Table of Contents
Overview
This Application Note is a configuration guide for the Ribbon SBC (Session Border Controller) 1000/2000 when connecting to Cisco Unified Communication Manager (CUCM) and PlusNet SIP Trunk.
The configuration guide supports features outlined in the Microsoft Technet web page:
- For additional information on Cisco Platform, visit http://www.cisco.com.
- For additional information on Ribbon SBC 1000/2000, visit https://ribboncommunications.com/.
Introduction
Interoperability compliance testing focuses on verifying inbound and outbound call flows between Ribbon SBC 1000/2000 and Cisco CUCM.
Audience
This technical document provides telecommunications engineers with information for configuring both the Ribbon SBC and the third-party product. Procedures in this document require navigating third-party equipment as well as applying Ribbon SBC Command Line Interface (CLI) commands. To complete the configuration and perform any troubleshooting, the engineer performing the procedures must understand the basic concepts of TCP/UDP, IP/Routing, and SIP/RTP.
This Application Note is offered as a convenience to Ribbon customers. The specifications and information regarding the product in this document are subject to change without notice. All statements, information, and recommendations contained in this document are believed to be accurate but are presented without warranty of any kind, express or implied, and are provided “AS IS”. Users must take full responsibility for the application of the specifications and information contained here.
The links are only internal to Ribbon partners and employees. They do not work outside of the Ribbon Network.
Requirements
The following table lists the hardware and software used in the reference configuration.
Test Equipment and Software
| Vendor | Equipment | Software Version |
|---|---|---|
| Ribbon Networks | SBC 2000 | V8.0.2 |
| Third-party Vendor | ||
| Cisco | Cisco Unified CM Administration | 12.0.1.21900 |
Cisco | Cisco SIP Phone 7841 | sip78xx.11-7-1-17 |
| VentaFax | Fax Machine VentaFax | 7.6.243.616 |
Reference Configuration
The following figure serves as a topology for the reference configuration. The figure shows the connectivity between third-party equipment and the Ribbon SBC 1000/2000.
Reference Configuration Topology
Support
For questions about information in this document, contact Ribbon Support in either of the following ways:
- Global Support Assistance Center +1-978-614-8589 or +1-888-391-3434 (English language Support)
- Web: https://ribboncommunications.com/services/ribbon-support-portal-login
Verify License
The interoperability test described in this document requires no special licensing.
CUCM 12.0.1 Configuration
The following new configurations are included in this section:
1. Security Profile
Select System > Security > SIP Trunk Security Profile
SIP Trunk Security Profile
2. SIP Profile
Select Device > Device Settings > SIP Profile
SIP Profile
SIP Profile1
SIP Profile2
SIP Profile3
3. SIP Trunk
Select Device > Trunk > Add New
SIP Trunk
SIP Trunk1
SIP Trunk2
SIP Trunk3
4. Route Group
Select Call Routing > Route/Hunt > Route Group > Add New
Route Group
5. Route List
Select Call Routing > Route/Hunt > Route List > Add New
Route List
6. Route Pattern
Select Call Routing > Route/Hunt > Route Pattern > Add New
Route Pattern
Route Pattern1
Ribbon SBC 1000/2000 Configuration
The following configuration steps provide an example of how to configure the Ribbon SBC 1000/2000 to interoperate with Skype 2015 and Virgin Media SIP Trunk:
- SIP Profile
- SIP Server
- Media System
- Media Profiles
- Media List
- Remote Authorization Tables
- Signaling Groups
- Transformation
- Call Routing Table
1. SIP Profile
SIP Profiles control how the Ribbon SBC 1000/2000 communicates with SIP devices. The profiles control important characteristics such as session timers, SIP Header customization, SIP timers, MIME payloads, and option tags.
Select Settings > SIP > SIP Profiles to access the SIP Profile screen.
The following figures show the default SIP profile used for the Ribbon 1000/2000 used for this configuration effort.
PlusNet SIP Profile
CUCM 12.0.1 SIP Profile
Fax SIP Profile
2. SIP Server
SIP Server Tables contain information about the SIP devices connected to the Ribbon SBC 1000/2000.
Select Settings > SIP > SIP Server Tables to access the SIP Server Tables screen.
The entries in the tables provide information about the IP addresses, ports, and protocols used to communicate with each SIP server. The entries also contain links to counters that are useful for troubleshooting, as shown in the following figures.
PlusNet SIP Servers
CUCM SIP Server
Fax SIP Server
3. Media System
The Media System Configuration contains system-wide settings for the Media System. Configuring the media system means setting the number of RTP/RTCP port pairs and the starting port.
Select Settings > Media > Media System Configuration to access the Media System configuration screen.
Media System
4. Media Profiles
Media Profiles specify the individual voice and fax compression codecs and their associated settings for inclusion into a Media List. Different codecs provide varying levels of compression, allowing the reduction of bandwidth requirements at the expense of voice quality.
Select Settings > Media > Media Profiles.
The following figures illustrate possible media profiles of the voice codecs used for the SBC 1000/2000. The examples are for reference only.
PlusNet Media Profile
CUCM Media Profile
Fax Media Profile
5. Media List
The Media List shows the selected voice and fax compression codecs and their associated settings.
Select Settings > Media > Media List to access the Media List configuration screen.
