[Mar 10 18:40:34] VERBOSE[11989] chan_sip.c: <--- SIP read from UDP:172.17.15.166:5062 ---> INVITE sip:31@172.17.15.245 SIP/2.0 Via: SIP/2.0/UDP 172.17.15.166:5062;branch=z9hG4bKa26f0dfe1007db6b From: "Ismailov Rafael" ;tag=04720d5e2e970e75 To: Contact: Supported: replaces, timer, path Call-ID: 0cac0c52f9273627@172.17.15.166 CSeq: 25528 INVITE User-Agent: Grandstream GXP1200 1.2.1.4 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 348 v=0 o=38 8000 8000 IN IP4 172.17.15.166 s=SIP Call c=IN IP4 172.17.15.166 t=0 0 m=audio 5048 RTP/AVP 0 8 4 18 2 97 9 3 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=rtpmap:9 G722/8000 a=rtpmap:3 GSM/8000 a=ptime:20 <-------------> [Mar 10 18:40:34] VERBOSE[11989] chan_sip.c: --- (13 headers 17 lines) --- [Mar 10 18:40:34] VERBOSE[11989] chan_sip.c: Sending to 172.17.15.166 : 5062 (no NAT) [Mar 10 18:40:34] VERBOSE[11989] chan_sip.c: Using INVITE request as basis request - 0cac0c52f9273627@172.17.15.166 [Mar 10 18:40:34] VERBOSE[11989] chan_sip.c: Found peer '38' for '38' from 172.17.15.166:5062 [Mar 10 18:40:34] VERBOSE[11989] chan_sip.c: Found RTP audio format 0 [Mar 10 18:40:34] VERBOSE[11989] chan_sip.c: Found RTP audio format 8 [Mar 10 18:40:34] VERBOSE[11989] chan_sip.c: Found RTP audio format 4 [Mar 10 18:40:34] VERBOSE[11989] chan_sip.c: Found RTP audio format 18 [Mar 10 18:40:34] VERBOSE[11989] chan_sip.c: Found RTP audio format 2 [Mar 10 18:40:34] VERBOSE[11989] chan_sip.c: Found RTP audio format 97 [Mar 10 18:40:34] VERBOSE[11989] chan_sip.c: Found RTP audio format 9 [Mar 10 18:40:34] VERBOSE[11989] chan_sip.c: Found RTP audio format 3 [Mar 10 18:40:34] VERBOSE[11989] chan_sip.c: Found audio description format PCMU for ID 0 [Mar 10 18:40:34] VERBOSE[11989] chan_sip.c: Found audio description format PCMA for ID 8 [Mar 10 18:40:34] VERBOSE[11989] chan_sip.c: Found audio description format G723 for ID 4 [Mar 10 18:40:34] VERBOSE[11989] chan_sip.c: Found audio description format G729 for ID 18 [Mar 10 18:40:34] VERBOSE[11989] chan_sip.c: Found audio description format G726-32 for ID 2 [Mar 10 18:40:34] VERBOSE[11989] chan_sip.c: Found audio description format iLBC for ID 97 [Mar 10 18:40:34] VERBOSE[11989] chan_sip.c: Found audio description format G722 for ID 9 [Mar 10 18:40:34] VERBOSE[11989] chan_sip.c: Found audio description format GSM for ID 3 [Mar 10 18:40:34] VERBOSE[11989] chan_sip.c: Capabilities: us - 0x4 (ulaw), peer - audio=0x1d0f (g723|gsm|ulaw|alaw|g726|g729|ilbc|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [Mar 10 18:40:34] VERBOSE[11989] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing) [Mar 10 18:40:34] VERBOSE[11989] chan_sip.c: Peer audio RTP is at port 172.17.15.166:5048 [Mar 10 18:40:34] VERBOSE[11989] chan_sip.c: Looking for 31 in DLPN_To_Town (domain 172.17.15.245) [Mar 10 18:40:34] VERBOSE[11989] chan_sip.c: list_route: hop: [Mar 10 18:40:34] VERBOSE[11989] chan_sip.c: <--- Transmitting (no NAT) to 172.17.15.166:5062 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.17.15.166:5062;branch=z9hG4bKa26f0dfe1007db6b;received=172.17.15.166 From: "Ismailov Rafael" ;tag=04720d5e2e970e75 To: Call-ID: 0cac0c52f9273627@172.17.15.166 CSeq: 25528 INVITE Server: Asterisk PBX SVN-trunk-r251310 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> [Mar 10 18:40:34] VERBOSE[11994] chan_sip.c: Audio is at 172.17.15.245 port 15046 [Mar 10 18:40:34] VERBOSE[11994] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Mar 10 18:40:34] VERBOSE[11994] chan_sip.c: <--- Reliably Transmitting (no NAT) to 172.17.15.166:5062 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.17.15.166:5062;branch=z9hG4bKa26f0dfe1007db6b;received=172.17.15.166 From: "Ismailov Rafael" ;tag=04720d5e2e970e75 To: ;tag=as00a805ed Call-ID: 0cac0c52f9273627@172.17.15.166 CSeq: 25528 INVITE Server: Asterisk PBX SVN-trunk-r251310 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Content-Length: 219 v=0 o=root 1532928926 1532928926 IN IP4 172.17.15.245 s=Asterisk PBX SVN-trunk-r251310 c=IN IP4 172.17.15.245 t=0 0 m=audio 15046 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Mar 10 18:40:34] VERBOSE[11989] chan_sip.c: <--- SIP read from UDP:172.17.15.166:5062 ---> ACK sip:31@172.17.15.245 SIP/2.0 Via: SIP/2.0/UDP 172.17.15.166:5062;branch=z9hG4bKfb7e40e2eb162816 From: "Ismailov Rafael" ;tag=04720d5e2e970e75 To: ;tag=as00a805ed Contact: Supported: path Call-ID: 0cac0c52f9273627@172.17.15.166 CSeq: 25528 ACK User-Agent: Grandstream GXP1200 1.2.1.4 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 <-------------> [Mar 10 18:40:34] VERBOSE[11989] chan_sip.c: --- (12 