Avaya CM 5.1
This is currently a work in progress.
I can dial "extensions" from the Avaya to sipXecs, send calls to voicemail and host a conference call (8 parties tested so far)
blind transfers from an automated attendant to a conference work, there may be other parts that do not yet work...
On the Avaya CM
change node-names ip
In the example below sipXecs is the IP node-name used, and domain.local is the SIP domain used
and add the sipeXecs name and address
now add a signalling group, the far end domain is deliberately blank
change signaling-group 60 Page 1 of 1 SIGNALING GROUP Group Number: 60 Group Type: sip Transport Method: tcp Co-Resident SES? y Near-end Node Name: procr Far-end Node Name: sipXecs Near-end Listen Port: 5062 Far-end Listen Port: 5060 Far-end Network Region: 1 Far-end Domain: Bypass If IP Threshold Exceeded? n DTMF over IP: rtp-payload Direct IP-IP Audio Connections? n IP Audio Hairpinning? n Enable Layer 3 Test? n Session Establishment Timer(min): 3 Alternate Route Timer(sec): 6
The "Near-end Listen Port" needs to be unique on the CM system
Now add a trunk group using the signalling group
change trunk-group 60 Page 1 of 22 TRUNK GROUP Group Number: 60 Group Type: sip CDR Reports: y Group Name: loewy-conf-2 COR: 1 TN: 4 TAC: 460 Direction: two-way Outgoing Display? n Dial Access? n Night Service: Queue Length: 0 Service Type: public-ntwrk Auth Code? n Signaling Group: 60 Number of Members: 30
Set the number of members to be the number of channels desired on the link
change trunk-group 60 Page 2 of 22 Group Type: sip TRUNK PARAMETERS Unicode Name? y Redirect On OPTIM Failure: 5000 SCCAN? n Digital Loss Group: 18 Preferred Minimum Session Refresh Interval(sec): 600
change trunk-group 60 Page 3 of 22 TRUNK FEATURES ACA Assignment? n Measured: none Maintenance Tests? y Numbering Format: public UUI Treatment: shared Maximum Size of UUI Contents: 128 Replace Restricted Numbers? n Replace Unavailable Numbers? n Send UCID? n Show ANSWERED BY on Display? y
change trunk-group 60 Page 4 of 22 SHARED UUI FEATURE PRIORITIES ASAI: 1 Universal Call ID (UCID): 2 MULTI SITE ROUTING (MSR) In-VDN Time: 3 VDN Name: 4 Collected Digits: 5 Other LAI Information: 6
change trunk-group 60 Page 5 of 22 PROTOCOL VARIATIONS Mark Users as Phone? n Prepend '+' to Calling Number? n Send Transferring Party Information? n Telephone Event Payload Type:
Obviously the the "extensions" that exist on the sipXecs side need to be suitable configured on the Avaya side. I did this by putting them into the uniform dialplan as aar numbers, and in the aar analysis table to route to a route pattern that points at the SIP trunk.
On the sipXecs server
In System | Servers | Configure
Ensure that SIP Trunking and Conferencing have ticks (I tick them all...)
In Devices | Gateways
Add an entry for CM, using the the port defined for the CM signalling group "Far-end Listen Port", set the transport to be TCP
You should now be able to dial from the Avaya to the sipXecs