PlusNet Media List
CUCM Media List
Fax Media List
6. Remote Authorization Tables
Remote Authorization Tables and their entries contain information used to respond to request message challenges by an upstream server. The Remote Authorization Tables on this page appear as options in Creating and Modifying Entries in the SIP Servers (For additional information about Remote Authorization Tables, see the Ribbon online SBC 1000/2000 documentation).
Select Settings > SIP > Remote Authorization Tables to access the Remote Authorization Tables configuration screen.
Remote Authorization Table
7. Signaling Groups
Signaling Groups allow telephony channels to be grouped for routing and shared configuration. These groups are the entity to which calls are routed, and the location from which Call Routes are selected. These are also the location from which Tone Tables and Action Sets are selected. In the case of SIP, Signaling Groups will specify protocol settings and links to server, media, and mapping tables.
Select Settings > Signaling Groups to access the Signaling Groups configuration screens.
PlusNet Signaling Group
CUCM Signaling Group
Fax Signaling Group
8. Transformation
Transformation Tables facilitate the conversion of names, numbers, and other fields when routing a call. For example, transformations can convert a public PSTN number into a private extension number, or a SIP address (URI). Every entry in a Call Routing Table requires a Transformation Table.
Select Settings > Transformation to access the Transformation configuration screen.
PlusNet Transformation
CUCM Transformation
Fax-Tenor Transformation
9. Call Routing Table
Call Routing allows calls to be carried between signaling groups, thus allowing calls to be carried between ports, and between protocols (like ISDN to SIP). Call Routing Tables define routes. The use of Call Routing Tables allows for flexible configuration of which calls will be carried, and also how the calls are translated. These tables are one of the central connection points of the system, linking Transformation Tables, Message translations, Cause Code Reroutes, Media Lists, and the three types of Signaling Groups (ISDN, SIP and, CAS).
Select Settings > Call Routing Table to access the Call Routing Table configuration screen.
PlusNet Call Routing
CUCM Call Routing
Fax Call Routing
Interoperability Test Results
The following table provides test results for interoperability compliance testing between Ribbon SBC 1000/2000 and CUCM
Interoperability Compliance Test Results
| Test Number | Test Scenario | Setup / Test Results | Status | Comment |
|---|---|---|---|---|
| 4.1 | Registration and authentication (registration mode) | The PBX is able to execute the correct resolution oft he DNS SRV record | Pass | |
| 4.3 | Basic call | With the basic call tests, the standard call scenarios and the CLIP/CLIR features are tested.
| Pass | |
| 4.3.1 | Normal call | Outgoing call from PBX to PSTN
| Pass | |
| 4.3.1.1 | Normal call | The geographic number in the PAI header corresponds to the location 11 of the user | Pass | Is it possible to configure several PAI per SIP trunk? Yes |
| 4.3.1.2 | Normal call | Display of A-number in B-party CLIP (national PSTN) | Pass | |
| 4.3.1.3 | Normal call | Display of A-number in B-party CLIP (international PSTN) | Pass | |
| 4.3.1.4 | Normal call | Display of A-number in B-party CLIP (mobile) | Pass | |
| 4.3.1.5 | Normal call | Call to mobile | Pass | |
| 4.3.1.6 | Normal call | Suppression of A-number => CUR | Pass | |
| 4.3.1.7 | Normal call | Outgoing call from analog extension | Pass | |
| 4.3.1.8 | Normal call | Outgoing call (> 5 min.) => PSTN | Pass | Hold connection for 5 minutes => RTP still correct? Yes |
| 4.3.2 | Normal call | Incoming call from PSTN (national) => PBX
| Pass | |
| 4.3.2.1 | Normal call | Display of A-number => CLIP | Pass | |
| 4.3.2.2 | Normal call | Incoming call from mobile => PBX | Pass | |
| 4.3.2.3 | Normal call | Suppression of A-number => CLIR | Pass | |
| 4.3.3 | Normal call | Two simultaneous outgoing/incoming calls | Pass | |
| 4.3.4 | Normal call | Enabled feature DND (do not disturb) | Pass | |
| 4.3.5 | Normal call | Test call with codec G.711 | Pass | |
| 4.3.6 | Normal call | Test call with codec G.722 (only SIP <=> SIP) | Fail | PlusNet didn´t support G.722 |
| 4.3.7 | Normal call | Test call with codec G.729 | Pass | |
| 4.3.2.8 | Clip No Screening | Outgoing call from PBX to PSTN
| Pass | |
| 4.3.2.8.1 | Clip No Screening | Despite Clip No Screening the geographic number in the PAI header corresponds to the location of the user | Pass | Is it possible to configure several PAI per SIP trunk? Yes |
| 4.3.2.8.2 | Clip No Screening | Display of A-number (NoSClip) at B-party (PSTN) | Pass | |
| 4.3.2.8.3 | Clip No Screening | Display of A-number (NoSClip) at B-party (international PSTN; depending on the destination carrier, the NoSClip telephone number may not be displayed in this case!) | Pass | |
| 4.3.2.8.3 | Clip No Screening | Call made from a PSTN line to an IP-PBX line with call forward to a line within the same IP-PBX, Answer Call.