headers 0 lines) --- [Mar 10 18:40:34] VERBOSE[11994] chan_sip.c: Audio is at 172.17.15.245 port 17430 [Mar 10 18:40:34] VERBOSE[11994] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Mar 10 18:40:34] VERBOSE[11994] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Mar 10 18:40:34] VERBOSE[11994] chan_sip.c: Reliably Transmitting (no NAT) to 172.17.15.156:5060: INVITE sip:31@172.17.15.156:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 172.17.15.245:5060;branch=z9hG4bK07269d9a Max-Forwards: 70 From: "Test" ;tag=as331324f7 To: Contact: Call-ID: 2e2f170d0ef48834251c4eee0b9610f2@172.17.15.245 CSeq: 102 INVITE User-Agent: Asterisk PBX SVN-trunk-r251310 Date: Wed, 10 Mar 2010 16:40:34 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Remote-Party-ID: "Test" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 273 v=0 o=root 944702901 944702901 IN IP4 172.17.15.245 s=Asterisk PBX SVN-trunk-r251310 c=IN IP4 172.17.15.245 t=0 0 m=audio 17430 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Mar 10 18:40:34] VERBOSE[11989] chan_sip.c: <--- SIP read from UDP:172.17.15.156:52462 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.17.15.245:5060;branch=z9hG4bK07269d9a From: "Test" ;tag=as331324f7 To: Call-ID: 2e2f170d0ef48834251c4eee0b9610f2@172.17.15.245 Date: Wed, 10 Mar 2010 16:40:40 GMT CSeq: 102 INVITE Server: Cisco-CP7960G/8.0 Contact: Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE Content-Length: 0 <-------------> [Mar 10 18:40:34] VERBOSE[11989] chan_sip.c: --- (11 headers 0 lines) --- [Mar 10 18:40:35] VERBOSE[11989] chan_sip.c: <--- SIP read from UDP:172.17.15.156:52463 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 172.17.15.245:5060;branch=z9hG4bK07269d9a From: "Test" ;tag=as331324f7 To: ;tag=0007509810c807492a7f72c6-2ab209aa Call-ID: 2e2f170d0ef48834251c4eee0b9610f2@172.17.15.245 Date: Wed, 10 Mar 2010 16:40:40 GMT CSeq: 102 INVITE Server: Cisco-CP7960G/8.0 Contact: Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE Remote-Party-ID: "Ismailov Rafael" ;party=called;id-type=subscriber;privacy=off;screen=yes Content-Length: 0 <-------------> [Mar 10 18:40:35] VERBOSE[11989] chan_sip.c: --- (12 headers 0 lines) --- [Mar 10 18:40:35] VERBOSE[11994] chan_sip.c: set_destination: Parsing for address/port to send to [Mar 10 18:40:35] VERBOSE[11994] chan_sip.c: set_destination: set destination to 172.17.15.166, port 5062 [Mar 10 18:40:35] VERBOSE[11994] chan_sip.c: Audio is at 172.17.15.245 port 15046 [Mar 10 18:40:35] VERBOSE[11994] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Mar 10 18:40:35] VERBOSE[11994] chan_sip.c: Reliably Transmitting (no NAT) to 172.17.15.166:5062: INVITE sip:38@172.17.15.166:5062;transport=udp SIP/2.0 Via: SIP/2.0/UDP 172.17.15.245:5060;branch=z9hG4bK29622931 Max-Forwards: 70 From: ;tag=as00a805ed To: "Ismailov Rafael" ;tag=04720d5e2e970e75 Contact: Call-ID: 0cac0c52f9273627@172.17.15.166 CSeq: 102 INVITE User-Agent: Asterisk PBX SVN-trunk-r251310 Require: timer Session-Expires: 1800;refresher=uas Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Remote-Party-ID: "Ismailov Rafael" ;party=called;privacy=off;screen=yes Content-Type: application/sdp Content-Length: 219 v=0 o=root 1532928926 1532928927 IN IP4 172.17.15.245 s=Asterisk PBX SVN-trunk-r251310 c=IN IP4 172.17.15.245 t=0 0 m=audio 15046 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Mar 10 18:40:35] VERBOSE[11989] chan_sip.c: <--- SIP read from UDP:172.17.15.166:5062 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.17.15.245:5060;branch=z9hG4bK29622931 From: ;tag=as00a805ed To: "Ismailov Rafael" ;tag=04720d5e2e970e75 Call-ID: 0cac0c52f9273627@172.17.15.166 CSeq: 102 INVITE User-Agent: Grandstream GXP1200 1.2.1.4 Session-Expires: 180;refresher=uas Min-SE: 180 Require: timer Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Supported: replaces, timer Content-Length: 155 v=0 o=38 8000 8001 IN IP4 172.17.15.166 s=SIP Call c=IN IP4 172.17.15.166 t=0 0 m=audio 5048 RTP/AVP 0 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 <-------------> [Mar 10 18:40:35] VERBOSE[11989] chan_sip.c: --- (15 headers 9 lines) --- [Mar 10 18:40:35] VERBOSE[11989] chan_sip.c: Found RTP audio format 0 [Mar 10 18:40:35] VERBOSE[11989] chan_sip.c: Found audio description format PCMU for ID 0 [Mar 10 18:40:35] VERBOSE[11989] chan_sip.c: Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [Mar 10 18:40:35] VERBOSE[11989] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing) [Mar 10 18:40:35] VERBOSE[11989] chan_sip.c: Peer audio RTP is at port 172.17.15.166:5048 [Mar 10 18:40:35] VERBOSE[11989] chan_sip.c: