure that the commamgr is logging using the correct time. Warning: this
will disrupt service on all iTAs and analogue interfaces:
sudo /etc/init.d/commamgr restart
6.8 Video call support (H.263 and H.263+)

The Com.X platform may be configured to support video calls. Care should be taken
to evaluate potential performance impacts of video codec translation on the platform
in question.
To enable video support:
Add the following to the [general] section of /etc/asterisk/sip.conf:
videosupport=yes
In /etc/asterisk/sip_custom.conf, on extensions supporting video, add h263 and h263p to the

supported codec list.
Restart the PBX.
Ensure that all phones that take part in video calls have these codecs included in
their codec list.
6.9 Recovering / resetting the Com.X comma user password

While GRUB is loading, press escape and select recovery mode and select Drop to
the root shell prompt.
If the system was booted from a USB disk, mount the hard drive as follows:
mount /dev/sda1 /mnt
chroot /mnt
Change the comma user password as below:

passwd comma
Enter the desired password.

reboot
6.10 Adding static routes to the network

It may be desirable to add static routes for routing of specific subnets via specific
gateways or over specific interfaces.
Applying networking changes from the Com.X GUI over-writes the

/etc/network/interfaces file. Any changes made to this file may be lost after
an apply.
Rather, static routes should be placed in /etc/network/if-up.d/static-routes. This
script will be run when-ever an IP interface comes up. The

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interface that triggered the script is available via the IFACE

environment variable.
For example:
#!/bin/bash
if [ "$IFACE" == "eth0" ]; then
sudo ip route add x.x.x.x via y.y.y.y
fi
if [ "$IFACE" == "eth1" ]; then
sudo ip route add x.x.x.x/24 dev eth1
fi
Ensure the script is executable:

sudo chmod +x /etc/network/if-up.d/static-routes
Restart the interface or alternatively the Com.X:

sudo ifdown eth<N> && sudo ifup eth<N>
Note: It is not recommended to have more than one default route

in the Com.X routing table.
6.11 Booting the Com.X in recovery mode
6.11.1. Booting in recovery mode
Using the serial console cable, while GRUB is loading, press escape and select
recovery mode and select Drop to the root shell prompt.
From the root recovery shell, perform the necessary tasks, such as scanning and
fixing hard drive problems, or resetting the comma user password.
Once completed, reboot the system, enter the BIOS and remove the USB drive from
the boot priority list.
6.12 Value Added Applications

6.12.1. ATMOS Voice Logger
Com.X 1.3 software has been integrated to offer a cloud based voice logging
solution. The ATMOS voice logger encrypts recordings and securely uploads them to
a remote cloud server for centralised security and management.
Integrity of the recording is guaranteed, and recordings archived in this manner are
meet requirements to be presented in legal proceedings.
For information on purchasing a license and subscription, please contact your
distributor.
Product collateral on the Call Centre Analyser is available at:

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Product brochure: http://www.farsouthnet.com/wp-content/uploads/2011/06/FSNVoice_Logger-Brochure.pdf
User Manual: http://www.farsouthnet.com/wp-content/uploads/2011/06/com.XAtmos-Voice-Logger-Setup-Procedure-1-0-2.pdf
6.12.2. Call Centre Analyser

Com.X Call Center Analyser is a flexible, real-time call queue monitoring package
that allows monitoring of a call centre's efficacy by presenting call queue data and
comprehensive reports on a customisable set of call queue statistics.
Queue data is available presented graphically, arranaged in csv or pdf format, and
via real-time monitoring tools and via comprehensive scheduled email reports.
For information on purchasing a Call Centre Analyser license, please contact your
distributor.
Product collateral on the Call Centre Analyser is available at:
Product brochure: http://www.farsouthnet.com/wp-content/uploads/2011/06/Com.XCall-Center-Analyser.pdf
User Guide: http://www.farsouthnet.com/wp-content/uploads/2011/06/Call-CenterAnalyser-User-Guide-1.01.pdf
Figure 104: Call Centre Analyser access available through the

Com.X GUI log in page.
6.13 Third Party Modules

Please note that Far South Networks does not actively support third party
modules and installations.
Note: third party configurations and data may not be backed up by the Com.X
backup utility.
6.14 Telephone Management Systems (TMS)

6.14.1. MAN 3000
6.14.1.1 Installing MAN 3000 support
The Com.X supports logging of call records to a MAN 3000 buffer across ethernet.