| Pass | Does the IP-PBX has configuration settings to send SIP status 181 messages to the soft switch? Yes |
| 4.3.2.8.4 | Clip No Screening | Display of A-number (NoSClip) at B-party (mobile) | Pass | |
| 4.3.3.9 | Special call situations | Outgoing call PBX => PSTN
| Pass | |
| 4.3.3.10 | Special call situations | Outgoing call PBX=> PSTN
| Pass | |
| 4.3.3.11 | Special call situations | Outgoing call PBX => PSTN
| Pass | |
| 4.3.3.12 | Special call situations | Outgoing call PBX=> PSTN
| Pass | |
| 4.3.3.13 | Special call situations | Incoming call PSTN => PBX
| Pass | |
| 4.3.3.14 | Special call situations | Incoming call PSTN => PBX
| Pass | |
| 4.3.3.15 | Special call situations | Incoming call PSTN => PBX
| Pass | |
| 4.3.3.16 | Special call situations | Incoming call PSTN => PBX
| Pass | |
| 4.3.4.17 | Call clearing | Incoming / outgoing call; clearing after established call. Correct clearing on both sides
| Pass | |
| 4.3.4.18 | Call clearing | Interrupting the network connection of the SIP terminal device during a call
| Pass | |
| 4.4.19 | Hold | PBX => PSTN and PSTN => PBX
| Pass | |
| 4.4.19.1 | Hold | Putting an external call on hold in the PBX | Pass | |
| 4.4.19.2 | Hold | If applicable, MoH (music on hold) at A-party (PSTN) | Pass | |
| 4.4.19.3 | Hold | HOLD RETRIEVE: retrieving the external call | Pass | |
| 4.4.19.4 | Hold | Clearing the connection of the A-party while it is put on hold | Pass | |
| 4.4.20 | Hold | PBX => PSTN and PSTN => PBX
| Pass | |
| 4.4.20.1 | Hold | Putting an external call on hold in the PSTN | Pass | |
| 4.4.20.2 | Hold | If applicable, MoH (music on hold) at A-party (PBX) | Pass | |
| 4.4.20.3 | Hold | HOLD RETRIEVE: retrieving the external call | Pass | |
| 4.4.20.4 | Hold | Clearing the connection of the A-party while it is put on hold | Pass | |
| 4.5.21 | Call transfer | Internal call is transferred to external party: internal => PBX => external | Pass | |
| 4.5.21.1 | Call transfer | Call transfer from PBX party => PSTN party with announcement (attendant | Pass | |
| 4.5.21.2 | Call transfer | Call transfer from PBX party => PSTN party without announcement (blind transfer) | Pass | |
| 4.5.21.3 | Call transfer | Call transfer from PBX party => PSTN party without announcement (blind transfer)
| Pass | |
| 4.5.21.4 | Call transfer | Call transfer from PBX party => PSTN party without announcement (blind transfer)
| Pass | |
| 4.5.22 | Call transfer | Transferred call from external party to PBX: PSTN => PBX | Pass | |
| 4.5.22.1 | Call transfer | Call transfer from PSTN => PBX party with announcement (attendant transfer) | Pass | |
| 4.5.22.2 | Call transfer | Call transfer from PSTN => PBX party without announcement (blind transfer) | Pass | |
| 4.5.22.3 | Call transfer | Call transfer from PSTN => PBX party without announcement (blind transfer)
| Pass | |
| 4.5.22.4 | Call transfer | Call transfer from PSTN => PBX party without announcement (blind transfer)
| Pass | |
| 4.5.23 | Call transfer | Call from external party transferred to another external party: external => PBX => external | Pass | |
| 4.5.23.1 | Call transfer | PSTN => PBX party => PSTN with announcement (attendant transfer) | Pass | |
| 4.5.23.2 | Call transfer | PSTN => PBX party => PSTN without announcement (blind transfer) | Pass | |
| 4.5.23.3 | Call transfer | PSTN => PBX party => PSTN without announcement (blind transfer)
| Pass | |
| 4.5.23.4 | Call transfer | PSTN => PBX party => PSTN without announcement (blind transfer)
| Pass | |
| 4.6.24 | Call diversion | PBX party CFU to external party (PSTN) | Pass | |
| 4.6.24.1 | Call diversion | Internal call (CFU) => PSTN | Pass | |
| 4.6.24.2 | Call diversion | PSTN party busy, rejects call, does not answer | Pass | |
| 4.6.24.3 | Call diversion | A-party clears in ring phase | Pass | |
| 4.6.24.4 | Call diversion | External => PBX party (CFU) => PSTN | Pass | |
| 4.6.24.5 | Call diversion | PSTN party busy, rejects call, does not answer | Pass | |
| 4.6.24.6 | Call diversion | A-party clears in ring phase | Pass | |
| 4.6.25 | Call diversion | PBX party CFNR to external party | Pass | |
| 4.6.25.1 | Call diversion | Internal call (CFNR) => PSTN | Pass | |
| 4.6.25.2 | Call diversion | PSTN party busy, rejects call, does not answer | Pass | |
| 4.6.25.3 | Call diversion | A-party clears in ring phase | Pass | |
| 4.6.25.4 | Call diversion | External => PBX party (CFNR) => PSTN | Pass | |
| 4.6.25.5 | Call diversion | PSTN party busy, rejects call, does not answer | Pass | |
| 4.6.25.6 | Call diversion | A-party clears in ring phase | Pass | |
| 4.6.26 | Call diversion | PBX party CFB to external party | Pass | |
| 4.6.26.1 | Call diversion | Internal call (CFB) => PSTN | Pass | |