set_destination: Parsing for address/port to send to [Mar 10 18:40:35] VERBOSE[11989] chan_sip.c: set_destination: set destination to 172.17.15.166, port 5062 [Mar 10 18:40:35] VERBOSE[11989] chan_sip.c: Transmitting (no NAT) to 172.17.15.166:5062: ACK sip:38@172.17.15.166:5062;transport=udp SIP/2.0 Via: SIP/2.0/UDP 172.17.15.245:5060;branch=z9hG4bK11865d1f Max-Forwards: 70 From: ;tag=as00a805ed To: "Ismailov Rafael" ;tag=04720d5e2e970e75 Contact: Call-ID: 0cac0c52f9273627@172.17.15.166 CSeq: 102 ACK User-Agent: Asterisk PBX SVN-trunk-r251310 Content-Length: 0 --- [Mar 10 18:40:37] VERBOSE[11989] chan_sip.c: <--- SIP read from UDP:172.17.15.156:52464 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.17.15.245:5060;branch=z9hG4bK07269d9a From: "Test" ;tag=as331324f7 To: ;tag=0007509810c807492a7f72c6-2ab209aa Call-ID: 2e2f170d0ef48834251c4eee0b9610f2@172.17.15.245 Date: Wed, 10 Mar 2010 16:40:43 GMT CSeq: 102 INVITE Server: Cisco-CP7960G/8.0 Contact: Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE Remote-Party-ID: "Ismailov Rafael" ;party=called;id-type=subscriber;privacy=off;screen=yes Supported: replaces,join,norefersub Content-Length: 206 Content-Type: application/sdp Content-Disposition: session;handling=optional v=0 o=Cisco-SIPUA 3975 0 IN IP4 172.17.15.156 s=SIP Call t=0 0 m=audio 17588 RTP/AVP 0 101 c=IN IP4 172.17.15.156 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv <-------------> [Mar 10 18:40:37] VERBOSE[11989] chan_sip.c: --- (15 headers 10 lines) --- [Mar 10 18:40:37] VERBOSE[11989] chan_sip.c: Found RTP audio format 0 [Mar 10 18:40:37] VERBOSE[11989] chan_sip.c: Found RTP audio format 101 [Mar 10 18:40:37] VERBOSE[11989] chan_sip.c: Found audio description format PCMU for ID 0 [Mar 10 18:40:37] VERBOSE[11989] chan_sip.c: Found audio description format telephone-event for ID 101 [Mar 10 18:40:37] VERBOSE[11989] chan_sip.c: Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [Mar 10 18:40:37] VERBOSE[11989] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Mar 10 18:40:37] VERBOSE[11989] chan_sip.c: Peer audio RTP is at port 172.17.15.156:17588 [Mar 10 18:40:37] VERBOSE[11989] chan_sip.c: list_route: hop: [Mar 10 18:40:37] VERBOSE[11989] chan_sip.c: set_destination: Parsing for address/port to send to [Mar 10 18:40:37] VERBOSE[11989] chan_sip.c: set_destination: set destination to 172.17.15.156, port 5060 [Mar 10 18:40:37] VERBOSE[11989] chan_sip.c: Transmitting (no NAT) to 172.17.15.156:5060: ACK sip:31@172.17.15.156:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 172.17.15.245:5060;branch=z9hG4bK4f6807a8 Max-Forwards: 70 From: "Test" ;tag=as331324f7 To: ;tag=0007509810c807492a7f72c6-2ab209aa Contact: Call-ID: 2e2f170d0ef48834251c4eee0b9610f2@172.17.15.245 CSeq: 102 ACK User-Agent: Asterisk PBX SVN-trunk-r251310 Content-Length: 0 --- [Mar 10 18:40:37] VERBOSE[11994] chan_sip.c: set_destination: Parsing for address/port to send to [Mar 10 18:40:37] VERBOSE[11994] chan_sip.c: set_destination: set destination to 172.17.15.166, port 5062 [Mar 10 18:40:37] VERBOSE[11994] chan_sip.c: Audio is at 172.17.15.245 port 15046 [Mar 10 18:40:37] VERBOSE[11994] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Mar 10 18:40:37] VERBOSE[11994] chan_sip.c: Reliably Transmitting (no NAT) to 172.17.15.166:5062: INVITE sip:38@172.17.15.166:5062;transport=udp SIP/2.0 Via: SIP/2.0/UDP 172.17.15.245:5060;branch=z9hG4bK2bba5bbf Max-Forwards: 70 From: ;tag=as00a805ed To: "Ismailov Rafael" ;tag=04720d5e2e970e75 Contact: Call-ID: 0cac0c52f9273627@172.17.15.166 CSeq: 103 INVITE User-Agent: Asterisk PBX SVN-trunk-r251310 Require: timer Session-Expires: 1800;refresher=uas Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Remote-Party-ID: "Ismailov Rafael" ;party=calling;privacy=off;screen=yes Content-Type: application/sdp Content-Length: 219 v=0 o=root 1532928926 1532928928 IN IP4 172.17.15.245 s=Asterisk PBX SVN-trunk-r251310 c=IN IP4 172.17.15.245 t=0 0 m=audio 15046 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Mar 10 18:40:37] VERBOSE[11989] chan_sip.c: <--- SIP read from UDP:172.17.15.166:5062 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.17.15.245:5060;branch=z9hG4bK2bba5bbf From: ;tag=as00a805ed To: "Ismailov Rafael" ;tag=04720d5e2e970e75 Call-ID: 0cac0c52f9273627@172.17.15.166 CSeq: 103 INVITE User-Agent: Grandstream GXP1200 1.2.1.4 Session-Expires: 180;refresher=uas Min-SE: 180 Require: timer Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Supported: replaces, timer Content-Length: 155 v=0 o=38 8000 8002 IN IP4 172.17.15.166 s=SIP Call c=IN IP4 172.17.15.166 t=0 0 m=audio 5048 