On Tue, Oct 21, 2008 at 9:56 AM, Juan Rodríguez <[EMAIL PROTECTED]> wrote:
> Try changing:
> exten => 101,1,Dial(SIP/101/20)
> to
> exten => 101,1,Dial(SIP/101|20) or exten => 101,1,Dial(SIP/101,20)
>
> because exten => 101,1,Dial(SIP/101/20) means you are trying to contact ext.
> 20 on through a trunk called 101.
Oh, typo, but that still didn't cure it....
Successful call from from 101 to 102
== Using SIP RTP CoS mark 5
-- Executing [EMAIL PROTECTED]:1] Dial("SIP/101-08220318",
"SIP/102,20") in new stack
== Using SIP RTP CoS mark 5
-- Called 102
-- SIP/102-08221a78 is ringing
-- SIP/102-08221a78 answered SIP/101-08220318
-- Packet2Packet bridging SIP/101-08220318 and SIP/102-08221a78
== Spawn extension (default, 102, 1) exited non-zero on 'SIP/101-08220318'
Failed call from 102 to 101
== Using SIP RTP CoS mark 5
-- Executing [EMAIL PROTECTED]:1] Dial("SIP/102-08221a78",
"SIP/101,20") in new stack
== Using SIP RTP CoS mark 5
-- Called 101
-- Got SIP response 400 "Bad Request" back from 68.156.63.118
-- SIP/101-0821e110 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [EMAIL PROTECTED]:2] Hangup("SIP/102-08221a78", "") in new
stack
== Spawn extension (default, 101, 2) exited non-zero on 'SIP/102-08221a78'
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users