[asterisk-users] Asterisk 1.8.7.0 connectivity to Avaya SM

Tue Feb 26 16:45:54 CST 2019

Thomas,

Does the Asterisk box need to do anything other than handle calls for this
one specific IVR? IE does it ever originate calls?

If it's only recieving calls then I'd turn on guest access and not even
bother with a peer.
Just set

[general]
context=transit-ivr
allowguest=yes

On Tue, Feb 26, 2019 at 3:13 PM Thomas Peters <TPeters at mcts.org> wrote:

> 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.
>
>
>
> Presently:
>
>    - 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 1.8.7.0 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
>
> *“asterisk” sip:asterisk@(null):0;tag=as682f2c53*
>
> as the “From” in the SIP header.
>
>    - He wants us to send
>
> *10.10.0.103 at mcts.org <10.10.0.103 at mcts.org>  *
>
> or more likely
>
> *<sip:10.10.0.103 at mcts.org <10.10.0.103 at mcts.org>>*
>
> instead.
>
>    - 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 1.8.7.0) 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:
>
>
>
> [general]
>
> ;context=transit-ivr
>
> context=incoming
>
> disallow=all
>
> allow=ulaw
>
> canreinvite=no
>
>
>
> [sipivr]
>
> host=dynamic
>
> secret=1NA6oZjTg1rjhZN8lArDgzLI7z8V2fxV
>
> type=peer
>
> ;context=transit-ivr
>
> context=incoming
>
> dtmfmode=inband
>
>
>
> The new section, with many failed experiments commented out, is after the
> [sipivr] section:
>
> [sessionmgr1]
>
> type=peer
>
> ;type=friend
>
> port=5060
>
> host=10.90.0.17
>
> dtmfmode=inband
>
> allowguest=yes
>
> qualify=yes
>
> realm=mcts.org
>
> promiscredir=yes
>
> ;Some have suggested using canreinvite=no with Avaya- didn't try that yet
>
> ;canreinvite=no
>
> canreinvite=yes
>
> transport=tcp
>
> ;context=incoming
>
> context=from-internal
>
> ;username=10.90.0.103
>
> fromdomain=mcts.org
>
> disallow=all
>
> allow=ulaw
>
> allow=alaw
>
> tcpenable=yes
>
> tcpbindaddr=0.0.0.0:5060
>
>
>
> 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[32142] chan_sip.c: Peer 'sessionmgr1' is now
> Reachable. (1ms / 2000ms)
>
>
>
> Can anyone help?
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
> Thomas M. Peters | Sr. Systems Administrator |  tpeters at mcts.org
> Desk: 414.343.1720 | Helpdesk: x3400 or  helpdesk at mcts.org
>
> *Milwaukee County Transit System <http://www.ridemcts.com/>*
>
>
>
> 1942 N 17th Street | Milwaukee, WI  53205
>
> Check us out on Facebook <https://www.facebook.com/mcts> & Twitter
> <https://twitter.com/RideMCTS>
>
>
> --
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