RTP/AVP 0 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 <-------------> [Mar 10 18:40:37] VERBOSE[11989] chan_sip.c: --- (15 headers 9 lines) --- [Mar 10 18:40:37] VERBOSE[11989] chan_sip.c: Found RTP audio format 0 [Mar 10 18:40:37] VERBOSE[11989] chan_sip.c: Found audio description format PCMU for ID 0 [Mar 10 18:40:37] VERBOSE[11989] chan_sip.c: Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [Mar 10 18:40:37] VERBOSE[11989] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing) [Mar 10 18:40:37] VERBOSE[11989] chan_sip.c: Peer audio RTP is at port 172.17.15.166:5048 [Mar 10 18:40:37] VERBOSE[11989] chan_sip.c: set_destination: Parsing for address/port to send to [Mar 10 18:40:37] VERBOSE[11989] chan_sip.c: set_destination: set destination to 172.17.15.166, port 5062 [Mar 10 18:40:37] VERBOSE[11989] chan_sip.c: Transmitting (no NAT) to 172.17.15.166:5062: ACK sip:38@172.17.15.166:5062;transport=udp SIP/2.0 Via: SIP/2.0/UDP 172.17.15.245:5060;branch=z9hG4bK096f9303 Max-Forwards: 70 From: ;tag=as00a805ed To: "Ismailov Rafael" ;tag=04720d5e2e970e75 Contact: Call-ID: 0cac0c52f9273627@172.17.15.166 CSeq: 103 ACK User-Agent: Asterisk PBX SVN-trunk-r251310 Content-Length: 0 --- [Mar 10 18:40:42] VERBOSE[11989] chan_sip.c: <--- SIP read from UDP:172.17.15.156:50239 ---> INVITE sip:38@172.17.15.245 SIP/2.0 Via: SIP/2.0/UDP 172.17.15.156:5060;branch=z9hG4bK2329dcd5 From: ;tag=0007509810c807492a7f72c6-2ab209aa To: "Test" ;tag=as331324f7 Call-ID: 2e2f170d0ef48834251c4eee0b9610f2@172.17.15.245 Max-Forwards: 70 Date: Wed, 10 Mar 2010 16:40:47 GMT CSeq: 101 INVITE User-Agent: Cisco-CP7960G/8.0 Contact: Accept: application/sdp Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE Remote-Party-ID: "Ismailov Rafael" ;party=called;id-type=subscriber;privacy=off;screen=yes Supported: replaces,join,norefersub Content-Length: 277 Content-Type: application/sdp Content-Disposition: session;handling=optional v=0 o=Cisco-SIPUA 3975 1 IN IP4 172.17.15.156 s=SIP Call t=0 0 m=audio 17588 RTP/AVP 0 8 18 101 c=IN IP4 172.17.15.156 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendonly <-------------> [Mar 10 18:40:42] VERBOSE[11989] chan_sip.c: --- (17 headers 13 lines) --- [Mar 10 18:40:42] VERBOSE[11989] chan_sip.c: Sending to 172.17.15.156 : 5060 (no NAT) [Mar 10 18:40:42] VERBOSE[11989] chan_sip.c: Found RTP audio format 0 [Mar 10 18:40:42] VERBOSE[11989] chan_sip.c: Found RTP audio format 8 [Mar 10 18:40:42] VERBOSE[11989] chan_sip.c: Found RTP audio format 18 [Mar 10 18:40:42] VERBOSE[11989] chan_sip.c: Found RTP audio format 101 [Mar 10 18:40:42] VERBOSE[11989] chan_sip.c: Found audio description format PCMU for ID 0 [Mar 10 18:40:42] VERBOSE[11989] chan_sip.c: Found audio description format PCMA for ID 8 [Mar 10 18:40:42] VERBOSE[11989] chan_sip.c: Found audio description format G729 for ID 18 [Mar 10 18:40:42] VERBOSE[11989] chan_sip.c: Found audio description format telephone-event for ID 101 [Mar 10 18:40:42] VERBOSE[11989] chan_sip.c: Capabilities: us - 0x4 (ulaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [Mar 10 18:40:42] VERBOSE[11989] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Mar 10 18:40:42] VERBOSE[11989] chan_sip.c: Peer audio RTP is at port 172.17.15.156:17588 [Mar 10 18:40:42] VERBOSE[11989] chan_sip.c: <--- Transmitting (no NAT) to 172.17.15.156:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.17.15.156:5060;branch=z9hG4bK2329dcd5;received=172.17.15.156 From: ;tag=0007509810c807492a7f72c6-2ab209aa To: "Test" ;tag=as331324f7 Call-ID: 2e2f170d0ef48834251c4eee0b9610f2@172.17.15.245 CSeq: 101 INVITE Server: Asterisk PBX SVN-trunk-r251310 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 <------------> [Mar 10 18:40:42] VERBOSE[11989] chan_sip.c: Audio is at 172.17.15.245 port 17430 [Mar 10 18:40:42] VERBOSE[11989] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Mar 10 18:40:42] VERBOSE[11989] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Mar 10 18:40:42] VERBOSE[11989] chan_sip.c: <--- Reliably Transmitting (no NAT) to 172.17.15.156:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.17.15.156:5060;branch=z9hG4bK2329dcd5;received=172.17.15.156 From: ;tag=0007509810c807492a7f72c6-2ab209aa To: "Test" ;tag=as331324f7 Call-ID: 2e2f170d0ef48834251c4eee0b9610f2@172.17.15.245 CSeq: 101 INVITE Server: Asterisk PBX SVN-trunk-r251310 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 273 v=0 o=root 944702901 944702902 IN IP4 172.17.15.245 s=Asterisk PBX SVN-trunk-r251310 c=IN IP4 172.17.15.245 t=0 0 m=audio 17430 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=recvonly <------------> [Mar 10 18:40:42] VERBOSE[11989] chan_sip.c: <--- SIP read from UDP:172.17.15.156:50239 ---> ACK sip:38@172.17.15.245 SIP/2.0 Via: SIP/2.0/UDP 172.17.15.156:5060;branch=z9hG4bK42580f97 From: ;tag=0007509810c807492a7f72c6-2ab209aa To: "Test" ;tag=as331324f7 Call-ID: 2e2f170d0ef48834251c4eee0b9610f2@172.17.15.245 Max-Forwards: 70 Date: Wed, 10 Mar 2010 16:40:48 GMT CSeq: 101 ACK User-Agent: Cisco-CP7960G/8.0 Remote-Party-ID: "Ismailov Rafael" ;party=called;id-type=subscriber;privacy=off;screen=yes Content-Length: 0 <-------------> [Mar 10 18:40:42] VERBOSE[11989] chan_sip.c: --- (11 headers 0 lines) --- [Mar 10 18:40:44] VERBOSE[11989] chan_sip.c: <--- SIP read from UDP:172.17.15.156:52465 ---> INVITE sip:37@172.17.15.245 SIP/2.0 Via: SIP/2.0/UDP 172.17.15.156:5060;branch=z9hG4bK245b569d From: "Ismailov Rafael" ;tag=0007509810c8074a054c0ae5-3b5b42ec To: Call-ID: 00075098-10c8002c-08ff8f74-2de12d44@172.17.15.156 Max-Forwards: 70 Date: Wed, 10 Mar 2010 16:40:49 GMT CSeq: 101 INVITE User-Agent: Cisco-CP7960G/8.0 Contact: Expires: 180 Accept: application/sdp Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE Remote-Party-ID: "Ismailov Rafael" ;party=calling;id-type=subscriber;privacy=off;screen=yes Supported: replaces,join,norefersub Content-Length: 277 Content-Type: application/sdp Content-Disposition: session;handling=optional v=0 o=Cisco-SIPUA 3144 0 IN IP4 172.17.15.156 s=SIP Call t=0 0 m=audio 28866 RTP/AVP 0 8 18 101 c=IN IP4 172.17.15.156 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv <-------------> [Mar 10 18:40:44] VERBOSE[11989] chan_sip.c: --- (18 headers 13 lines) --- [Mar 10 18:40:44] VERBOSE[11989] chan_sip.c: Sending to 172.17.15.156 : 5060 (no NAT) [Mar 10 18:40:44] VERBOSE[11989] chan_sip.c: Using INVITE request as basis request - 00075098-10c8002c-08ff8f74-2de12d44@172.17.15.156 [Mar 10 18:40:44] VERBOSE[11989] chan_sip.c: Found peer '31' for '31' from 172.17.15.156:52465 [Mar 10 18:40:44] VERBOSE[11989] chan_sip.c: <--- Reliably Transmitting (no NAT) to 172.17.15.156:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 172.17.15.156:5060;branch=z9hG4bK245b569d;received=172.17.15.156 From: "Ismailov Rafael" ;tag=0007509810c8074a054c0ae5-3b5b42ec To: ;tag=as43c40d0b Call-ID: 00075098-10c8002c-08ff8f74-2de12d44@172.17.15.156 CSeq: 101 INVITE Server: Asterisk PBX SVN-trunk-r251310 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="17d9f8bf" Content-Length: 0 <------------> [Mar 10 18:40:44] VERBOSE[11989] chan_sip.c: Scheduling destruction of SIP dialog '00075098-10c8002c-08ff8f74-2de12d44@172.17.15.156' in 7232 ms (Method: INVITE) [Mar 10 18:40:44] VERBOSE[11989] chan_sip.c: <--- SIP read from UDP:172.17.15.156:52466 ---> ACK sip:37@172.17.15.245 SIP/2.0 Via: SIP/2.0/UDP 172.17.15.156:5060;branch=z9hG4bK245b569d From: "Ismailov Rafael" ;tag=0007509810c8074a054c0ae5-3b5b42ec To: ;tag=as43c40d0b Call-ID: 00075098-10c8002c-08ff8f74-2de12d44@172.17.15.156 Date: Wed, 10 Mar 2010 16:40:49 GMT CSeq: 101 ACK Content-Length: 0 <-------------> [Mar 10 18:40:44] VERBOSE[11989] chan_sip.c: --- (8 headers 0 lines) --- [Mar 10 18:40:44] VERBOSE[11989] chan_sip.c: <--- SIP read from UDP:172.17.15.156:52467 ---> INVITE sip:37@172.17.15.245 SIP/2.0 Via: SIP/2.0/UDP 172.17.15.156:5060;branch=z9hG4bK0ffb7587 From: "Ismailov Rafael" ;tag=0007509810c8074a054c0ae5-3b5b42ec To: Call-ID: 00075098-10c8002c-08ff8f74-2de12d44@172.17.15.156 Max-Forwards: 70 Date: Wed, 10 Mar 2010 16:40:49 GMT CSeq: 102 INVITE User-Agent: Cisco-CP7960G/8.0 Contact: Authorization: Digest username="31",realm="asterisk",uri="sip:37@172.17.15.245",response="074d57968d8620bcf577c25492561512",nonce="17d9f8bf",algorithm=MD5 Expires: 180 Accept: application/sdp Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE Remote-Party-ID: "Ismailov Rafael" ;party=calling;id-type=subscriber;privacy=off;screen=yes Supported: replaces,join,norefersub Content-Length: 277 Content-Type: application/sdp Content-Disposition: session;handling=optional v=0 o=Cisco-SIPUA 3144 0 IN IP4 172.17.15.156 s=SIP Call t=0 0 m=audio 28866 RTP/AVP 0 8 18 101 c=IN IP4 172.17.15.156 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv <-------------> [Mar 10 18:40:44] VERBOSE[11989] chan_sip.c: --- (19 headers 13 lines) --- [Mar 10 18:40:44] VERBOSE[11989] chan_sip.c: Sending to 172.17.15.156 : 5060 (no NAT) [Mar 10 18:40:44] VERBOSE[11989] chan_sip.c: Using INVITE request as basis request - 00075098-10c8002c-08ff8f74-2de12d44@172.17.15.156 [Mar 10 18:40:44] VERBOSE[11989] chan_sip.c: Found peer '31' for '31' from 172.17.15.156:52467 [Mar 10 