This document is intended for Ribbon's C20, AS, and SBC technical staff and any individual tasked with the integration of Ribbon's C20-AS-SBC with the SIP Trunking solution with the Avaya IP Office on the enterprise side. This document includes the configuration guide for the Ribbon series SBC Core, Enterprise SBC, C20, and AS to provide the SIP Trunk solution with Avaya IP Office PBX.
This is a general reference document that requires user input during the configuration of the Ribbon SBC - SWe Core, Enterprise SBC Edge 1000, C20, AS, and Avaya IP Office.
All test results and observations for Ribbon's C20-AS-SBCs and Avaya IPO interoperability have been mentioned in a separate Test Report document and are not part of this Configuration Guide.
Configuration Guides provide the system integrator with the necessary information to construct an integration similar to what has been built for certification or Interoperability Test (IOT) purposes. The IOT reference architecture is shown below in the Network Diagram section.
This configuration guide is divided into the following main sections:
Avaya IP Office
Ribbon SBC - SWe Core
Ribbon Enterprise 1K - Edge SBC
Ribbon AS
Ribbon C20
This is a technical document intended for telecommunications engineers with the purpose of configuring both the Ribbon and third-party product. Navigating third-party product as well as the Ribbon SBC command line interface (CLI) is required. Understanding about the Ribbon SBC Edge 1000, SBC SWe Core, SESM AS, C20, and Avaya IPO is also necessary to complete the configurations and for the basic troubleshooting.
The sample configuration uses the following equipment and software:
The main IOT Report for this activity captures the results where the registration model is a static PBX setup from the Avaya IP Office perspective to the AS.
This section provides the procedure for configuring the Avaya IP Office to support connectivity to the Ribbon C20-AS SIP Trunking solution through the SBC. This section requires you to have proper knowledge of the Avaya IP Office usage, configuration, and support in general, and experience with the product platform. Assuming that the basic configuration was already setup, the following screen captures show the SIP trunk configuration on the Avaya IP Office during the test execution. Avaya IP Office is configured using the Avaya IP Office Manager PC application with administrator login credentials.
The Avaya IP Office Manager was loaded onto the tester’s PC and allowed user login and access to the Avaya IP Office PBX. With Avaya IP Office Manager loaded on your local PC, select Program Files (x86) > Avaya > IP Office > Manager. Select the “Manager” application.
Enter the Service User Password for the Administrator user.
To access the System settings, click the name of the IP Office system. Select Sonus IP Office > System > system name “Sonus IP Office” and then click the System tab.
Access to the IP Office was gained through the LAN side of the PBX (LAN1). The SIP PBX phones also registered through the LAN side of the PBX.
To access the LAN1 settings, click the name of the IP Office system. Select Sonus IP Office > System > system name “Sonus IP Office” and then click the LAN1 tab.
Enter the IP Address and IP Mask for the LAN side of the IP Office PBX.
Check the SIP Registrar Enable box to allow the SIP phones to register to the IP Office PBX.
The UDP and TCP Protocols were set to 5060.
The UDP and TCP Public Ports were set to 5060.
The SIP Trunk to the E-SBC and C20 SIP Trunking Solution uses the WAN connection of the IP Office. To access the LAN2 settings, click the name of the IP Office system. Select Sonus IP Office > System > system name “Sonus IP Office” and then click the LAN2 tab.
During Interoperability testing, LAN2 was not used because LAN1 was connected to the Ribbon Enterprise SBC Edge 1000 and the Enterprise SBC had public internet connectivity over WAN.
To access the Voicemail settings, click the name of the IP Office system. Select Sonus IP Office > System > system name “Sonus IP Office” and then click the Voicemail tab.
Voicemail pro was installed on Enterprise network and was integrated with the IPO PBX. The voicemail server was hosted with IPO.
IPO SIP Line is a PBX SIP Trunk on IPO towards the Enterprise SBC Edge 1000.
Mention Transport with the next hop or Proxy IP address. During interoperability, the SBC Edge 1000 packet interface IP address was configured on the IPO SIP Trunk and SIP Line with transport protocol UDP and port 5060.
The SBC SWe is deployed with the correct build and running with the necessary license.
This section provides a sample of the Ribbon SBC SWe configuration used during the interoperability testing. The following commands and configurations are only for reference, other configurations are also possible based on the customer's requirement.
