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