18:40:44] VERBOSE[11989] chan_sip.c: Found RTP audio format 0 [Mar 10 18:40:44] VERBOSE[11989] chan_sip.c: Found RTP audio format 8 [Mar 10 18:40:44] VERBOSE[11989] chan_sip.c: Found RTP audio format 18 [Mar 10 18:40:44] VERBOSE[11989] chan_sip.c: Found RTP audio format 101 [Mar 10 18:40:44] VERBOSE[11989] chan_sip.c: Found audio description format PCMU for ID 0 [Mar 10 18:40:44] VERBOSE[11989] chan_sip.c: Found audio description format PCMA for ID 8 [Mar 10 18:40:44] VERBOSE[11989] chan_sip.c: Found audio description format G729 for ID 18 [Mar 10 18:40:44] VERBOSE[11989] chan_sip.c: Found audio description format telephone-event for ID 101 [Mar 10 18:40:44] VERBOSE[11989] chan_sip.c: Capabilities: us - 0x4 (ulaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [Mar 10 18:40:44] VERBOSE[11989] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Mar 10 18:40:44] VERBOSE[11989] chan_sip.c: Peer audio RTP is at port 172.17.15.156:28866 [Mar 10 18:40:44] VERBOSE[11989] chan_sip.c: Looking for 37 in DLPN_To_Town (domain 172.17.15.245) [Mar 10 18:40:44] VERBOSE[11989] chan_sip.c: list_route: hop: [Mar 10 18:40:44] VERBOSE[11989] chan_sip.c: <--- Transmitting (no NAT) to 172.17.15.156:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.17.15.156:5060;branch=z9hG4bK0ffb7587;received=172.17.15.156 From: "Ismailov Rafael" ;tag=0007509810c8074a054c0ae5-3b5b42ec To: Call-ID: 00075098-10c8002c-08ff8f74-2de12d44@172.17.15.156 CSeq: 102 INVITE Server: Asterisk PBX SVN-trunk-r251310 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 <------------> [Mar 10 18:40:44] VERBOSE[11995] chan_sip.c: Audio is at 172.17.15.245 port 17262 [Mar 10 18:40:44] VERBOSE[11995] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Mar 10 18:40:44] VERBOSE[11995] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Mar 10 18:40:44] VERBOSE[11995] chan_sip.c: <--- Reliably Transmitting (no NAT) to 172.17.15.156:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.17.15.156:5060;branch=z9hG4bK0ffb7587;received=172.17.15.156 From: "Ismailov Rafael" ;tag=0007509810c8074a054c0ae5-3b5b42ec To: ;tag=as4f55130d Call-ID: 00075098-10c8002c-08ff8f74-2de12d44@172.17.15.156 CSeq: 102 INVITE Server: Asterisk PBX SVN-trunk-r251310 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 275 v=0 o=root 1390242811 1390242811 IN IP4 172.17.15.245 s=Asterisk PBX SVN-trunk-r251310 c=IN IP4 172.17.15.245 t=0 0 m=audio 17262 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Mar 10 18:40:44] VERBOSE[11995] chan_sip.c: Audio is at 172.17.15.245 port 12102 [Mar 10 18:40:44] VERBOSE[11995] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Mar 10 18:40:44] VERBOSE[11995] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Mar 10 18:40:44] VERBOSE[11995] chan_sip.c: Reliably Transmitting (no NAT) to 172.17.17.72:5060: INVITE sip:37@172.17.17.72:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 172.17.15.245:5060;branch=z9hG4bK08a08053 Max-Forwards: 70 From: "Ismailov Rafael" ;tag=as79a80205 To: Contact: Call-ID: 6844c204626ee5714c77c2433a82d5da@172.17.15.245 CSeq: 102 INVITE User-Agent: Asterisk PBX SVN-trunk-r251310 Date: Wed, 10 Mar 2010 16:40:44 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Remote-Party-ID: "Ismailov Rafael" ;party=calling;privacy=off;screen=yes Content-Type: application/sdp Content-Length: 273 v=0 o=root 296178105 296178105 IN IP4 172.17.15.245 s=Asterisk PBX SVN-trunk-r251310 c=IN IP4 172.17.15.245 t=0 0 m=audio 12102 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Mar 10 18:40:44] VERBOSE[11989] chan_sip.c: Retransmitting #1 (no NAT) to 172.17.15.156:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 172.17.15.156:5060;branch=z9hG4bK0ffb7587;received=172.17.15.156 From: "Ismailov Rafael" ;tag=0007509810c8074a054c0ae5-3b5b42ec To: ;tag=as4f55130d Call-ID: 00075098-10c8002c-08ff8f74-2de12d44@172.17.15.156 CSeq: 102 INVITE Server: Asterisk PBX SVN-trunk-r251310 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 275 v=0 o=root 1390242811 1390242811 IN IP4 172.17.15.245 s=Asterisk PBX SVN-trunk-r251310 c=IN IP4 172.17.15.245 t=0 0 m=audio 17262 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Mar 10 18:40:44] VERBOSE[11989] chan_sip.c: <--- SIP read from UDP:172.17.17.72:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.17.15.245:5060;branch=z9hG4bK08a08053 From: "Ismailov Rafael" ;tag=as79a80205 To: Call-ID: 6844c204626ee5714c77c2433a82d5da@172.17.15.245 CSeq: 102 INVITE User-Agent: Grandstream GXP1200 1.1.6.16 Content-Length: 0 <-------------> [Mar 10 18:40:44] VERBOSE[11989] chan_sip.c: --- (8 headers 0 lines) --- [Mar 10 