Create Codec Entry with the supported codec in the network.
set profiles media codecEntry G729A-IOT-TEST dtmf relay rfc2833 set profiles media codecEntry G729A-IOT-TEST packetSize 20 commit set profiles media codecEntry G711_ALAW_PTIME_20 dtmf relay rfc2833 set profiles media codecEntry G711_ALAW_PTIME_20 packetSize 20 commit
Configuring the RTCP interval.
set system media mediaRtcpControl senderReportInterval 5 commit
This configuration only applies if the SBC has been deployed with (hardware) DSP resources. If it has not, executing this configuration step has no negative impact.
The subsequent configuration section (Packet Service Profiles) does not attempt transcoding, so the lack of compression resources will not impact the overall SBC configuration in this document.
set system mediaProfile compression 75 tone 25 commit
set profiles media packetServiceProfile CORE_PSP codec codecEntry1 G729A-IOT-TEST set profiles media packetServiceProfile CORE_PSP codec codecEntry2 G711_ALAW_PTIME_20 set profiles media packetServiceProfile CORE_PSP packetToPacketControl transcode transcoderFreeTransparency set profiles media packetServiceProfile CORE_PSP packetToPacketControl codecsAllowedForTranscoding thisLeg "" set profiles media packetServiceProfile CORE_PSP packetToPacketControl codecsAllowedForTranscoding otherLeg "" set profiles media packetServiceProfile CORE_PSP rtcpOptions rtcp enable set profiles media packetServiceProfile CORE_PSP preferredRtpPayloadTypeForDtmfRelay 101 commit
Create an IP signaling profile for the C20/AS side. The IPSP will be specified within the SIP trunk group configuration.
set profiles signaling ipSignalingProfile CORE_IPSP set profiles signaling ipSignalingProfile CORE_IPSP ipProtocolType sipOnly set profiles signaling ipSignalingProfile CORE_IPSP commonIpAttributes flags includeTransportTypeInContactHeader enable set profiles signaling ipSignalingProfile CORE_IPSP commonIpAttributes flags minimizeRelayingOfMediaChangesFromOtherCallLegAll enable set profiles signaling ipSignalingProfile CORE_IPSP commonIpAttributes flags relayDataPathModeChangeFromOtherCallLeg enable set profiles signaling ipSignalingProfile CORE_IPSP commonIpAttributes flags noPortNumber5060 enable set profiles signaling ipSignalingProfile CORE_IPSP commonIpAttributes relayFlags dialogEventPackage enable set profiles signaling ipSignalingProfile CORE_IPSP commonIpAttributes relayFlags info enable set profiles signaling ipSignalingProfile CORE_IPSP commonIpAttributes relayFlags notify enable set profiles signaling ipSignalingProfile CORE_IPSP commonIpAttributes relayFlags refer enable set profiles signaling ipSignalingProfile CORE_IPSP commonIpAttributes relayFlags statusCode4xx6xx enable set profiles signaling ipSignalingProfile CORE_IPSP commonIpAttributes relayFlags updateWithoutSdp enable set profiles signaling ipSignalingProfile CORE_IPSP commonIpAttributes transparencyFlags authcodeHeaders enable set profiles signaling ipSignalingProfile CORE_IPSP commonIpAttributes transparencyFlags mwiBody enable set profiles signaling ipSignalingProfile CORE_IPSP commonIpAttributes transparencyFlags referredByHeader enable set profiles signaling ipSignalingProfile CORE_IPSP commonIpAttributes transparencyFlags sipfragBody enable set profiles signaling ipSignalingProfile CORE_IPSP commonIpAttributes transparencyFlags unknownBody enable set profiles signaling ipSignalingProfile CORE_IPSP egressIpAttributes flags disable2806Compliance enable set profiles signaling ipSignalingProfile CORE_IPSP egressIpAttributes privacy transparency enable set profiles signaling ipSignalingProfile CORE_IPSP ingressIpAttributes flags sendUpdatedSDPin200Ok enable set profiles signaling ipSignalingProfile CORE_IPSP egressIpAttributes transport type1 udp set profiles signaling ipSignalingProfile CORE_IPSP egressIpAttributes transport type2 tcp commit
Create an IP interface group and assign its interface and IP address.
