Any idea whats wrong ? On Fri, Feb 20, 2009 at 2:32 AM, David @ULC <[email protected]> wrote:
> > --- (12 headers 0 lines) --- > Sending to 192.168.0.50 : 12714 (NAT) > Transmitting (NAT) to 192.168.0.50:12714: > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.0.50:12714 > ;branch=z9hG4bK-d87543-930550325154e53d-1--d87543-;received=192.168.0.50;rport=12714 > From: "cc106"<sip:[email protected] <sip%[email protected]> > >;tag=7f1cff22 > To: > "817275691533"<sip:[email protected]<sip%[email protected]> > >;tag=as02559696 > Call-ID: NmZlY2E3ZDk0MDVmN2M1MGVkOGJlOTBiYjg5ODIxNTU. > CSeq: 3 BYE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Contact: <sip:[email protected] <sip%[email protected]>> > Content-Length: 0 > X-Asterisk-HangupCause: Normal Clearing > > > > --- > Scheduling destruction of call > '[email protected]' in 32000 ms > set_destination: Parsing <sip:8.14.xxx.xxx:5060;transport=udp> for > address/port to send to > set_destination: set destination to 8.14.xxx.xxx, port 5060 > Reliably Transmitting (no NAT) to 8.14.xxx.xxx:5060: > BYE sip:8.14.xxx.xxx:5060;transport=udp SIP/2.0 > Via: SIP/2.0/UDP 59.xxx.xx.xx:5060;branch=z9hG4bK59c0212a;rport > From: "cc106" <sip:[email protected]>;tag=as3f9466a7 > To: <sip:[email protected]>;tag=1902000923108720995156225 > Call-ID: [email protected] > CSeq: 103 BYE > User-Agent: Asterisk PBX > Max-Forwards: 70 > Content-Length: 0 > > > --- > == Spawn extension (default, 817275691533, 2) exited non-zero on > 'SIP/cc106-b7a1a9d0' > -- Executing DeadAGI("SIP/cc106-b7a1a9d0", "agi:// > 127.0.0.1:4577/call_log") in new stack > -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 > -- Executing DeadAGI("SIP/cc106-b7a1a9d0", "agi:// > 127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----16-----12)") > in new stack > -- AGI Script agi:// > 127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----16-----12) > completed, returning 0 > Destroying call 'NmZlY2E3ZDk0MDVmN2M1MGVkOGJlOTBiYjg5ODIxNTU.' > vicidialnow*CLI> > <-- SIP read from 8.14.xxx.xxx:5060: > SIP/2.0 200 OK > CSeq: 102 INVITE > Via: SIP/2.0/UDP 59.xxx.xx.xx:5060;branch=z9hG4bK3a111ef4;rport > From: "V0219160007000134649" <sip:[email protected]>;tag=as79fae976 > Call-ID: [email protected] > To: <sip:[email protected]>;tag=1902000923098720982816221 > Contact: <sip:8.14.xxx.xxx:5060;transport=udp> > Content-Type: application/sdp > Content-Length: 225 > > v=0 > o=VoipSwitch 7220 7220 IN IP4 8.14.xxx.xxx > s=VoipSIP > i=Audio Session > c=IN IP4 8.14.xxx.xxx > t=0 0 > m=audio 6220 RTP/AVP 18 101 > a=rtpmap:18 G729/8000/1 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=sendrecv > > --- (9 headers 11 lines) --- > Found RTP audio format 18 > Found RTP audio format 101 > Peer audio RTP is at port 8.14.xxx.xxx:6220 > Found description format G729 > Found description format telephone-event > Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x100 (g729)/video=0x0 > (nothing), combined - 0x100 (g729) > Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 > (telephone-event), combined - 0x1 (telephone-event) > list_route: hop: <sip:8.14.xxx.xxx:5060;transport=udp> > set_destination: Parsing <sip:8.14.xxx.xxx:5060;transport=udp> for > address/port to send to > set_destination: set destination to 8.14.xxx.xxx, port 5060 > Transmitting (no NAT) to 8.14.xxx.xxx:5060: > ACK sip:8.14.xxx.xxx:5060;transport=udp SIP/2.0 > Via: SIP/2.0/UDP 59.xxx.xx.xx:5060;branch=z9hG4bK6eef7893;rport > From: "V0219160007000134649" <sip:[email protected]>;tag=as79fae976 > To: <sip:[email protected]>;tag=1902000923098720982816221 > Contact: <sip:[email protected]> > Call-ID: [email protected] > CSeq: 102 ACK > User-Agent: Asterisk PBX > Max-Forwards: 70 > Content-Length: 0 > > >
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