| 4.6.26.2 | Call diversion | PSTN party busy, rejects call, does not answer | Pass | |
| 4.6.26.3 | Call diversion | A-party clears in ring phase | Pass | |
| 4.6.26.4 | Call diversion | External => PBX party (CFB) => PSTN | Pass | |
| 4.6.26.5 | Call diversion | PSTN party busy, rejects call, does not answer | Pass | |
| 4.6.26.6 | Call diversion | A-party clears in ring phase | Pass | |
| 4.6.27 | Call diversion | External party (PSTN, mobile, etc.) CFU to PBX party: | Pass | |
| 4.6.27.1 | Call diversion | External (CFU) => PBX party | Pass | |
| 4.6.27.2 | Call diversion | PBX party busy, rejects call, does not answer | Pass | |
| 4.6.27.3 | Call diversion | A-party clears in ring phase | Pass | |
| 4.6.27.4 | Call diversion | External (CFNR) => PBX party | Pass | |
| 4.6.27.5 | Call diversion | PBX party busy, rejects call, does not answer | Pass | |
| 4.6.27.6 | Call diversion | A-party clears in ring phase | Pass | |
| 4.6.28 | Call diversion | External party (PSTN, mobile, etc.) CFB to PBX party: | Pass | |
| 4.6.28.1 | Call diversion | PBX party busy, rejects call, does not answer | Pass | |
| 4.6.28.2 | Call diversion | A-party clears in ring phase | Pass | |
| 4.6.29 | Call diversion | Call deflection: diversion during the ring phase | Pass | |
| 4.6.29.1.1 | Call diversion | Internal call PBX party CD => PBX party | Pass | |
| 4.6.29.1.2 | Call diversion | PBX party busy | Pass | |
| 4.6.29.1.3 | Call diversion | PBX party does not answer | Pass | |
| 4.6.29.1.4 | Call diversion | A-party clears in ring phase | Pass | |
| 4.6.29.2.1 | Call diversion | External call to PBX party CD => PBX party | Pass | |
| 4.6.29.2.2 | Call diversion | PBX party busy | Pass | |
| 4.6.29.2.3 | Call diversion | PBX party does not answer | Pass | |
| 4.6.29.2.4 | Call diversion | A-party clears in ring phase | Pass | |
| 4.6.29.3.1 | Call diversion | External call to PBX party CD => external PSTN party | Pass | |
| 4.6.29.3.2 | Call diversion | PSTN party busy | Pass | |
| 4.6.29.3.3 | Call diversion | PSTN party does not answer | Pass | |
| 4.6.29.3.4 | Call diversion | A-party clears in ring phase | Pass | |
| 4.7.30 | Call waiting | Incoming call during active internal call | Pass | |
| 4.7.30.1 | Call waiting | Call waiting tone | Pass | |
| 4.7.30.2 | Call waiting | Display of number of waiting party | Pass | |
| 4.7.30.3 | Call waiting | Acceptance of waiting call | Pass | |
| 4.7.30.4 | Call waiting | Putting the waiting call on hold | Pass | |
| 4.7.30.5 | Call waiting | Retrieve of waiting call | Pass | |
| 4.7.30.6 | Call waiting | Holding the 2nd call | Pass | |
| 4.7.30.7 | Call waiting | Terminating the active call | Pass | |
| 4.7.30.8 | Call waiting | Disregarding the waiting call | Pass | |
| 4.7.30.9 | Call waiting | Rejecting the waiting call | Pass | |
| 4.8.31 | 3-party conference | Establishing a conference according to the operating instructions of the PBX Internal - internal - external | Pass | |
| 4.8.31.1 | 3-party conference | Selecting a party (internal or external); 3rd party is put on hold
| Pass | |
| 4.8.32 | 3-party conference | Establishing a conference according to the operating instructions of the PBX | Pass | |
| 4.8.32.1 | 3-party conference | Selecting a party (external); 3rd party is put on hold
| Pass | |
| 4.9.33 | Pick up | Picking up a call from another extension of the PBX | Pass | |
| 4.10.34 | Call list | Entries in the call list
| Pass | |
| 4.10.35 | Call list | Call-back from call list
| Pass | |
| 4.11.36 | DTMF | DTMF support (G.711)
| Pass | |
| 4.11.37 | DTMF | DTMF support (G.729) | Pass | |
| 4.12.38 | Fax | Fax reception (G.711 only) | Pass | |
| 4.12.38.1 | Fax | Network-side T.38 re-invite rejected by PBX (response 488) or only G.711 codec is confirmed
| Pass | |
| 4.12.39 | Fax | Fax sending (G.711 only) | Pass | |
| 4.12.39.1 | Fax | Network-side T.38 re-invite rejected by PBX (response 488) or only G.711 codec is confirmed
| Pass | |
| 4.12.40 | Fax | Fax reception via T.38 | Pass | |
| 4.12.40.1 | Fax | Re-invite to T.38 by PBX or network | Pass | |
| 4.12.41 | Fax | Fax sending via T.38 (not possible in conjunction with the encryption option) | Pass | |
| 4.12.41.1 | Fax | Re-invite to T.38 by PBX or network
| Pass | |
| 5.42 | Redundancy | Test of redundancy:
| Pass | |
| 5.42.1 | Redundancy | 1. Register all available PBX systems the Plusnet SBC. | Pass |
Conclusion
This Application Notes document describes the steps required to configure the Ribbon SBC 1000/2000 to successfully interoperate with the Cisco CUCM and PlusNet SIP Trunk. All feature and serviceability test cases have been completed. The majority of test cases passed with noted exceptions and observations provided in Interoperability Test Results.