set addressContext default ipInterfaceGroup CORE_LIF ipInterface IPIF2 ceName IOTSBC1 portName pkt0 set addressContext default ipInterfaceGroup CORE_LIF ipInterface IPIF2 ipAddress 172.28.xxx.xx set addressContext default ipInterfaceGroup CORE_LIF ipInterface IPIF2 prefix xx set addressContext default ipInterfaceGroup CORE_LIF ipInterface IPIF2 mode inService state enabled commit
This Zone groups the set of objects that communicate to the SESM AS.
set addressContext default zone CORE id 3 set addressContext default zone CORE remoteDeviceType appServer commit
A SIP signaling port is a logical address that sends and receives SIP call signaling packets and is permanently bound to a specific zone.
set addressContext default zone CORE sipSigPort 3 ipInterfaceGroupName CORE_LIF ipAddressV4 172.28.xxx.xx portNumber 5060 transportProtocolsAllowed sip-udp,sip-tcp set addressContext default zone CORE sipSigPort 3 state enabled mode inService commit
DNS Groups set DNS objects that may be used for DNS resolution within a particular Zone.
set addressContext default dnsGroup EXT_DNS set addressContext default dnsGroup EXT_DNS type ip interface IPIF2 server DNS2 ipAddress x.x.x.x state enabled set addressContext default zone CORE dnsGroup EXT_DNS commit
Create a SIP Trunk Group for the SESM-AS side and assign the IPSP and PSP configured above. For ingressIpPrefix
, replace x.x.x.x
with the IP address prefix that you want to allow. Multiple SESM IP addresses can be configured.
set addressContext default zone CORE sipTrunkGroup CORE_STG media mediaIpInterfaceGroupName CORE_LIF set addressContext default zone CORE sipTrunkGroup CORE_STG signaling honorMaddrParam enabled set addressContext default zone CORE sipTrunkGroup CORE_STG policy media packetServiceProfile CORE_PSP set addressContext default zone CORE sipTrunkGroup CORE_STG policy signaling ipSignalingProfile CORE_IPSP set addressContext default zone CORE sipTrunkGroup CORE_STG services dnsSupportType a-srv-naptr set addressContext default zone CORE sipTrunkGroup CORE_STG ingressIpPrefix x.x.x.x x set addressContext default zone CORE sipTrunkGroup CORE_STG signaling relayNonInviteRequest enabled set addressContext default zone CORE sipTrunkGroup CORE_STG media sdpAttributesSelectiveRelay enabled set addressContext default zone CORE sipTrunkGroup CORE_STG mode inService state enabled commit
Create a default route to the subnet's next hop IP for the interface and IP Interface Group.
set addressContext default staticRoute X.X.X.X X x.X.X.X CORE_LIF IPIF2 preference 100 commit
Create an IP Peer with the AS IP address and assign it to the CORE zone.
set addressContext default zone CORE ipPeer CORE_PEER ipAddress x.x.x.x ipPort 5060 commit
Create a Packet Service Profile (PSP) for the SBC Edge side. The PSP will be specified within the SIP Trunk Group Configuration.
set profiles media packetServiceProfile ACCESS_PSP codec codecEntry1 G729A-IOT-TEST set profiles media packetServiceProfile ACCESS_PSP codec codecEntry2 G711_ALAW_PTIME_20 set profiles media packetServiceProfile ACCESS_PSP packetToPacketControl transcode transcoderFreeTransparency set profiles media packetServiceProfile ACCESS_PSP packetToPacketControl codecsAllowedForTranscoding thisLeg "" set profiles media packetServiceProfile ACCESS_PSP packetToPacketControl codecsAllowedForTranscoding otherLeg "" set profiles media packetServiceProfile ACCESS_PSP rtcpOptions rtcp enable set profiles media packetServiceProfile ACCESS_PSP preferredRtpPayloadTypeForDtmfRelay 101 commit
Create an IP Signaling Profile (IPSP) for the SBC Edge side. The IPSP will be specified within the SIP Trunk Group Configuration.
