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....@mcts.org<mailto:10.10.0....@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....@mcts.org<mailto:10.10.0....@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 <viljo...@verishare.co.za> Sent: Thursday, 28 February 2019 08:18 To: 'asterisk-users@lists.digium.com' <asterisk-users@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....@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. -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