18:40:44] VERBOSE[11989] chan_sip.c: <--- SIP read from UDP:172.17.17.72:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 172.17.15.245:5060;branch=z9hG4bK08a08053 From: "Ismailov Rafael" ;tag=as79a80205 To: ;tag=fea228a24b360836 Call-ID: 6844c204626ee5714c77c2433a82d5da@172.17.15.245 CSeq: 102 INVITE User-Agent: Grandstream GXP1200 1.1.6.16 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 <-------------> [Mar 10 18:40:44] VERBOSE[11989] chan_sip.c: --- (10 headers 0 lines) --- [Mar 10 18:40:44] VERBOSE[11989] chan_sip.c: <--- SIP read from UDP:172.17.15.156:52468 ---> ACK sip:37@172.17.15.245 SIP/2.0 Via: SIP/2.0/UDP 172.17.15.156:5060;branch=z9hG4bK153dcc3c From: "Ismailov Rafael" ;tag=0007509810c8074a054c0ae5-3b5b42ec To: ;tag=as4f55130d Call-ID: 00075098-10c8002c-08ff8f74-2de12d44@172.17.15.156 Max-Forwards: 70 Date: Wed, 10 Mar 2010 16:40:50 GMT CSeq: 102 ACK User-Agent: Cisco-CP7960G/8.0 Authorization: Digest username="31",realm="asterisk",uri="sip:37@172.17.15.245",response="074d57968d8620bcf577c25492561512",nonce="17d9f8bf",algorithm=MD5 Remote-Party-ID: "Ismailov Rafael" ;party=calling;id-type=subscriber;privacy=off;screen=yes Content-Length: 0 <-------------> [Mar 10 18:40:44] VERBOSE[11989] chan_sip.c: --- (12 headers 0 lines) --- [Mar 10 18:40:45] VERBOSE[11989] chan_sip.c: Reliably Transmitting (no NAT) to 172.17.17.72:5060: OPTIONS sip:37@172.17.17.72:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 172.17.15.245:5060;branch=z9hG4bK5cf03cf7 Max-Forwards: 70 From: "asterisk" ;tag=as63a86b23 To: Contact: Call-ID: 0c2bed8103f5dce617148ed9734190cc@172.17.15.245 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-trunk-r251310 Date: Wed, 10 Mar 2010 16:40:45 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- [Mar 10 18:40:45] VERBOSE[11989] chan_sip.c: <--- SIP read from UDP:172.17.17.72:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.17.15.245:5060;branch=z9hG4bK5cf03cf7 From: "asterisk" ;tag=as63a86b23 To: ;tag=fea228a24b360836 Call-ID: 0c2bed8103f5dce617148ed9734190cc@172.17.15.245 CSeq: 102 OPTIONS User-Agent: Grandstream GXP1200 1.1.6.16 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Supported: replaces, timer Content-Length: 0 <-------------> [Mar 10 18:40:45] VERBOSE[11989] chan_sip.c: --- (11 headers 0 lines) --- [Mar 10 18:40:45] VERBOSE[11989] chan_sip.c: Really destroying SIP dialog '0c2bed8103f5dce617148ed9734190cc@172.17.15.245' Method: OPTIONS [Mar 10 18:40:45] VERBOSE[11989] chan_sip.c: Reliably Transmitting (no NAT) to 172.17.15.166:5062: OPTIONS sip:38@172.17.15.166:5062;transport=udp SIP/2.0 Via: SIP/2.0/UDP 172.17.15.245:5060;branch=z9hG4bK5703b17a Max-Forwards: 70 From: "asterisk" ;tag=as004ec02c To: Contact: Call-ID: 0918d81469249da3345669bc2f594f3e@172.17.15.245 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-trunk-r251310 Date: Wed, 10 Mar 2010 16:40:45 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- [Mar 10 18:40:45] VERBOSE[11989] chan_sip.c: <--- SIP read from UDP:172.17.15.166:5062 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.17.15.245:5060;branch=z9hG4bK5703b17a From: "asterisk" ;tag=as004ec02c To: ;tag=as00a805ed Call-ID: 0918d81469249da3345669bc2f594f3e@172.17.15.245 CSeq: 102 OPTIONS User-Agent: Grandstream GXP1200 1.2.1.4 Session-Expires: 180;refresher=uas Min-SE: 180 Require: timer Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Supported: replaces, timer Content-Length: 0 <-------------> [Mar 10 18:40:45] VERBOSE[11989] chan_sip.c: --- (14 headers 0 lines) --- [Mar 10 18:40:45] VERBOSE[11989] chan_sip.c: Really destroying SIP dialog '0918d81469249da3345669bc2f594f3e@172.17.15.245' Method: OPTIONS [Mar 10 18:40:45] VERBOSE[11989] chan_sip.c: Reliably Transmitting (no NAT) to 172.17.15.156:5060: OPTIONS sip:31@172.17.15.156:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 172.17.15.245:5060;branch=z9hG4bK25b603de Max-Forwards: 70 From: "asterisk" ;tag=as30dc775e To: Contact: Call-ID: 336da81f26f5dac42543c82f6526c4db@172.17.15.245 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-trunk-r251310 Date: Wed, 10 Mar 2010 16:40:45 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- [Mar 10 18:40:45] VERBOSE[11989] chan_sip.c: Reliably Transmitting (no NAT) to 172.17.15.157:5060: OPTIONS sip:39@172.17.15.157:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 172.17.15.245:5060;branch=z9hG4bK713bb898 Max-Forwards: 70 From: "asterisk" ;tag=as074b19ab To: Contact: Call-ID: 0b9858056dc2d71c61a89e7d5d3e2bf2@172.17.15.245 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-trunk-r251310 Date: Wed, 10 Mar 2010 16:40:45 