[asterisk-users] Asterisk 220.127.116.11 connectivity to Avaya SM
TPeters at mcts.org
Tue Feb 26 16:11:26 CST 2019
Hello all, I hope someone can help me with this old Asterisk version. I have to run this version because of a custom IVR written on it. Porting it would take much too long and we'd have to hire a consultant because of all the hooks it has into Oracle databases and real-time information.
We have a brand-new Avaya phone system in place and we will be cutting over to it in late March 2019.
* We have an Asterisk 13.3.2 server with no phones registered to it, acting as a PSTN gateway. Calls come into it and get distributed to other Asterisk boxes with phones.
* If a call comes in from the provider marked as having been dialed as xxx-xxx-6711 (those are digits, not a pattern) it gets routed to the IVR box
* The IVR box runs Asterisk 18.104.22.168 and a custom IVR.
Where we have to get to:
* The new Avaya Session Manager has to have a working SIP trunk to the IVR so it can pass calls that come into xxx-xxx-6711 to it.
What the problem is:
* I don't fully understand what's going on here, neither how it works now, nor what I need to do to make Avaya's SM happy.
* When I do sip show peers on my IVR box, I see the Avaya session manager:
jerec*CLI> sip show peers
Name/username Host Dyn Forcerport ACL Port Status
sessionmgr1 10.10.0.17 5060 OK (1 ms)
* The Avaya engineer says he is seeing "SIP/2.0 400 Bad FROM header" in his trace screen, and his SM status screen shows "500 NOT REACHABLE" as the status for our IVR.
* He says we are sending
as the "From" in the SIP header.
* He wants us to send
10.10.0.103 at mcts.org<mailto:10.10.0.103 at mcts.org>
or more likely
<sip:10.10.0.103 at mcts.org<mailto:10.10.0.103 at mcts.org>>
* Pings from either end to the other work just fine.
* nmap doesn't show port 5060 open. It shows only port 22/tcp open. But then again, my main asterisk PBX doesn't show that port open either. So I don't think that means anything.
The IVR machine (Asterisk 22.214.171.124) sip.conf file has an old section for the old PSTN gateway, and a new section I just added for the session manager.
Old section for existing connections to the IVR:
The new section, with many failed experiments commented out, is after the [sipivr] section:
;Some have suggested using canreinvite=no with Avaya- didn't try that yet
Nothing I tried seems to make it stop sending asterisk@(null) in the header. This is supposed to be a sip trunk, not an extension, so I think I should NOT be user a username or secret. I'm not even sure what promiscredir does, or if it's helping or harming me.
There's virtually nothing in the logs about this connection, other than this:
[Feb 26 16:05:42] NOTICE chan_sip.c: Peer 'sessionmgr1' is now Reachable. (1ms / 2000ms)
Can anyone help?
Thomas M. Peters | Sr. Systems Administrator | tpeters at mcts.org<mailto:tpeters at mcts.org>
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