SM SIP Trunking with CM 1

[halj]

Lets say I have 2 SIP trunks (TCP) between CM and SM.

How SM decides what trunk it will use to send the call to CM?
Asking this because I was getting some calls on SIP trunk 1. For some reason now I am getting calls from the same extensions on another SIP trunk.

From CM to SM we know about ARS and AAR, but what about SM to CM?

 

[kyle555]

SM has a Routing Pattern which has a SIP Entity as destination.
Entities have Entity Links, but you don't get the luxury of picking a specific Entity Link.

So, if you happened to have a CM SIP Entity and 2 links - TCP 5060 and TLS 5061, SM would pick for you.

Generally, you don't want to mix things up and you'd try to have a design where the stuff coming in would line up with the way you setup AAR/ARS for the stuff going out.

Upon receiving a SIP invite, CM will match IP/port. It'll find all sig groups that match. So, if you had sig/trunk group 101,201,301,401, etc to SM on the same IP and Port, CM would consider them all a match.

CM would then start counting from the lowest - 101 - and evaluate the far end domain in the URI. If the far end domain of the sig group fits in the URI, then it's a match.

So, a sig group with FE domain avaya.com would match an INVITE coming in @voicemail.avaya.com. Conversely, a CM sig group with FE domain voicemail.avaya.com would not match an INVITE @avaya.com.

Logic goes that you put your most explicit subdomains - east.voicemail.avaya.com as the lowest numbered sig groups and your least explicit ones as the highest numbered sig groups to not have things overlap.

 

[mattKnight]

That explanation is worth a star!

Take Care

Matt
I have always wished that my computer would be as easy to use as my telephone.
My wish has come true. I no longer know how to use my telephone.