Table of Contents
Overview
This Application Note is a configuration guide for the Ribbon SBC (Session Border Controller) 1000/2000 when connecting to Cisco Unified Communication Manager (CUCM) and PlusNet SIP Trunk.
The configuration guide supports features outlined in the Microsoft Technet web page:
- For additional information on Cisco Platform, visit http://www.cisco.com.
- For additional information on Ribbon SBC 1000/2000, visit https://ribboncommunications.com/.
Introduction
Interoperability compliance testing focuses on verifying inbound and outbound call flows between Ribbon SBC 1000/2000 and Cisco CUCM.
Audience
This technical document provides telecommunications engineers with information for configuring both the Ribbon SBC and the third-party product. Procedures in this document require navigating third-party equipment as well as applying Ribbon SBC Command Line Interface (CLI) commands. To complete the configuration and perform any troubleshooting, the engineer performing the procedures must understand the basic concepts of TCP/UDP, IP/Routing, and SIP/RTP.
This Application Note is offered as a convenience to Ribbon customers. The specifications and information regarding the product in this document are subject to change without notice. All statements, information, and recommendations contained in this document are believed to be accurate but are presented without warranty of any kind, express or implied, and are provided “AS IS”. Users must take full responsibility for the application of the specifications and information contained here.
The links are only internal to Ribbon partners and employees. They do not work outside of the Ribbon Network.
Requirements
The following table lists the hardware and software used in the reference configuration.
Test Equipment and Software
| Vendor | Equipment | Software Version |
|---|---|---|
| Ribbon Networks | SBC 2000 | V8.0.2 |
| Third-party Vendor | ||
| Cisco | Cisco Unified CM Administration | 12.0.1.21900 |
Cisco | Cisco SIP Phone 7841 | sip78xx.11-7-1-17 |
| VentaFax | Fax Machine VentaFax | 7.6.243.616 |
Reference Configuration
The following figure serves as a topology for the reference configuration. The figure shows the connectivity between third-party equipment and the Ribbon SBC 1000/2000.
Reference Configuration Topology
Support
For questions about information in this document, contact Ribbon Support in either of the following ways:
- Global Support Assistance Center +1-978-614-8589 or +1-888-391-3434 (English language Support)
- Web: https://ribboncommunications.com/services/ribbon-support-portal-login
Verify License
The interoperability test described in this document requires no special licensing.
CUCM 12.0.1 Configuration
The following new configurations are included in this section:
1. Security Profile
Select System > Security > SIP Trunk Security Profile
SIP Trunk Security Profile
2. SIP Profile
Select Device > Device Settings > SIP Profile
SIP Profile
SIP Profile1
SIP Profile2
SIP Profile3
3. SIP Trunk
Select Device > Trunk > Add New
SIP Trunk
SIP Trunk1
SIP Trunk2
SIP Trunk3
4. Route Group
Select Call Routing > Route/Hunt > Route Group > Add New
Route Group
5. Route List
Select Call Routing > Route/Hunt > Route List > Add New
Route List
6. Route Pattern
Select Call Routing > Route/Hunt > Route Pattern > Add New
Route Pattern
Route Pattern1
Ribbon SBC 1000/2000 Configuration
The following configuration steps provide an example of how to configure the Ribbon SBC 1000/2000 to interoperate with Skype 2015 and Virgin Media SIP Trunk:
- SIP Profile
- SIP Server
- Media System
- Media Profiles
- Media List
- Remote Authorization Tables
- Signaling Groups
- Transformation
- Call Routing Table
1. SIP Profile
SIP Profiles control how the Ribbon SBC 1000/2000 communicates with SIP devices. The profiles control important characteristics such as session timers, SIP Header customization, SIP timers, MIME payloads, and option tags.
Select Settings > SIP > SIP Profiles to access the SIP Profile screen.
The following figures show the default SIP profile used for the Ribbon 1000/2000 used for this configuration effort.
PlusNet SIP Profile
CUCM 12.0.1 SIP Profile
Fax SIP Profile
2. SIP Server
SIP Server Tables contain information about the SIP devices connected to the Ribbon SBC 1000/2000.
Select Settings > SIP > SIP Server Tables to access the SIP Server Tables screen.
The entries in the tables provide information about the IP addresses, ports, and protocols used to communicate with each SIP server. The entries also contain links to counters that are useful for troubleshooting, as shown in the following figures.
PlusNet SIP Servers
CUCM SIP Server
Fax SIP Server
3. Media System
The Media System Configuration contains system-wide settings for the Media System. Configuring the media system means setting the number of RTP/RTCP port pairs and the starting port.
Select Settings > Media > Media System Configuration to access the Media System configuration screen.
Media System
4. Media Profiles
Media Profiles specify the individual voice and fax compression codecs and their associated settings for inclusion into a Media List. Different codecs provide varying levels of compression, allowing the reduction of bandwidth requirements at the expense of voice quality.
Select Settings > Media > Media Profiles.
The following figures illustrate possible media profiles of the voice codecs used for the SBC 1000/2000. The examples are for reference only.
PlusNet Media Profile
CUCM Media Profile
Fax Media Profile
5. Media List
The Media List shows the selected voice and fax compression codecs and their associated settings.
Select Settings > Media > Media List to access the Media List configuration screen.