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To enable support for MAN 3000 on a Com.X unit, from the shell, install the
MAN3000 support package:
sudo aptitude install commaman3000
For Man3000 releases prior to 11.02:

Install the TCPIP2MAN3000 software on the MAN3000 Windows platform from:
C:\Program files\Man3000\App Feature Installation\Install_TCPIP2MAN3000.exe
Run the TCPIP2MAN3000 software and configure it with the IP address of the
Com.X PBX, using port 8989
For Man3000 releases post 11.02:
Activate the Man3000 TCP/IP configuration by enabling Soft Online Authentication,
or your Man3000 USB software dongle:
Figure 105: Man3000 settings menu

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Figure 106: Man3000 port settings
Configure the TCP/IP settings with the IP address of the Com.X PBX:
Figure 107: Man 3000 TCP/IP configuration
Installing the PBX templates:


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Download the Far South Networks MAN3000 PBX format template from the Far
South Networks website at http://farsouthnet.com/support/technical-notes/(right-click
Save Target As, or Save Link As)
Alternatively, download the formats directly (using a downloader such as wget) from:
http://archive.farsouthnet.com/support/fsnInbound.PBX
http://archive.farsouthnet.com/support/fsnOutbound.PBX
Configure the MAN3000 software to use the Far South Networks Inbound PBX
format for PABX format 1 and Outbound PBX format for PABX format 2:
Figure 108: Man3000 format settings

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Figure 109: Importing a Man3000 format template

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6.14.1.2 CDR format
Records are streamed to MAN 3000 in the following format:
Column: 1 2
Type / Line Type / Line Date
Time
Channel
Source
Destination
Bill
Account Code
Table 49: Man3000 CDR format column headers

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Field
Content
Type / Line
In the first column, a value of 'I' indicates that the record is for
an incoming call. Any other value indicates the line number as
defined in /etc/commagui/man3000.patterns for an outgoing
call, or 9 in case the channel for the outgoing call was not
pattern matched.
Type / Line
In the second column, a value of 'O' indicates that the record is

for an outbound call or 9 as the line number for an incoming
call.
This process cannot correctly cater for all conditions, since the
specific configuration details of any given system is variable.
The outbound pattern match requires the following conditions in
order to match a call as outbound:
The call context must be from-internal
The destination may not be a local extension
The destination may not include Local
It may not be a follow-me call or internal transfer.
If any of these conditions are not met, the call is flagged as

internal.
Configuring a line number pattern match as described in
section 6.14.1.3 overrides the checks above and is the
recommended way of ensuring that outbound calls are correctly
classified.
Date
The date of the start of call (including channel access / call

setup / ringing) in the following format: ccyy/mm/dd
Time
The time of the start of call (including channel access / call

setup / ringing) in the following format: hh:mm:ss
Channel
The channel from which the call originated in the inbound case
or over which the call was placed in the outbound case. The
values of this field will be pattern matched against the entries in
/etc/commagui/man3000.patterns to determine the line number.
If a match cannot be made, the line number is set to 9.
Source
The originating number for the call. This can be the CID of the
caller for an incoming call, or the CID (extension number) of the
caller for an outgoing call.
Note: If the CID on the trunk used to place the outgoing call
overrides the caller's CID (e.g. extension number), the trunk's
override CID will be logged here.
Destination
The destination number of the call in the outbound case, or the

destination extension or service of the call in the inbound case.
Bill
The billable duration of the call in seconds, excluding call setup