set profiles signaling ipSignalingProfile ACCESS_IPSP set profiles signaling ipSignalingProfile ACCESS_IPSP ipProtocolType sipOnly set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes flags includeTransportTypeInContactHeader enable set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes flags minimizeRelayingOfMediaChangesFromOtherCallLegAll enable set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes flags relayDataPathModeChangeFromOtherCallLeg enable set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes flags noPortNumber5060 enable set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes relayFlags dialogEventPackage enable set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes relayFlags info enable set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes relayFlags notify enable set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes relayFlags refer enable set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes relayFlags statusCode4xx6xx enable set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes relayFlags updateWithoutSdp enable set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes transparencyFlags authcodeHeaders enable set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes transparencyFlags mwiBody enable set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes transparencyFlags referredByHeader enable set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes transparencyFlags sipfragBody enable set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes transparencyFlags unknownBody enable set profiles signaling ipSignalingProfile ACCESS_IPSP egressIpAttributes flags disable2806Compliance enable set profiles signaling ipSignalingProfile ACCESS_IPSP egressIpAttributes privacy transparency enable set profiles signaling ipSignalingProfile ACCESS_IPSP ingressIpAttributes flags sendUpdatedSDPin200Ok enable set profiles signaling ipSignalingProfile ACCESS_IPSP egressIpAttributes transport type1 udp set profiles signaling ipSignalingProfile ACCESS_IPSP egressIpAttributes transport type2 tcp commit
Create IP Interface Group and assign its interface and IP address for connectivity towards the SBC Edge.
set addressContext default ipInterfaceGroup ACCESS_LIF ipInterface IPIF0 ceName IOTSBC1 portName pkt1 set addressContext default ipInterfaceGroup ACCESS_LIF ipInterface IPIF0 ipAddress xxx.xx.xxx.xx set addressContext default ipInterfaceGroup ACCESS_LIF ipInterface IPIF0 prefix xx set addressContext default ipInterfaceGroup ACCESS_LIF ipInterface IPIF0 mode inService state enabled commit
This Zone groups the set of objects that communicate with the SBC Edge 1000.
set addressContext default zone ACCESS id 2 commit
set addressContext default zone ACCESS id 2 sipSigPort 2 ipInterfaceGroupName ACCESS_LIF ipAddressV4 XXX.XXX.XXX.XXX portNumber 5060 transportProtocolsAllowed sip-tcp,sip-udp set addressContext default zone ACCESS id 2 sipSigPort 2 mode inService state enabled commit
Create a SIP Trunk Group towards the Enterprise SBC Edge 1000 side and assign the PSP and IPSP configured above.
set addressContext default zone ACCESS sipTrunkGroup ACCESS_STG media mediaIpInterfaceGroupName ACCESS_LIF set addressContext default zone ACCESS sipTrunkGroup ACCESS_STG signaling honorMaddrParam enabled set addressContext default zone ACCESS sipTrunkGroup ACCESS_STG policy media packetServiceProfile ACCESS_PSP set addressContext default zone ACCESS sipTrunkGroup ACCESS_STG policy signaling ipSignalingProfile ACCESS_IPSP set addressContext default zone ACCESS sipTrunkGroup ACCESS_STG services dnsSupportType a-srv-naptr set addressContext default zone ACCESS sipTrunkGroup ACCESS_STG ingressIpPrefix x.x.x.x x set addressContext default zone ACCESS sipTrunkGroup ACCESS_STG signaling relayNonInviteRequest enabled set addressContext default zone ACCESS sipTrunkGroup ACCESS_STG media sdpAttributesSelectiveRelay enabled set addressContext default zone ACCESS sipTrunkGroup ACCESS_STG mode inService state enabled commit
Create an IP Peer with the Fully-Qualified Domain Name (FQDN) or IP address of the endpoint and assign it to the PSTN Side.
set addressContext default zone ACCESS ipPeer ACCESS_PEER ipAddress x.x.x.x ipPort 5060 commit
set addressContext default staticRoute X.X.X.X X X.X.X.1 ACCESS_LIF IPIF0 preference 100 commit
Create a Routing Label with a single Routing Label Route to bind the egress Trunk Group for the C20-AS and the Enterprise SBC Edge 1000 IP Peer.
set global callRouting routingLabel ACCESS_RL routingLabelRoute 1 trunkGroup ACCESS_STG ipPeer ACCESS_PEER inService inService set global callRouting routingLabel CORE_RL routingLabelRoute 1 trunkGroup CORE_STG ipPeer CORE_PEER inService inService commit
Routing must be put in place to send calls to the correct destination. Additional routing options may be used based on the requirements.
