You have to check and verify the SIP trunk details, as ext to ext works once the pbx is up, but to call out, it should go through your provider.....so just recheck your provider's details. Regards Sandesh
On Wed, Oct 14, 2009 at 8:24 AM, Ott Rose <[email protected]> wrote: > here is the debug from the CLI. I think I know where the problem is I just > can figure out how to fix it. The IP in the From and To i think is where the > problem is. When I make an outbound call. i get the message "the call cannot > be completed as dialed". if i call another ext it works. I posted the debug > for both calls. > > > > > > > ==============outbound call=========================== > > <--- Transmitting (NAT) to 10.0.0.46:5060 ---> > SIP/2.0 183 Session Progress > Via: SIP/2.0/UDP 10.0.0.46:5060 > ;branch=z9hG4bKfd2143ede5319cf9b.273e4b904b0cc3101;received=10.0.0.46 > From: "ext" <sip:[email protected] <sip%[email protected]>>;tag=9d9e3944ba > To: "93214545" <sip:[email protected] <sip%[email protected]> > >;tag=as290bd498 > Call-ID: 401d30b0a1893e80 > CSeq: 13401 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Contact: <sip:[email protected] <sip%[email protected]>> > Content-Type: application/sdp > Content-Length: 254 > > v=0 > o=root 3609 3609 IN IP4 10.0.0.8 > s=session > c=IN IP4 10.0.0.8 > t=0 0 > m=audio 14398 RTP/AVP 0 8 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > > ===================================================== > > ================ext to ext=============================== > SIP/2.0 200 OK > Via: SIP/2.0/UDP 10.0.0.46:5060 > ;branch=z9hG4bK280378de3608b5bf1.2576ced30c6198b74;received=10.0.0.46 > From: "ext" <sip:[email protected] <sip%[email protected]>>;tag=d729237fcc > To: "111" <sip:[email protected] <sip%[email protected]>>;tag=as553ab5e9 > Call-ID: c7cc32657c620790 > CSeq: 8007 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Contact: <sip:[email protected] <sip%[email protected]>> > Content-Type: application/sdp > Content-Length: 254 > > v=0 > o=root 3609 3609 IN IP4 10.0.0.8 > s=session > c=IN IP4 10.0.0.8 > t=0 0 > m=audio 10414 RTP/AVP 0 8 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > > > ------------------------------ > Hotmail: Trusted email with Microsoft’s powerful SPAM protection. Sign up > now. <http://clk.atdmt.com/GBL/go/177141664/direct/01/> > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2009 - October 13 - 15 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
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