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Field
Content
(ring) time.
Account Code
The account code assigned to the Com.X, FlexPath or

extension (defaults to 0000).
Table 50: Man3000 CDR format field descriptions
Example CDR output:

9 I 2010/05/07 06:36:23 n/a
9905
*97
37
9 I 2010/05/07 06:56:11 n/a
9906
*98
13
0000
9 I 2010/05/07 06:56:38 n/a
9906
*98
0000
9 I 2010/05/07 06:56:43 n/a
9906
*98
0000
2 I 2010/05/07 06:59:15 SIP/9905-00000e9 9902

9 I 2010/05/07 07:12:39 n/a
9930
9905
*97
0000
18
20
0000
0000
2 I 2010/05/07 07:20:46 SIP/9905-00000e9 9906
9905
0000
2 I 2010/05/07 07:39:04 SIP/9901-00000e9 9921
9901
22
0000
O 3 2010/05/07 07:39:52 mISDN/1-u3809
01154XXXXX
2 I 2010/05/07 07:40:23 SIP/9901-00000e9 9921
01266YYYYY
9901
12
26
0000
0000
3 I 2010/05/07 07:42:08 mISDN/1-u3813
01154XXXXX
02159ZZZZZ
37
0000
3 I 2010/05/07 07:42:24 mISDN/2-u3818
01154XXXXX
01231AAAAA
25
0000
6.14.1.3 Defining outbound line numbers

By default all outbound calls will be labeled with line number 9 to indicate that the
outbound line is unknown to the Com.X. One-digit line numbers (0-8) can be
assigned to outbound channels for recognition in the MAN 3000 system.
This assignment is done by creating a /etc/commagui/man3000.patterns file in the
Com.X file system. This file can be created on a separate system or laptop and
copied onto the Com.X as follows:
scp man3000.patterns comma@<ip-of-com.x>:/etc/commagui/
The man3000.patterns file contains entries of the following format:

pattern=line
For example:
mISDN/=1
Zap/=2
SIP/hookd=3
Pattern matching matches the <pattern> if it is part of the outgoing call record
channel field. If more than one entry matches, the last entry matched determines the
line number. The same line number may be assigned to multiple patterns.

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6.14.1.4 Logging records from a specific date
The Com.X remembers the last record logged and will only log records once new
records have become available. To configure the Com.X to start logging records
again at some date in the past, modify the /var/lib/commaman3000/lastdate file
located in the Com.X file system. This file can be created on a separate system or
laptop and copied onto the Com.X as follows:
scp lastdate comma@<ip-of-com.x>:/var/lib/commaman3000/
The format of lastdate is as follows: ccyy/mm/dd hh:mm:ss

For example:
2010/05/04 10:15:34
In this example, all calls placed after 10:15 and 35 seconds on 4 May 2010 will be
streamed to MAN3000.
6.14.2. Asterisk TMS systems

The Com.X may support TMS systems that are compatible with Asterisk.
We strongly recommend installing third party TMS software on a separate machine,
accessing the Com.X CDR files and database with read-only access.
Third party TMS systems have access to Com.X CDR records via one of the
following tools:
After installation of the commaman3000 package, CDRs are served on the

port described in section 6.14.1.
By default the commaman3000 server module will only output new CDR
entries every 10 seconds. To disable this back-off and deliver data on every
request, edit the file /usr/bin/commaman3000srv and remove the sleep 10 statement at
the end of the file
Downloading and parsing the Master.csv file (please see section 5.3.2)
Accessing the CDR database (please see section 5.3.3)

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6.15 Setting up a streaming music server

