Hello,
One of our clients recently migrated from ISDN to SIP, ever since they started having problems calling us. I know that their Invites come with a min-se of 1200, our CM has the min-se at 57600, "ours is way high but that's how they got it working way before I got here" but I know that we have to match their value to allow calls to go through, our idea to no mess the current deployment was to create a vector working under vector routing table so that calls coming with their DID's be routed based on ANI towards a new trunk that does have the timer matching to what the customer has. we gave the customer a new number to dial, this new number is what was assigned to the VDN that will eventually have matched ANI's send out a new route pattern that leads towards the new trunk. If I make a test call from a cell phone I see the routing of the call traversing the expected path and I see the 200 OK coming back from CM with the new timer, but calls from our client still don't work, their calls don't even hit the VDN, their calls go across the SBC and session manager as expected, session manager hands out the invite to CM but CM won't debug anything on the list trace VDN, however session manager receives a response back from the CM as 422 session timer too small, it is like the configuration we made for them is invisible and calls still go the same path that makes it fail, that is what I don't understand. I saw that their calls come in with e164, so tested by adding +1 within the vector routing table and also deleting the +1 in the route pattern but it does nothing, how come CM will not display the incoming call? I believed it was a direct call to the VDN and would start to get routed from there, at that point I got lost and is where I am at, I am still learning AVAYA by the way.
Regards,
David M