PlusNet Media List
CUCM Media List
Fax Media List
6. Remote Authorization Tables
Remote Authorization Tables and their entries contain information used to respond to request message challenges by an upstream server. The Remote Authorization Tables on this page appear as options in Creating and Modifying Entries in the SIP Servers (For additional information about Remote Authorization Tables, see the Ribbon online SBC 1000/2000 documentation).
Select Settings > SIP > Remote Authorization Tables to access the Remote Authorization Tables configuration screen.
Remote Authorization Table
7. Signaling Groups
Signaling Groups allow telephony channels to be grouped for routing and shared configuration. These groups are the entity to which calls are routed, and the location from which Call Routes are selected. These are also the location from which Tone Tables and Action Sets are selected. In the case of SIP, Signaling Groups will specify protocol settings and links to server, media, and mapping tables.
Select Settings > Signaling Groups to access the Signaling Groups configuration screens.
PlusNet Signaling Group
CUCM Signaling Group
Fax Signaling Group
8. Transformation
Transformation Tables facilitate the conversion of names, numbers, and other fields when routing a call. For example, transformations can convert a public PSTN number into a private extension number, or a SIP address (URI). Every entry in a Call Routing Table requires a Transformation Table.
Select Settings > Transformation to access the Transformation configuration screen.
PlusNet Transformation
CUCM Transformation
Fax-Tenor Transformation
9. Call Routing Table
Call Routing allows calls to be carried between signaling groups, thus allowing calls to be carried between ports, and between protocols (like ISDN to SIP). Call Routing Tables define routes. The use of Call Routing Tables allows for flexible configuration of which calls will be carried, and also how the calls are translated. These tables are one of the central connection points of the system, linking Transformation Tables, Message translations, Cause Code Reroutes, Media Lists, and the three types of Signaling Groups (ISDN, SIP and, CAS).
Select Settings > Call Routing Table to access the Call Routing Table configuration screen.
PlusNet Call Routing
CUCM Call Routing
Fax Call Routing
Interoperability Test Results
The following table provides test results for interoperability compliance testing between Ribbon SBC 1000/2000 and CUCM
Interoperability Compliance Test Results
| Test Number | Test Scenario | Setup / Test Results | Status | Comment |
|---|---|---|---|---|
| 4.1 | Registration and authentication (registration mode) | The PBX is able to execute the correct resolution oft he DNS SRV record | Pass | |
| 4.3 | Basic call | With the basic call tests, the standard call scenarios and the CLIP/CLIR features are tested.
| Pass | |
| 4.3.1 | Normal call | Outgoing call from PBX to PSTN
| Pass | |
| 4.3.1.1 | Normal call | The geographic number in the PAI header corresponds to the location 11 of the user | Pass | Is it possible to configure several PAI per SIP trunk? Yes |
| 4.3.1.2 | Normal call | Display of A-number in B-party CLIP (national PSTN) | Pass | |
| 4.3.1.3 | Normal call | Display of A-number in B-party CLIP (international PSTN) | Pass | |
| 4.3.1.4 | Normal call | Display of A-number in B-party CLIP (mobile) | Pass | |
| 4.3.1.5 | Normal call | Call to mobile | Pass | |
| 4.3.1.6 | Normal call | Suppression of A-number => CUR | Pass | |
| 4.3.1.7 | Normal call | Outgoing call from analog extension | Pass | |
| 4.3.1.8 | Normal call | Outgoing call (> 5 min.) => PSTN | Pass | Hold connection for 5 minutes => RTP still correct? Yes |
| 4.3.2 | Normal call | Incoming call from PSTN (national) => PBX
| Pass | |
| 4.3.2.1 | Normal call | Display of A-number => CLIP | Pass | |
| 4.3.2.2 | Normal call | Incoming call from mobile => PBX | Pass | |
| 4.3.2.3 | Normal call | Suppression of A-number => CLIR | Pass | |
| 4.3.3 | Normal call | Two simultaneous outgoing/incoming calls | Pass | |
| 4.3.4 | Normal call | Enabled feature DND (do not disturb) | Pass | |
| 4.3.5 | Normal call | Test call with codec G.711 | Pass | |
| 4.3.6 | Normal call | Test call with codec G.722 (only SIP <=> SIP) | Fail | PlusNet didn´t support G.722 |
| 4.3.7 | Normal call | Test call with codec G.729 | Pass | |
| 4.3.2.8 | Clip No Screening | Outgoing call from PBX to PSTN
| Pass | |
| 4.3.2.8.1 | Clip No Screening | Despite Clip No Screening the geographic number in the PAI header corresponds to the location of the user | Pass | Is it possible to configure several PAI per SIP trunk? Yes |
| 4.3.2.8.2 | Clip No Screening | Display of A-number (NoSClip) at B-party (PSTN) | Pass | |
| 4.3.2.8.3 | Clip No Screening | Display of A-number (NoSClip) at B-party (international PSTN; depending on the destination carrier, the NoSClip telephone number may not be displayed in this case!) | Pass | |
| 4.3.2.8.3 | Clip No Screening | Call made from a PSTN line to an IP-PBX line with call forward to a line within the same IP-PBX, Answer Call.
