Dale, Sorry for taking so long to answer, I've been traveling.
Thanks so much for the suggestion, your solution worked perfectly. I'm not sure why I didn't notice that the IAX trunk was working in the other direction. Once again, thanks for your help. Mitch Date: Mon, 25 Jun 2012 05:44:37 -0500 From: Dale Noll <dn...@wi.rr.com> Subject: Re: [asterisk-users] IAX Trunk issue. To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <4fe84115.60...@wi.rr.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed On 06/24/2012 07:53 PM, Mitchell Johnson wrote: > I'm testing a few IAX trunk scenarios in a controlled lab. From server2 > extension 5000 (Server IP Address 172.16.200.212) I dial 6001 which goes > across the IAX trunk to server 1 (IP address 172.16.200.210). Instead of > ringing the 6001 phone, it plays tt-weasels (the s extension). When I dial > 6099 it also plays tt-weasels as it's supposed to, but it's not the > tt-weasels under its extension. It also dials the s extension. > > I only placed the s extension in the dial plan to verify that the traffic was > going across the IAX trunk and hitting the correct context. > > Any help would be greatly appreciated. > > Thanks Mitch > > > > [phones] > exten => _60XX,1,Dial(IAX2/trunk-1) > exten => _X.,1,Dial(IAX2/trunk-1) > exten => 5000,1,Dial(SIP/${EXTEN}) > exten => 5000,n,Hangup > same => n,Hangup() > exten => 5099,1,Playback(tt-monkeys) > exten => 5099,n,HangUp You are not telling asterisk-1 where you want the call to go, so it is going to 's'. Try adding the extension to the Dial() command on asterisk-2. Change Dial(IAX2/trunk-1) to Dial(IAX2/trunk-1/${EXTEN}) Note: It appears that you are doing it correctly from asterisk-1 towards asterisk-2 exten => _5XXX,1,Dial(${IAXTrunk}/${EXTEN}) Assuming, of course, that the variable IAXTrunk is properly set. Dale -- "The truth speaks for itself. I'm just the messenger." Lyta Alexander - Babylon 5
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users