The configuration of both standard and username routes are done to ensure that no matter how the called party is addressed (a number or username) the SBC routes the message to the Core. Create Route entries for standard Trunk Group routing with Matching Criteria and a Routing Label destination.
set global callRouting route trunkGroup ACCESS_STG IOTSBC1 standard Sonus_NULL Sonus_NULL all all ALL none Sonus_NULL routingLabel CORE_RL set global callRouting route trunkGroup ACCESS_STG IOTSBC1 username Sonus_NULL Sonus_NULL all all ALL none Sonus_NULL routingLabel CORE_RL set global callRouting route trunkGroup CORE_STG IOTSBC1 standard Sonus_NULL Sonus_NULL all all ALL none Sonus_NULL routingLabel ACCESS_RL commit
Configuring the SBC 1000 is done through the SBC’s integrated web server. This guide assumes that the operator has already done the initial configuration that positions the SBC Edge 1000 on the IP network.
To start the configuration process, use a standard web browser to connect to the IP or FQDN address of the SBC. Supply the username and password to complete the login process.
Use the Initial Task configuration procedure to position the SBC Edge 1000 between the Ribbon SBC Core and Avaya IPO PBX. This task will create SIP components and call routing basics.
Create profiles with a specific set of characteristics that correspond to the Avaya IPO PBX. This profile creation includes configuring the following entities on the SBC Edge 1000.
Create a Media Profile towards the Interactive Intelligence side with G711a law as the first codec, G711u law as the second codec, and T.38 Fax as the third codec.
To create or modify a Media Profile:
Media List allows you to specify a set of codecs and fax profiles that are allowed on a given SIP Signaling Group. They contain one or more Media Profiles, which you must first define in Media List. These lists allow you to accommodate specific transmission requirements and SIP devices that only implement a subset of the available voice codecs.
To create or modify a Media List:
To create or modify an existing SIP Profile:
SIP Server Tables contain information about the SIP devices connected to the SBC Edge 1000. The entries in the tables provide information about the IP Addresses, ports, and transport protocols used to communicate with each server. The Table Entries also contain links to counters that are useful for troubleshooting.
To create or modify an existing SIP Server Table:
Signaling groups allow telephony channels to be grouped together for the purposes of routing and shared configuration. They are the entity to which calls are routed, as well as the location from which Call Routes are selected. They are also the location from which Tone Tables and Action Sets are selected. In the case of SIP, they specify protocol settings and link to server, media, and mapping tables.
To create or modify an existing Signaling Group:
IP Address/FQDN in Federated IP/FQDN: Specifies the IP Address or Fully Qualified Domain Name of a server from which the SBC Edge 1000 will accept SIP messages. It should have the IP of Avaya IPO PBX.
Call Routing allows calls to be carried between signaling groups, which allows calls to be carried between ports and between protocols (for example, ISDN to SIP). Routes are defined by Call Routing Tables, which allow for flexible configuration of which calls are carried and how they are translated. These tables are one of the central connection points of the system, linking Transformation Tables, Message Translations, Cause Code Reroute Tables, Media Lists and the three types of Signaling Groups (ISDN, SIP and CAS).
To create or modify an existing Call Routing Table:
The following are definitions of fields in the Call Routing Table:
Destination Signaling Group: Specifies the Signaling Groups used as the destination of calls.
Media Mode: Media Mode should be RTP DSP.
Media Transcoding: It should be enabled in order for the SBC Edge 1000 to transcode.
Media List: Specifies the Media List used for this call route.
Create profiles with a specific set of characteristics that correspond to the SBC Core. It includes configuring the following entities on the SBC Edge 1000:
Create Media Profile towards Interactive Intelligence side with G711a law as the first codec, G711u law as the second codec, and T.38 Fax as the third.
To create or modify a Media Profile:
Media List allows you to specify a set of codecs and fax profiles that are allowed on a given SIP Signaling Group. They contain one or more Media Profiles, which must first be defined in Media List. These lists allow you to accommodate specific transmission requirements and SIP devices that only implement a subset of the available voice codecs.
To create or modify a Media List:
To create or modify an existing SIP Profile:
SIP Server Tables contain information about the SIP devices connected to the SBC Edge 1000. The entries in the tables provide information about the IP Addresses, ports, and transport protocols used to communicate with each server. The Table Entries also contain links to counters that are useful for troubleshooting.
