<--- SIP read from 10.0.0.1:5060 ---> INVITE sip:23003@10.0.0.2;transport=UDP;TCID-0 SIP/2.0 From: ORAS;tag=a000001-13c4-16d2e-59298f8-249e To: Call-ID: 9794a14c-a000001-13c4-16d2e-59298f8-1920@10.0.0.1 CSeq: 1 INVITE Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK-16d2e-59298f8-6b7f Max-Forwards: 70 Contact: Content-Type: application/SDP Content-Length:288 v=0 o=a0000 1234567890 1234567890 IN IP4 10.0.0.1 s=SIP Call test c=IN IP4 10.0.0.1 t=0 0 m=audio 10002 RTP/AVP 8 97 0 18 a=fmtp:97 0-15 a=fmtp:18 annexb=yes a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 <-------------> --- (10 headers 13 lines) --- Sending to 10.0.0.1 : 5060 (no NAT) Using INVITE request as basis request - 9794a14c-a000001-13c4-16d2e-59298f8-1920@10.0.0.1 Found peer 'harris' Found RTP audio format 8 Found RTP audio format 97 Found RTP audio format 0 Found RTP audio format 18 Peer audio RTP is at port 10.0.0.1:10002 Found description format PCMA for ID 8 Found description format telephone-event for ID 97 Found description format PCMU for ID 0 Found description format G729 for ID 18 Capabilities: us - 0x3f1fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|jpeg|png|h261|h263|h263p|h264), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.0.0.1:10002 Looking for 23003 in default (domain 10.0.0.2) list_route: hop: <--- Transmitting (NAT) to 10.0.0.1:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK-16d2e-59298f8-6b7f;received=10.0.0.1 From: ORAS;tag=a000001-13c4-16d2e-59298f8-249e To: Call-ID: 9794a14c-a000001-13c4-16d2e-59298f8-1920@10.0.0.1 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Executing [23003@default:1] Dial("SIP/5060-0822ef78", "SIP/23003") in new stack -- Called 23003 -- SIP/23003-0823b6c0 is ringing <--- Transmitting (NAT) to 10.0.0.1:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK-16d2e-59298f8-6b7f;received=10.0.0.1 From: ORAS;tag=a000001-13c4-16d2e-59298f8-249e To: ;tag=as5779f29f Call-ID: 9794a14c-a000001-13c4-16d2e-59298f8-1920@10.0.0.1 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> Audio is at 10.0.0.2 port 11854 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Transmitting (NAT) to 10.0.0.1:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK-16d2e-59298f8-6b7f;received=10.0.0.1 From: ORAS;tag=a000001-13c4-16d2e-59298f8-249e To: ;tag=as5779f29f Call-ID: 9794a14c-a000001-13c4-16d2e-59298f8-1920@10.0.0.1 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 300 v=0 o=root 24756 24756 IN IP4 10.0.0.2 s=session c=IN IP4 10.0.0.2 t=0 0 m=audio 11854 RTP/AVP 8 0 18 97 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 =sendrecv <------------> -- Call on SIP/23003-0823b6c0 left from hold -- SIP/23003-0823b6c0 answered SIP/5060-0822ef78 Audio is at 10.0.0.2 port 11854 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (NAT) to 10.0.0.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK-16d2e-59298f8-6b7f;received=10.0.0.1 From: ORAS;tag=a000001-13c4-16d2e-59298f8-249e To: ;tag=as5779f29f Call-ID: 9794a14c-a000001-13c4-16d2e-59298f8-1920@10.0.0.1 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 300 v=0 o=root 24756 24757 IN IP4 10.0.0.2 s=session c=IN IP4 10.0.0.2 t=0 0 m=audio 11854 RTP/AVP 8 0 18 97 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> -- Packet2Packet bridging SIP/5060-0822ef78 and SIP/23003-0823b6c0 voip*CLI> <--- SIP read from 10.0.0.1:5060 ---> ACK sip:23003@10.0.0.2 SIP/2.0 From: ORAS;tag=a000001-13c4-16d2e-59298f8-249e To: ;tag=as5779f29f Call-ID: 9794a14c-a000001-13c4-16d2e-59298f8-1920@10.0.0.1 CSeq: 1 ACK Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK-16d38-592c144-274c Max-Forwards: 70 Contact: Content-Length:0 <-------------> --- (9 headers 0 lines) --- voip*CLI> <--- SIP read from 10.0.0.1:5060 ---> BYE sip:23003@10.0.0.2;TCID-0 SIP/2.0 From: ORAS;tag=a000001-13c4-16d2e-59298f8-249e To: ;tag=as5779f29f Call-ID: 9794a14c-a000001-13c4-16d2e-59298f8-1920@10.0.0.1 CSeq: 2 BYE Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK-16d38-592c14a-6caa Max-Forwards: 70 Content-Length:0 <-------------> --- (8 headers 0 lines) --- Sending to 10.0.0.1 : 5060 (NAT) <--- Transmitting (NAT) to 10.0.0.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK-16d38-592c14a-6caa;received=10.0.0.1 From: ORAS;tag=a000001-13c4-16d2e-59298f8-249e To: ;tag=as5779f29f Call-ID: 9794a14c-a000001-13c4-16d2e-59298f8-1920@10.0.0.1 CSeq: 2 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> == Spawn extension (default, 23003, 1) exited non-zero on 'SIP/5060-0822ef78' Really destroying SIP dialog '9794a14c-a000001-13c4-16d2e-59298f8-1920@10.0.0.1' Method: BYE voip*CLI>