Though setting up a streaming server is dependent on the software and hardware
platform selected, we include an example configuration here that is compatible with
Ubuntu linux, using the free GNU mp3 daemon for your convenience.
Download the GNU MP3 streamer from http://www.gnu.org/software/gnump3d/
The .tar.gz file can be downloaded from:
http://ftp.cc.uoc.gr/mirrors/nongnu.org/gnump3d/gnump3d-3.0.tar.gz
Alternatively download from:

http://archive.farsouthnet.com/support/tools/gnump3d-3.0.tar.gz
Extract the archive and install the software:

tar xvfz ../gnump3d-3.0.tar.gz
sudo make install
Configure the music location, port and other options by editing the following
configuration file, then run the server:
sudo vi /etc/gnump3d/gnump3d.conf
sudo /usr/bin/gnump3d
Configure the music collection on the streaming server by pointing your browser to:
http://<ip>:<port>
Configure a streaming category as in section 4.8.1:

/usr/bin/mpg123 -q -s --mono -r 8000 -f 8192 -b 100 -@ http://<ip>:<port>/recurse.m3u
Warning: If the above application entry is incorrectly configured, applying might

result in the apply freezing up. To fix this, restart the Com.X, correct the application
entry, or remove the streaming music on hold category and apply.

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6.16 Microsoft Outlook integration

The Com.X product range is compatible with Microsoft Outlook by means of the
Outcall or asttapi plugins.
6.16.1. Outcall
Outcall supports only SIP extensions, but in addition to making outbound calls also
pops up contact details for incoming calls. Outcall has been tested with the Com.X
using Outlook 2000 and Outlook 2007.
Download the Outcall installer from http://code.google.com/p/outcall/downloads/list
Install the application and configure as follows:
Tab
Field
Value
Server
Server
IP address of Com.X
Server
Username
AMI username, see below
Server
Password
AMI password, see below
Server
Port
5038
Dialing Rules
Outgoing context
from-internal (or the flexpath context for class

of service configurations)
Table 51: Outcall configuration

Figure 110: Outcall dialing rules configuration

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Figure 111: Outcall server configuration
Add the extension to be used for the PC running outlook. This must be a SIP
extension:
Figure 112: Outcall extension configuration

Log in via the ssh shell onto the Com.X and create an AMI user account as follows
in /etc/asterisk/manager.d/outcall.conf:
[testuser]
secret = fsntestuser
deny=0.0.0.0/0.0.0.0
permit=192.168.0.0/255.255.255.0
read = call,user
write = call,user
Restart the manager module:


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sudo asterisk -rx "reload manager
In Outlook, select the contact to call and select the Outcall button. When a call is
received by the Com.X, the calling contact, or a new contact window should appear
in Outlook.
Figure 113: Outlook contacts integration
6.16.2. asttapi
asttapi allows dialing from both analogue and SIP extensions, but does not pop up
contact details on incoming calls like Outcall.
Download and install the asttapi setup program from
http://sourceforge.net/projects/asttapi/files/asttapi/
Select Start Control panel Phone and modem options
Select the Advanced tab and select 'Omnis TAPI Driver for Asterisk', then Configure
Figure 114: asttapi configuration

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Field
Value
Host
IP address of Com.X
Port
5038
User
AMI username, see below
Password
AMI password, see below
User Channel
The extension channel (e.g. SIP/8263 or

Zap/8)
Line name
The name of the line as displayed in Outlook

(e.g. Far South Networks Com.X)
Context
from-internal (or the flexpath context for class

of service configurations)
Called ID
The caller ID for the extension
Table 52: asttapi configuration
Apply the configuration. You may be required to restart Windows.

To place a call, right-click a contact and select Call contact and select Dialing
options.
Figure 115: Calling a contact
Figure 116: Selecting Asterisk as dial option
Select the correct line name. You may have to scroll down the list if the line is not
visible.

