search for: iaxtrunk

...n is to start simple. Get the servers talking with each other, successfully pass calls between themselves then add in the security layers. Dial plan routing would probably be the same or close to it, you could keep the same password or make it different for each server, just be mindful of the IAX2/iaxtrunk:1234@XXX.187.142.204/${EXTEN} statement in the exten => Dial command, pass the correct username:password@ipaddress to the correct server and you should be fine. JR >Hi JR > Thanks for the mail , I am trying out Asterisk >and learning it , U mentioned that if all the tree >b...

...at=yes extensions.conf [internal_users] exten => 6000,1,Answer() exten => 6000,2,Playback(hello-world) exten => 6000,3,Hangup() exten => 6001,1,Dial(SIP/6001) exten => 6002,1,Dial(SIP/6002) exten => 6099,1,Playback(tt-weasels) exten => 6099,n,HangUp exten => _5XXX,1,Dial(${IAXTrunk}/${EXTEN}) same => n,Hangup() exten => s,1,Answer() exten => s,n,Playback(tt-weasels) exten => s,n,Hangup() IAX.conf [trunk-1] type=friend username=trunk-1 trunk=yes requiretoken=no secret=password host=172.16.200.212 context=internal_users auth=plaintext disallow=all ;allow=ulaw ;all...

...re 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium...

I'vce got three Asterisk systems here that I'd like to be able to place calls between with IAX. As usual, I've spent several hours playing with it, really getting nowhere. Asterisk is so mentally draining. Each system, pbx1, pbx2, pbx3, should be able to connect to every other. Do I need separate user/peers or can I combine them into a single user=friend for each system? if I place a

...itExten() The ringing sound doesn't work for any extension if I use this one. I just get silence until someone answers. How come? I use Asterisk 1.4.10. I have attached my extensions.conf file to this email. Thanks, Simon -------------- next part -------------- [globals] SIPTRUNK=4185555555 IAXTRUNK=5145555555 [default] exten => s,1,Answer() exten => s,2,Background(viagenie) exten => s,3,WaitExten() exten => i,1,Background(invalid) exten => i,n,Goto(s,1) exten => t,1,Background(please-try-again) exten => t,n,Goto(s,1) [phones] exten => 101,1,Dial(SIP/101,15) exten =...