[asterisk-users] Asterisk 1.8.7.0 connectivity to Avaya SM

Wed Feb 27 08:16:51 CST 2019

Thanks for the reply John.

About 85-90% of what this box has to do is just handle calls, but it also has options to transfer calls to the main phone system, which up to now has been another asterisk box. For example, you can hit 6 to be transferred to the Lost & Found Department.

I do have allowguest set to “yes” already, but of course I also have type=peer and the other stuff for a sip trunk.

The Avaya engineer is telling me that I need to change my “From” header, and I don’t know how to do that.

I have to figure this out in the next few days, or I’m in deep doo-doo.

-T

Thomas M. Peters | Sr. Systems Administrator |  tpeters at mcts.org<mailto:tpeters at mcts.org>
Desk: 414.343.1720 | Helpdesk: x3400 or  helpdesk at mcts.org<mailto: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>

From: asterisk-users <asterisk-users-bounces at lists.digium.com> On Behalf Of John Kiniston
Sent: Tuesday, February 26, 2019 4:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com>
Subject: Re: [asterisk-users] Asterisk 1.8.7.0 connectivity to Avaya SM

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<mailto: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<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>>

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<http://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<http://mcts.org>
disallow=all
allow=ulaw
allow=alaw
tcpenable=yes
tcpbindaddr=0.0.0.0:5060<http://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<mailto:tpeters at mcts.org>
Desk: 414.343.1720 | Helpdesk: x3400 or  helpdesk at mcts.org<mailto: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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