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6.17 fail2ban integration

Com.X systems are shipped with fail2ban installed and enabled.
fail2ban provides a comprehensive and flexible mechanism for preventing
unauthorized access to Com.X systems. The fail2ban application acts as a security
monitor and firewall manager, detecting multiple failed attempts to access a service
(such as ssh or SIP) from the same IP address and then blocks the address for a
configured period of time.
6.17.1. Configuration
The default configuration blocks the source IP address of 4 consecutive SSH access
failures for 10 minutes and 10 consecutive SIP failures for 1 hour:
Note: Ensure that fail2ban is running by checking the Services screen under the
Status tab.
6.18
ViBE
The ViBE application from Voipex addresses the common issue of limited ADSL
upstream bandwidth. With many small offices relying on an ADSL line for both data
and VOIP traffic, typically, the upstream link becomes a limiting factor in the number
of simultaneous calls the network can support.
The audio stream of a typical VOIP call is only 8kbit/s, however the protocol and
network overheads introduced during transmission can often exceed a further
34kbits/s bandwidth requirements.
The ViBE application creates a tunnel between two ViBE end points, significantly
reducing protocol overheads on multiple packets and dynamically optimizing voice
compresssion, resulting in a much higher permissible call density over a single ADSL
link.
Where a typical upstream ADSL link of 256kbits/s can support a maximum of 6
simultaneous G729 VOIP calls, a ViBE-enabled line of the same bandwidth can
support a maximum of approximately 28 similar simultaneous calls.
For further details on ViBE configuration and integration, please see the white paper
located at :
http://farsouthnet.com/wp-content/uploads/2011/05/Runway-X-ViBE.pdf
Note: The White paper referenced above describes the integration of a Runway
branded Com.X PBX with ViBE. The document is, however, relevant for Runway
and Com.X products.

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6.19 Flash operator panel

6.19.1. Flash Operator Panel demo version
Com.X units featuring 1.2 software include a demo version of the Flash operator
panel version 2 (FOP2), which shows graphically and dynamically the status of
extensions, trunks and advanced features.
From the operator panel, calls can be answered, transferred, originated, recorded
and ended, using simple clicks of a mouse.
The demo version of the Flash operator panel included as stock on Com.X products
allows for up to 15 extensions, trunks, queues or parking lots. By default, the panel
will show buttons for the first 15 such mechanisms configured, but not for any
configured thereafter. Should the full version of the Flash operator panel be required,
licenses can be purchased through Far South Networks.
6.19.2. FOP2 access

By default, full access to the Flash operator panel is granted to only the user of the
operator extension. This extension has access to the full suite of features on the flash
operator panel, regular users have access to a limited operator panel, containing only
some of the complete functionality.
Note: If no operator extension is configured, then the extension with the lowest
numerical DID will be assigned as the default operator extension for FOP2 access.
6.19.3. Setting the operator extension

The operator extension of the system is configured on the Com.X GUI, from
General Settings page, found in the advanced menu. The operator extension will
be permitted full access to the operator panel, whilst other extensions have limited, or
no access.
Figure 117: An Operator extension can be manually configured.
6.19.4. Limiting Flash Operator Panel access to operator only

On the General Settings list on the Setup page of the GUI, the flash operator
panel can be configured to allow (limited) access to users of all extensions, and full
access to the operator user or to allow access only to the operator extension. Setting
the FOP only available to Operator field to true will bar any other extensions from
accessing the flash operator panel at all, whilst setting this field to false will allow
full access to the operator user, and limited access to all other extensions. Select the
desired setting and then Review/Apply to activate the changes.

Page 200
Figure 118: FOP access can be limited to the operator only, if required.
6.19.5. Accessing Flash operator panel

The flash operator panel is accessed by selecting the Operator Panel icon on the
launch page of the GUI.
Figure 119: Select "Operator panel" to access the Flash

operator panel
A log in page will be presented. Enter the extension number and the associated
(voicemail) pin of that extension to gain access.
Figure 120: Login with an extension number and voicemail pin.

