Hello All, We're running CM 5.2.1, and also using a 3rd party IVR system, Microlog. The interface between the two is a T1/SIP gateway. We'd like use the SIP capabilities. So my question is can we do it without Session Manager or is it required? Thanks In Advance.
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SIP Trunking
- Thread starter tone70
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ewaaijenberg
Technical User
If the CM to the Microlog direct connection is not working you can also try a CM - SBCe - Microlog connection.
Best is to have a devconnect certified connection for Avaya support purposes.
Best is to have a devconnect certified connection for Avaya support purposes.
Loose outline:
(1) change node-names ip -> add microlog and its IP address
(2) add signalling group (next or define a #), group type sip, transport tcp, near end is procr, far end is node name from step 1
(3) add trunk (next or define a #), group type sip, using signalling group from step 2, (probably) set event payload type to 101
(4) set uniform dialplan to send a test number out aar, set aar to receive test number and point to trunk from step 3
(5) you probably need to set the microlog side to accept calls from the ip address of your pbx
start a sip trace on the tac of the new trunk group (list trace tac 8101/s), make call to test number, see what happens. Hopefully you can view the sip logs from the microlog side as well.
(1) change node-names ip -> add microlog and its IP address
(2) add signalling group (next or define a #), group type sip, transport tcp, near end is procr, far end is node name from step 1
(3) add trunk (next or define a #), group type sip, using signalling group from step 2, (probably) set event payload type to 101
(4) set uniform dialplan to send a test number out aar, set aar to receive test number and point to trunk from step 3
(5) you probably need to set the microlog side to accept calls from the ip address of your pbx
start a sip trace on the tac of the new trunk group (list trace tac 8101/s), make call to test number, see what happens. Hopefully you can view the sip logs from the microlog side as well.
IPOthermia
Vendor
Just a side note to clarify that AVAYA CM 6.0 supports direct SIP.
Ran AVST voicemail with direct SIP with CM 6.0
Ran Modular Messaging with direct SIP with CM 6.2
Currently running AVST with CM 6.3 also direct sip
Seems to work well, please note I said something nice about SIP before you read the last paragraph.
CM 5.2.1 was a bit of an odd duck when it came to setting up SIP, at least I thought it was. Might be a good time for an upgrade if you have 5.2.1.
We do not have system manager/session manager. I don't like purchasing 'Interoperability' servers for the purpose of making "supposedly interoperable" standards, interoperate. After all, the telephony network was founded on interoperability and the whole premise of SIP was for the endpoints to setup up communication without involving a server as in (Session Initiated). What a JUNK protocol it turned out to be in the last 20 years. Maybe SIP will have all of the features and flexibility of H323 just prior to WebRTC and HTML5 driving a stake through it.
Signed with all my Love.
Ran AVST voicemail with direct SIP with CM 6.0
Ran Modular Messaging with direct SIP with CM 6.2
Currently running AVST with CM 6.3 also direct sip
Seems to work well, please note I said something nice about SIP before you read the last paragraph.
CM 5.2.1 was a bit of an odd duck when it came to setting up SIP, at least I thought it was. Might be a good time for an upgrade if you have 5.2.1.
We do not have system manager/session manager. I don't like purchasing 'Interoperability' servers for the purpose of making "supposedly interoperable" standards, interoperate. After all, the telephony network was founded on interoperability and the whole premise of SIP was for the endpoints to setup up communication without involving a server as in (Session Initiated). What a JUNK protocol it turned out to be in the last 20 years. Maybe SIP will have all of the features and flexibility of H323 just prior to WebRTC and HTML5 driving a stake through it.
Signed with all my Love.
no SIP without SES" comes up a lot with a lot of misinformation. SES/SM is required for SIP endpoints/phones as this is what the phone actually registers to.
For trunking, direct SIP was always available but the older CM releases only supported TLS for the transport method. 99% of LEC's and other equipment would only support TCP/UDP (not TLS) for the SIP transport so an SES was required to do the conversion. As of CM5.2 (and possibly as far back as CM5.0) TCP became an available option to use for the TRANSPORT METHOD on the signaling group form. Once TCP became an available option, you could then implement SIP trunks via TCP which is supported by most vendors without requiring an SES or SM. So the question is, does your IVR support TCP as the SIP transport? Your 5.2.1 can use TLS or TCP but not UDP for direct SIP trunking.
-CL
For trunking, direct SIP was always available but the older CM releases only supported TLS for the transport method. 99% of LEC's and other equipment would only support TCP/UDP (not TLS) for the SIP transport so an SES was required to do the conversion. As of CM5.2 (and possibly as far back as CM5.0) TCP became an available option to use for the TRANSPORT METHOD on the signaling group form. Once TCP became an available option, you could then implement SIP trunks via TCP which is supported by most vendors without requiring an SES or SM. So the question is, does your IVR support TCP as the SIP transport? Your 5.2.1 can use TLS or TCP but not UDP for direct SIP trunking.
-CL
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- #9
IPOthermia
Vendor
The SIP signaling group in 6.0, 6.2, and 6.3 only lists TCP and TLS as available options. If you were going to use UDP, something would have to translate the UDP to work with the PBX's requirement for TCP or TLS. This is where System Manager/Session Manager would enter the picture. This is also necessary if you are using SIP phones since the embedded SES in 5.2.1 goes away after 6.0 of CM. If you are running on a standalone SES you may be able to continue running with it, but it is NOT supported by AVAYA after version 6.0 of CM. We were able to load our standalone SES license on the latest version of SES running on our VWare platform. Again, not supported by AVAYA, or as AVAYA put it in the documents to update from 5.2.1 to 6.0 of CM. "SES servers will not be supported in any form factor". AVAYA wants YOU to spend some MONEY!!!
I don't know much about Microlog, but a quick Google search hints that H.323 might be an option. Of course H.323 doesn't "enjoy" the associated expenses, or internet hype (dogma) as SIP but it would work well if the other vendor still supports it.
I don't know much about Microlog, but a quick Google search hints that H.323 might be an option. Of course H.323 doesn't "enjoy" the associated expenses, or internet hype (dogma) as SIP but it would work well if the other vendor still supports it.
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