To create or modify an existing SIP Server Table:
Signaling groups allow telephony channels to be grouped together for the purposes of routing and shared configuration. They are the entity to which calls are routed, as well as the location from which Call Routes are selected. They are also the location from which Tone Tables and Action Sets are selected. In the case of SIP, they specify protocol settings and link to server, media, and mapping tables.
To create or modify an existing Signaling Group:
IP Address/FQDN in Federated IP/FQDN: Specifies the IP Address or Fully Qualified Domain Name of a server from which the SBC Edge 1000 will accept SIP messages. It should have the IP of Avaya IPO PBX.
Call Routing allows calls to be carried between signaling groups, which allows calls to be carried between ports and between protocols (for example, ISDN to SIP). Routes are defined by Call Routing Tables, which allow for flexible configuration of which calls are carried and how they are translated. These tables are one of the central connection points of the system, linking Transformation Tables, Message Translations, Cause Code Reroute Tables, Media Lists and the three types of Signaling Groups (ISDN, SIP and CAS).
To create or modify an existing Call Routing Table:
The following are descriptions of some fields in the Call Routing Table:
Destination Signaling Group: Specifies the Signaling Groups used as the destination of calls.
Media Mode: Media Mode should be RTP DSP.
Media Transcoding: It should be enabled in order for SBC Edge 1000 to transcode.
Media List: Specifies the Media List used for this call route.
To configure the Static SIP PBX you will need access to the Ribbon AS System Management Console.
The following documentation was used to configure the AS: 630-01981-01_02.01_1.2_C20_ Enterprise SIP Trunking Solution C20-EXPERiUS SIP Trunking Solution Guide.
The Static SIP PBX does not send SIP REGISTER messages to notify the network of their signaling address, therefore, the address of the SIP PBX must be provisioned in the network. Configure the address of the private side of the SBC that represents the static SIP PBX.
Select Network Data and Mtc > Addresses. Add an address as shown in the following example.
Configure an external node by mapping the Static SIP PBX address to the node name. Select Network Data and Mtc > External Nodes. Add an External Node as shown in the example below.
From Network Data and Mtc, select C20 Converged Softswitch Integration, SIP PBX, and then C20 SIP PBX. Add a C20 SIP PBX as referenced below using the C20 CIP PBX “tkv4st.”
The AYT Profile is mandatory if the AYT Audit was selected in the C20 SIP PBX setup as shown in the preceding screen capture. Here you will select SIP OPTIONS or INFO messages will be used and valid responses to the AYT audit. The AYT audits will be sent by the C20/AS to monitor the connectivity to the SIP PBX. From Network Data and Mtc select AYT Profiles.
SIPTrunkingdefaultOptions was used during testing.
A new Header Map was created for this test. This maps specific SIP messages to specific PRI messages. The default Header Map was modified (see the following screen capture). From Network Data and Mtc select the C20 Converged Softswitch Integration folder and then SIP/PRI Header Mapping.
The PRI header Original Called Party Number was changed from the History-Info header to the Diversion header. This instructs the AS to use the Diversion header containing the Original Called party Number. When the Call is sent to the CIM Voice Mail server, the Diversion header is in the INVITE message.
A custom SIP Profile may be required to support different types of SIP PBX. In this case Ribbon recommends you copy the SipPBX SIP Profile and only change the settings required to support the particular SIP PBX. From Network Data and Mtc select the SIP Profiles folder and then SIP Profiles.
Then give the profile a name and description.
The following default settings are viewed when clicking the Select Requests link.
The following default settings are viewed when clicking the Select Headers link.
The following default settings are viewed when clicking the Select Allow Methods link.
Configure the AS to require authentication for both SIP INVITE and REGISTER transactions. From Network Elements select Session Managers, SessionManagerX, and then Authorized Methods.
SESM Configuration Parameters and associated values are set globally within the AS and are not unique to the SIP PBX. From Network Elements select Session Managers, SESMx, and then Configuration Parameters.
For each AS SESM, set the AuthenticationWithIntegrity parameter to false and Parm Group = Authentication
This parameter determines the length of time, in minutes, between endpoint audits. Duration is used to detect abandoned calls. A value of zero deactivates the duration parameter.