Page 201
6.19.6. Using FOP2

Once logged in to the Flash operator panel, the extension of the logged in user will
be displayed on the top left button, with the extension number in bold. Operator
extensions will be presented with more functionality options than regular users,
controlled by black button icons listed above the extensions. Hovering a cursor above
an icon displays a brief description of the function triggered by the button.
Figure 121: Extension 1000 logged in. Notice the nine icons above the
extensions each of which performs a function.
Figure 122: Extension 2000 is logged in on this operator panel. Notice only
four function buttons above the listed extensions.
Trunks, parking lot slots and queues are shown also displayed on the Flash operator
panel, and also offer visual indications of their busy state.
Figure 123: Trunks, queues and parking slots are also displayed. Notice the Sip Trunk
has one active call, indicated by the orange colour and the external caller, 5000.
6.19.7. Generating internal phone calls

To generate a call, first select the target extension, click on it so that the outline
becomes blue. Then click on the dial icon. Generating calls is a feature enabled for
operator and regular Flash operator

Page 202
panel users.
Figure 124: Dialling extension 2000 from extension 1000. Notice the blue outline
around extension 2000.
This first causes extension 1000 to ring, and upon a user answering, dials extension
2000 from extension 1000.
Figure 125: Notice the change of colour of extension 1000, as well as the incoming call
from "1000" shown on extension 2000.
6.19.8. Answering Calls

Incoming calls to other extensions can be answered through the Flash operator panel
by simply clicking on the ringing extension (again, notice the blue outline) and then
on the Call Pickup function button.
Figure 126: Picking up a call to extension 1003. Pushing the "Call Pickup" button will
connect the caller to the extension of the FOP user.
The call pickup feature on the Flash operator panel is available to operator and
regular FOP users.
6.19.9. Transferring calls

A Flash operator panel user is able to transfer an active call in which they are
involved, by selecting the target extension, while a call is active, and clicking on the
transfer feature button.
Figure 127: The call between extension 1000 and external caller 5000 will be transferred
to extension 1001.
A call can be transferred by operator and regular FOP users.


Page 203
The operator user is allowed further transfer capabilities with an option to transfer to
a voicemail box. The operator can transfer a call between themselves and another
party to the voicemail of any extension on which voicemail is enabled. To transfer to
the voicemail of an extension, simply select the target extension (the voicemail box of
which is going to receive the call) while a call is active and then click the Transfer to
mailbox function button.
6.19.10. Queue Management

6.19.10.1 Adding/removing dynamic agents
Any SIP extension can be added or removed from a queue by a FOP2 administrator
user. Simply click on the status indicator of the extension and select the add to
queue option.
Figure 128: Extension managemnet from FOP2 interface.
The dynamic agents serving a queue are displayed in the button for the queue.
Figure 129: The SIP/ extension shows the port hosting the dynamic
agent.
6.19.10.2 Picking up a call from a queue

Any user logged in to Fop2 has the ability to pick up a call from within a queue. The
user need not be an agent of the queue, and can select an individual call from within
the queue, based on the calling number.
Figure 130: Pickup particular calls in a queue.


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6.19.11. Listening to calls

Operator users are able to listen in on calls in which they are not involved.
6.19.11.1 Listen
The listen feature allows the operator to listen in on a call in which they are not
involved. To listen to call, select the local extension involved in the call (notice again,
the blue outline) and then click the Listen feature button.
Figure 131: Select the active call, then click "Listen"
After clicking Listen the operators extension will ring, and on answer will play the
audio of the selected call. The operator extension is muted and neither party of the
call is informed of the operator listening.
6.19.11.2 Listen and Whisper
The listen and whisper feature allows the operator to listen in on a call in which they
are not involved. Further,any audio from generated on the operator handset is
audible only to the internal extension, and not to the external party. To listen and
whisper, select the local extension involved in the call (notice again, the blue outline)
and then click the Listen and Whisper feature button.
Figure 132: The Listen and whisper feature allows the operator to listen in on other
calls, and talk to the local user on the call.
6.20 Nagios
The Nagios application provides an expandable and highly customisable platform for
monitoring of system services and network health. Com.X units are shipped with a
Nagios Core suite preconfigured to monitor the state of selected services and
interfaces on the system.
6.20.1. Hosts
Nagios can be configured to monitor network and service status on mulitple hosts.
The Nagios application on the Com.X system has the localhost defined as the only
host, however more hosts can be defined and monitored, if required.
Please consult the Nagios Core administrator guide for more details:
http://nagios.sourceforge.net/docs/nagioscore-3-en.pdf

