I just checked again to make sure. I am not seeing anything at all on gateway on failed calls. Again 2 out of 5 test calls were failed to reach gateway.
----- Original Message ----- From: "Paul Belanger" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[email protected]> Sent: Tuesday, August 09, 2005 5:33 PM Subject: Re: [Asterisk-Users] SIP-Trunk problem, Please help!!! > Can you see the INVITE if you put up a trace on your gateway > (209.XXX.XXX.113)? Asterisk is not getting anything back that is why it > retransmits 5 times. > > PB > > OMS wrote: > > INVITE sip:[EMAIL PROTECTED] SIP/2.0 > > Via: SIP/2.0/UDP 24.XX.XXX.101:5060;branch=z9hG4bK6adcaf4f > > From: "512538XXX" <sip:[EMAIL PROTECTED]>;tag=as5329d8fe > > To: <sip:[EMAIL PROTECTED]> > > Contact: <sip:[EMAIL PROTECTED]> > > Call-ID: [EMAIL PROTECTED] > > CSeq: 102 INVITE > > User-Agent: Asterisk PBX > > Date: Tue, 09 Aug 2005 20:36:43 GMT > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > > Content-Type: application/sdp > > Content-Length: 242 > > > > v=0 > > o=root 2251 2251 IN IP4 24.XX.XXX.101 > > s=session > > c=IN IP4 24.XX.XXX.101 > > t=0 0 > > m=audio 15202 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 - - - - > _______________________________________________ > Asterisk-Users mailing list > [email protected] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