Make sure Parm Group = LongCall.
Defines a name for the SIP PBX to be used in other associated provisioning entities.
From Network Elements select Session Managers > SessionManager1 > C20 Converged Softswitch Integration > SIP PBX Integration > SIP PBX Configuration. The SIP PBX Configuration used for our setup was Tkv4StaticSipPbx.
Establishes links between the AS SIP PBX entity and the C20 by associating the SIP PBX name, defined in the AS SIP PBX Configuration, with an ISDN Logical Terminal ID (LTID) and Virtual Media Gateway defined in the C20.
After the SIP PBX is configured in the Provisioning Manager shown in the following screen capture, you will be able to bring up the link.
The AS Provisioning Manager is the AS web-based interface accessible at the following link: http://<prov blade IP address>:8443/prov.
Defines the SIP PBX Entity, established through the AS System Manager, as an AS Service Node. In the Translations tab select Service Node.
The following are definitions of fields in the Service Node screen:
From the Solution tab select SIP PBX then create a SIP PBX. Specify a username and password for the SIPPBX that will be used to authenticate associated transactions specified in the Authorized Methods.
The following is a SIP PBX example.
Add a Gateway of type VOIP_VPN associated with the SIP PBX Entity (in this case tekVizion4SIP-PBX). The IP address must be the service address of the AS SESM on which the SIP PBX is configured.
Define Carrier(s) that will be used to assign the ISDN PRI trunks in C20 associated with the SIP PBX Entity. In this example, carrier EPG_002 for (SIP PBX) gateway tekVizion4SIP-PBX has 23 channels assigned starting at 2. These will be used as bearer channels. Carrier EPG_001 for tekVizion4SIP-PBX has a single channel (1) that will serve as the ISDN PRI “D channel” for signaling.
The node number is (92).
Define the Trunk Group name and size.
Define an ISDN PRI trunk group supporting the SIP PBX.
GRPKEY is derived from table CLLI.
GRPTYP = PRA is required for ISDN PRI trunk group.
LTID is derived from table LTDEF and cannot be datafilled manually. Therefore, enter $ for this field when initially adding the SIP PBX trunk group.
Define a trunk subgroup entry for the SIP PBX ISDN PRI trunk group SGRPKEY value TEKVIZION is derived from table CLLI; 0 means it is the 0th trunk subgroup for TEKVIZION.
SGRPVAR ISDN is required to indicate ISDN PRI.
PMTYPE, GWCNO, GWCNODENO, and GWCTRMNO are all values based on the SIP PBX gateway defined by the Gateway Controller provisioning.
The value for GWCTRMNO (92), defined in Gateway Controller Carrier provisioning, is the D channel for the SIP PBX ISDN PRI trunk group.
Define trunk members for the SIP PBX trunk group using the bearer path carrier channels (total 23) defined at the Gateway Controller.
Define the Logical Terminal ID ISDN 15 for use with the SIP PBX trunk group.
Map the Logical Terminal ID, defined in table LTDEF, to the SIP PBX trunk group
(TEKVIZION_4).
Define translation paths for LTID ISDN 15.
Public routes use XLARTE (translation route selector) XLALEC.
Define service data associated with LTID ISDN 15.
OPTION PRI_IP_PROT is used to define the IP protocol for ISDN 14 as SIP (required for SIP PBX trunks).
If using centralized C20 voice mail, table MSGRTE is used to route message waiting indications back to the mailbox subscriber.
In this example, any MWI for DNs in this range are sent to the PRA trunk group TEKVIZION_4 (LTID ISDN 15).
PSTN to Avaya IP Office call.
An example of the translation flow to route calls from PSTN line to SIP PBX trunk members is shown in the TRAVER outputs below.
An example of the translation flow to route calls from SIP PBX trunk members to PSTN line is shown in the TRAVER outputs below.
Dynamic PBX Registration for Avaya IPO is supported by C20-AS. Currently the Ribbon SBC Edge does not support the Dynamic Registration feature. So dynamic PBX registration related configuration is out of scope in this configuration guide.
This detailed document describes the configuration steps required for the Ribbon SESM AS, C20, SBC Core, and SBC Edge 1000 to successfully interoperate with Avaya IPO over SIP Trunking solution.