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6.20.2. Accessing Nagios

The Nagios login portal is presented on the landing page of the Com.X GUI.
Point your browser to the IP of your Com.X and select the Monitor icon.
Enter the default username (nagiosadmin) and password (admin).
Figure 133: The landing page of the Com.X Gui.
6.20.3. Services
The installed Nagios Core package is preconfigured to monitor an assortment of
sevices and features as listed in Table 206. To access feedback on all monitored
services and click the Services button under the Current Status heading.

Page 206
Service
What is monitored?
BRI mISDN Port

Status
Provides the status of each BRI mISDN port on the system.

Including the link type (Point-to-point or point-to-mulitpoint),
the network or terminal status of the port, as well as the
status of the L1 and L2 links.
Current Load
Offers a report of the current, medium and long term load

averages of the system.
Current Users
Monitors the number of users logged in to the system.
Disk Space
Monitors the available disk space on the system and reports

when free space becomes critically low.
Ethernet LAN
Interfaces
Provides the status of each individual ethernet port on the

system. Indicates whether the port is enabled (up or down)
and whether or not a connection is active (running or not
running).
Ethernet Vibe
Interface
Pending Final Release.
Ethernet WAN
Interface
Pending Final Release.
FXO ports
The activity status of each FXO port is displayed

individually. This allows reporting on unused ports to be
disabled.
HTTP
Pending final release!
PRI Spans
Provides details on PRI Span status. If a PRI span is

reporting a fault, Nagios informs the user of the fault and
which span is reporting it.
SIP TRUNKS
Indvidually lists each SIP trunk and its connection status.
SSH
Pending final release!
Total Processes
Indicates the total number of processes running on the

system.
Table 53: Services and interfaces monitored by the Nagios application.
Services for which a warning is detected are highlighted in yellow, whilst services that
are in error are highlighted in red.Healthy services feature a green highlight.
6.20.4. Disabling active Nagios Checks on a Service or Port

Reporting on individual FXO, mISDN and Ethernet ports, allows for the disabling of
Nagios monitoring on unused ports. From the Services menu, click on the desire port
or service.

Page 207
Figure 134: Options available for configuring

Nagios monitoring on a port or a service.
Select Disable active checks of this service . This will prevent the Nagios
application from monitoring the service or port.
Thereafter, select Submit passive check result for this service and set the check
result to OK. This will set the status to OK until such time as active checks are reenabled and provide a different result.
6.20.5. Customised Monitoring and Reporting

Nagios offers an incredibly customisable platform. Scripts can be written to monitor
any thing of interest, and added to the Nagios application.
Please see the Nagios wesite: www.nagios.com or click the Documentation link on
the Nagios web GUI.
Illustration 3: The
Documentation link
provides a link to the
documentation page of the
nagios website.

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7 Troubleshooting
Please refer to the Com.X Troubleshooting Guide for trouble-shooting information.
This guide can be accessed from http://farsouthnet.com/support/

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Version 3.0.5, 09 January 2015

2010 2014 Far South Networks

Document ported from release 1.2 version 2.0.5

Include Ntework and Firewall administration.

Added 1.3 sip handset provisioning

Added faxing over IP

Added Fop2 usuage, Added Polycom handets and disoclaimer on Yealinki

2010 2015 Far South Networks