| Pass | Does the IP-PBX has configuration settings to send SIP status 181 messages to the soft switch? Yes |
| 4.3.2.8.4 | Clip No Screening | Display of A-number (NoSClip) at B-party (mobile) | Pass | |
| 4.3.3.9 | Special call situations | Outgoing call PBX => PSTN
| Pass | |
| 4.3.3.10 | Special call situations | Outgoing call PBX=> PSTN
| Pass | |
| 4.3.3.11 | Special call situations | Outgoing call PBX => PSTN
| Pass | |
| 4.3.3.12 | Special call situations | Outgoing call PBX=> PSTN
| Pass | |
| 4.3.3.13 | Special call situations | Incoming call PSTN => PBX
| Pass | |
| 4.3.3.14 | Special call situations | Incoming call PSTN => PBX
| Pass | |
| 4.3.3.15 | Special call situations | Incoming call PSTN => PBX
| Pass | |
| 4.3.3.16 | Special call situations | Incoming call PSTN => PBX
| Pass | |
| 4.3.4.17 | Call clearing | Incoming / outgoing call; clearing after established call. Correct clearing on both sides
| Pass | |
| 4.3.4.18 | Call clearing | Interrupting the network connection of the SIP terminal device during a call
| Pass | |
| 4.4.19 | Hold | PBX => PSTN and PSTN => PBX
| Pass | |
| 4.4.19.1 | Hold | Putting an external call on hold in the PBX | Pass | |
| 4.4.19.2 | Hold | If applicable, MoH (music on hold) at A-party (PSTN) | Pass | |
| 4.4.19.3 | Hold | HOLD RETRIEVE: retrieving the external call | Pass | |
| 4.4.19.4 | Hold | Clearing the connection of the A-party while it is put on hold | Pass | |
| 4.4.20 | Hold | PBX => PSTN and PSTN => PBX
| Pass | |
| 4.4.20.1 | Hold | Putting an external call on hold in the PSTN | Pass | |
| 4.4.20.2 | Hold | If applicable, MoH (music on hold) at A-party (PBX) | Pass | |
| 4.4.20.3 | Hold | HOLD RETRIEVE: retrieving the external call | Pass | |
| 4.4.20.4 | Hold | Clearing the connection of the A-party while it is put on hold | Pass | |
| 4.5.21 | Call transfer | Internal call is transferred to external party: internal => PBX => external | Pass | |
| 4.5.21.1 | Call transfer | Call transfer from PBX party => PSTN party with announcement (attendant | Pass | |
| 4.5.21.2 | Call transfer | Call transfer from PBX party => PSTN party without announcement (blind transfer) | Pass | |
| 4.5.21.3 | Call transfer | Call transfer from PBX party => PSTN party without announcement (blind transfer)
| Pass | |
| 4.5.21.4 | Call transfer | Call transfer from PBX party => PSTN party without announcement (blind transfer)
| Pass | |
| 4.5.22 | Call transfer | Transferred call from external party to PBX: PSTN => PBX | Pass | |
| 4.5.22.1 | Call transfer | Call transfer from PSTN => PBX party with announcement (attendant transfer) | Pass | |
| 4.5.22.2 | Call transfer | Call transfer from PSTN => PBX party without announcement (blind transfer) | Pass | |
| 4.5.22.3 | Call transfer | Call transfer from PSTN => PBX party without announcement (blind transfer)
| Pass | |
| 4.5.22.4 | Call transfer | Call transfer from PSTN => PBX party without announcement (blind transfer)
| Pass | |
| 4.5.23 | Call transfer | Call from external party transferred to another external party: external => PBX => external | Pass | |
| 4.5.23.1 | Call transfer | PSTN => PBX party => PSTN with announcement (attendant transfer) | Pass | |
| 4.5.23.2 | Call transfer | PSTN => PBX party => PSTN without announcement (blind transfer) | Pass | |
| 4.5.23.3 | Call transfer | PSTN => PBX party => PSTN without announcement (blind transfer)
| Pass | |
| 4.5.23.4 | Call transfer | PSTN => PBX party => PSTN without announcement (blind transfer)
| Pass | |
| 4.6.24 | Call diversion | PBX party CFU to external party (PSTN) | Pass | |
| 4.6.24.1 | Call diversion | Internal call (CFU) => PSTN | Pass | |
| 4.6.24.2 | Call diversion | PSTN party busy, rejects call, does not answer | Pass | |
| 4.6.24.3 | Call diversion | A-party clears in ring phase | Pass | |
| 4.6.24.4 | Call diversion | External => PBX party (CFU) => PSTN | Pass | |
| 4.6.24.5 | Call diversion | PSTN party busy, rejects call, does not answer | Pass | |
| 4.6.24.6 | Call diversion | A-party clears in ring phase | Pass | |
| 4.6.25 | Call diversion | PBX party CFNR to external party | Pass | |
| 4.6.25.1 | Call diversion | Internal call (CFNR) => PSTN | Pass | |
| 4.6.25.2 | Call diversion | PSTN party busy, rejects call, does not answer | Pass | |
| 4.6.25.3 | Call diversion | A-party clears in ring phase | Pass | |
| 4.6.25.4 | Call diversion | External => PBX party (CFNR) => PSTN | Pass | |
| 4.6.25.5 | Call diversion | PSTN party busy, rejects call, does not answer | Pass | |
| 4.6.25.6 | Call diversion | A-party clears in ring phase | Pass | |
| 4.6.26 | Call diversion | PBX party CFB to external party | Pass | |
| 4.6.26.1 | Call diversion | Internal call (CFB) => PSTN | Pass | |
