[asterisk-users] Asterisk 1.8.7.0 connectivity to Avaya SM (Thomas Peters)

[Stefan Viljoen]

E. g. You can also then try in /etc/hosts to put

10.10.0.103 mcts.org

e. g. if any local reverse lookup is done "10.10.0.103" resolves to "mcts.org" and the converse for normal DNS lookups.

You can even try in /etc/asterisk/asterisk.conf to make the "systemname" line be

systemname=10.10.0.103

This is also why if you have 

systemname=mcts.org

in /etc/asterisk/asterisk.conf a DNS lookup on "mcts.org" on the linux box itself will give "10.10.0.103" or "127.0.0.1" depending on what is in /etc/hosts, e. g. to try and force your Asterisk instance if it needs / tries to ID itself when sending SIP messages it will identify itself as "10.10.0.103" in one part of the sip message, while you also set its domain name to be "mcts.org" hopefully forcing it to put that to the right of the "@" in 

10.10.0.103 at mcts.org<mailto:10.10.0.103 at mcts.org>

Also, does the Avaya device behave at all like a Linux / WIndows machine e. g. maybe it needs to do R-DNS on itself as well, e. g. if the Avaya people on their side can edit their version of the Linux "/etc/hosts" and put in there

10.10.0.103 mcts.org

so that if THEY do a R-DNS lookup on "mcts.org" (if your Asterisk co-operates and passes this in to them in SIP, see above & previous message) and they do reverse dns lookup on it they get "10.10.0.103" in

10.10.0.103 at mcts.org<mailto:10.10.0.103 at mcts.org>

All conjecture and guesswork though, but maybe something makes a light-bulb light up...

Not probably relevant, but I had once to do this with Asterisk 1.8.32.3 in much the same situation, e. g. we had a 2nd Linux box running ViciDial (bascially also Asterisk) that would sometimes dial through to "our" Asterisk server to get outgoing lines.

I put together this in "our" Asterisk server's sip.conf:

[9090] ;This is the SIP identity VICIDial uses to dial out when it wants to
type=peer
user=9090
host=10.1.51.174
disallow=all
allow=g729
allow=alaw
allow=ulaw
context=local
call-limit=20
limitonpeer=yes
canreinvite=yes
qualify=no
deny=0.0.0.0/0.0.0.0
permit=10.1.51.174/255.0.0.0

This caused "our" Asterisk to treat "their" Asterisk as just another SIP device, e. g. no difference at all between it and a normal VOIP phone on the network as far as "our" Asterisk was concerned. "Our" Asterisk could not distinguish if it was getting a call from a phone, or from this ViciDial server.

Hope some of this helps.

Regards

Stefan

-----Original Message-----
From: Stefan Viljoen <viljoens at verishare.co.za> 
Sent: Thursday, 28 February 2019 08:18
To: 'asterisk-users at lists.digium.com' <asterisk-users at lists.digium.com>
Subject: Asterisk 1.8.7.0 connectivity to Avaya SM (Thomas Peters)
 
Hi Thomas

What is your IVR box's domain name in Linux?

With a hostname of, for example, "mcts.org" do you have a line like this in /etc/hosts:

127.0.0.1	mcts.org

in your /etc/hosts?

Additionally, in /etc/asterisk/asterisk.conf, is there a line

systemname =

that is -uncommented- and contains the string you need to the right of the @ e. g.

mcts.org

?

The immediate above just a shot in the dark but could affect how your IVR box identifies itself.

If the Linux system host name of the IVR machine is not "mcts.org" maybe you can change the machine's name via (for Centos 7 at least - if you have NetworManager running):

nmcli general hostname mcts.org
systemctl restart systemd-hostnamed

and then reboot the machine which implies also restarting asterisk.

If I remember right this will cause the Asterisk instance to identify itself as 

"mcts.org" sip:mcts.org at mcts.org:0;tag=...
                                ^               ^
                                |                |
>From asterisk.conf                 |
                                         From nmcli general hostname mcts.org

?

Might still not solve your precise issue but might be a step in the right direction??

Stefan
---
>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.