| 4.6.26.2 | Call diversion | PSTN party busy, rejects call, does not answer | Pass | |
| 4.6.26.3 | Call diversion | A-party clears in ring phase | Pass | |
| 4.6.26.4 | Call diversion | External => PBX party (CFB) => PSTN | Pass | |
| 4.6.26.5 | Call diversion | PSTN party busy, rejects call, does not answer | Pass | |
| 4.6.26.6 | Call diversion | A-party clears in ring phase | Pass | |
| 4.6.27 | Call diversion | External party (PSTN, mobile, etc.) CFU to PBX party: | Pass | |
| 4.6.27.1 | Call diversion | External (CFU) => PBX party | Pass | |
| 4.6.27.2 | Call diversion | PBX party busy, rejects call, does not answer | Pass | |
| 4.6.27.3 | Call diversion | A-party clears in ring phase | Pass | |
| 4.6.27.4 | Call diversion | External (CFNR) => PBX party | Pass | |
| 4.6.27.5 | Call diversion | PBX party busy, rejects call, does not answer | Pass | |
| 4.6.27.6 | Call diversion | A-party clears in ring phase | Pass | |
| 4.6.28 | Call diversion | External party (PSTN, mobile, etc.) CFB to PBX party: | Pass | |
| 4.6.28.1 | Call diversion | PBX party busy, rejects call, does not answer | Pass | |
| 4.6.28.2 | Call diversion | A-party clears in ring phase | Pass | |
| 4.6.29 | Call diversion | Call deflection: diversion during the ring phase | Pass | |
| 4.6.29.1.1 | Call diversion | Internal call PBX party CD => PBX party | Pass | |
| 4.6.29.1.2 | Call diversion | PBX party busy | Pass | |
| 4.6.29.1.3 | Call diversion | PBX party does not answer | Pass | |
| 4.6.29.1.4 | Call diversion | A-party clears in ring phase | Pass | |
| 4.6.29.2.1 | Call diversion | External call to PBX party CD => PBX party | Pass | |
| 4.6.29.2.2 | Call diversion | PBX party busy | Pass | |
| 4.6.29.2.3 | Call diversion | PBX party does not answer | Pass | |
| 4.6.29.2.4 | Call diversion | A-party clears in ring phase | Pass | |
| 4.6.29.3.1 | Call diversion | External call to PBX party CD => external PSTN party | Pass | |
| 4.6.29.3.2 | Call diversion | PSTN party busy | Pass | |
| 4.6.29.3.3 | Call diversion | PSTN party does not answer | Pass | |
| 4.6.29.3.4 | Call diversion | A-party clears in ring phase | Pass | |
| 4.7.30 | Call waiting | Incoming call during active internal call | Pass | |
| 4.7.30.1 | Call waiting | Call waiting tone | Pass | |
| 4.7.30.2 | Call waiting | Display of number of waiting party | Pass | |
| 4.7.30.3 | Call waiting | Acceptance of waiting call | Pass | |
| 4.7.30.4 | Call waiting | Putting the waiting call on hold | Pass | |
| 4.7.30.5 | Call waiting | Retrieve of waiting call | Pass | |
| 4.7.30.6 | Call waiting | Holding the 2nd call | Pass | |
| 4.7.30.7 | Call waiting | Terminating the active call | Pass | |
| 4.7.30.8 | Call waiting | Disregarding the waiting call | Pass | |
| 4.7.30.9 | Call waiting | Rejecting the waiting call | Pass | |
| 4.8.31 | 3-party conference | Establishing a conference according to the operating instructions of the PBX Internal - internal - external | Pass | |
| 4.8.31.1 | 3-party conference | Selecting a party (internal or external); 3rd party is put on hold
| Pass | |
| 4.8.32 | 3-party conference | Establishing a conference according to the operating instructions of the PBX | Pass | |
| 4.8.32.1 | 3-party conference | Selecting a party (external); 3rd party is put on hold
| Pass | |
| 4.9.33 | Pick up | Picking up a call from another extension of the PBX | Pass | |
| 4.10.34 | Call list | Entries in the call list
| Pass | |
| 4.10.35 | Call list | Call-back from call list
| Pass | |
| 4.11.36 | DTMF | DTMF support (G.711)
| Pass | |
| 4.11.37 | DTMF | DTMF support (G.729) | Pass | |
| 4.12.38 | Fax | Fax reception (G.711 only) | Pass | |
| 4.12.38.1 | Fax | Network-side T.38 re-invite rejected by PBX (response 488) or only G.711 codec is confirmed
| Pass | |
| 4.12.39 | Fax | Fax sending (G.711 only) | Pass | |
| 4.12.39.1 | Fax | Network-side T.38 re-invite rejected by PBX (response 488) or only G.711 codec is confirmed
| Pass | |
| 4.12.40 | Fax | Fax reception via T.38 | Pass | |
| 4.12.40.1 | Fax | Re-invite to T.38 by PBX or network | Pass | |
| 4.12.41 | Fax | Fax sending via T.38 (not possible in conjunction with the encryption option) | Pass | |
| 4.12.41.1 | Fax | Re-invite to T.38 by PBX or network
| Pass | |
| 5.42 | Redundancy | Test of redundancy:
| Pass | |
| 5.42.1 | Redundancy | 1. Register all available PBX systems the Plusnet SBC. | Pass |
Conclusion
This Application Notes document describes the steps required to configure the Ribbon SBC 1000/2000 to successfully interoperate with the Cisco CUCM and PlusNet SIP Trunk. All feature and serviceability test cases have been completed. The majority of test cases passed with noted exceptions and observations provided in Interoperability Test Results.
Overview
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