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- [Mar 10 18:40:45] VERBOSE[11989] chan_sip.c: <--- SIP read from UDP:172.17.15.156:52469 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.17.15.245:5060;branch=z9hG4bK25b603de From: "asterisk" ;tag=as30dc775e To: ;tag=0007509810c8074b44630541-0be4682f Call-ID: 336da81f26f5dac42543c82f6526c4db@172.17.15.245 Date: Wed, 10 Mar 2010 16:40:51 GMT CSeq: 102 OPTIONS Server: Cisco-CP7960G/8.0 Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE Accept: application/sdp,multipart/mixed,multipart/alternative Accept-Encoding: identity Accept-Language: en Supported: replaces,join,norefersub Content-Length: 237 Content-Type: application/sdp Content-Disposition: session;handling=optional v=0 o=Cisco-SIPUA 6073 0 IN IP4 172.17.15.156 s=SIP Call t=0 0 m=audio 0 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> [Mar 10 18:40:45] VERBOSE[11989] chan_sip.c: --- (16 headers 11 lines) --- [Mar 10 18:40:45] VERBOSE[11989] chan_sip.c: Really destroying SIP dialog '336da81f26f5dac42543c82f6526c4db@172.17.15.245' Method: OPTIONS [Mar 10 18:40:45] VERBOSE[11989] chan_sip.c: <--- SIP read from UDP:172.17.15.157:52463 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.17.15.245:5060;branch=z9hG4bK713bb898 From: "asterisk" ;tag=as074b19ab To: ;tag=0007855c8ba5066f1a591d82-14481999 Call-ID: 0b9858056dc2d71c61a89e7d5d3e2bf2@172.17.15.245 Date: Wed, 10 Mar 2010 16:40:49 GMT CSeq: 102 OPTIONS Server: Cisco-CP7940G/8.0 Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE Accept: application/sdp,multipart/mixed,multipart/alternative Accept-Encoding: identity Accept-Language: en Supported: replaces,join,norefersub Content-Length: 238 Content-Type: application/sdp Content-Disposition: session;handling=optional v=0 o=Cisco-SIPUA 17823 0 IN IP4 172.17.15.157 s=SIP Call t=0 0 m=audio 0 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> [Mar 10 18:40:45] VERBOSE[11989] chan_sip.c: --- (16 headers 11 lines) --- [Mar 10 18:40:45] VERBOSE[11989] chan_sip.c: Really destroying SIP dialog '0b9858056dc2d71c61a89e7d5d3e2bf2@172.17.15.245' Method: OPTIONS [Mar 10 18:40:47] VERBOSE[11989] chan_sip.c: <--- SIP read from UDP:172.17.17.72:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.17.15.245:5060;branch=z9hG4bK08a08053 From: "Ismailov Rafael" ;tag=as79a80205 To: ;tag=fea228a24b360836 Call-ID: 6844c204626ee5714c77c2433a82d5da@172.17.15.245 CSeq: 102 INVITE User-Agent: Grandstream GXP1200 1.1.6.16 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Supported: replaces, timer Content-Length: 209 v=0 o=37 8000 8000 IN IP4 172.17.17.72 s=SIP Call c=IN IP4 172.17.17.72 t=0 0 m=audio 5004 RTP/AVP 0 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 <-------------> [Mar 10 18:40:47] VERBOSE[11989] chan_sip.c: --- (12 headers 11 lines) --- [Mar 10 18:40:47] VERBOSE[11989] chan_sip.c: Found RTP audio format 0 [Mar 10 18:40:47] VERBOSE[11989] chan_sip.c: Found RTP audio format 101 [Mar 10 18:40:47] VERBOSE[11989] chan_sip.c: Found audio description format PCMU for ID 0 [Mar 10 18:40:47] VERBOSE[11989] chan_sip.c: Found audio description format telephone-event for ID 101 [Mar 10 18:40:47] VERBOSE[11989] chan_sip.c: Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [Mar 10 18:40:47] VERBOSE[11989] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Mar 10 18:40:47] VERBOSE[11989] chan_sip.c: Peer audio RTP is at port 172.17.17.72:5004 [Mar 10 18:40:47] VERBOSE[11989] chan_sip.c: list_route: hop: [Mar 10 18:40:47] VERBOSE[11989] chan_sip.c: set_destination: Parsing for address/port to send to [Mar 10 18:40:47] VERBOSE[11989] chan_sip.c: set_destination: set destination to 172.17.17.72, port 5060 [Mar 10 18:40:47] VERBOSE[11989] chan_sip.c: Transmitting (no NAT) to 172.17.17.72:5060: ACK sip:37@172.17.17.72:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 172.17.15.245:5060;branch=z9hG4bK21e200d6 Max-Forwards: 70 From: "Ismailov Rafael" ;tag=as79a80205 To: ;tag=fea228a24b360836 Contact: Call-ID: 6844c204626ee5714c77c2433a82d5da@172.17.15.245 CSeq: 102 ACK User-Agent: Asterisk PBX SVN-trunk-r251310 Content-Length: 0 --- [Mar 10 18:40:47] VERBOSE[11995] chan_sip.c: set_destination: Parsing for address/port to send to [Mar 10 18:40:47] VERBOSE[11995] chan_sip.c: set_destination: set destination to 172.17.15.156, port 5060 [Mar 10 18:40:47] VERBOSE[11995] chan_sip.c: Audio is at 172.17.15.245 port 17262 [Mar 10 18:40:47] VERBOSE[11995] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Mar 10 18:40:47] VERBOSE[11995] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Mar 10 18:40:47] VERBOSE[11995] chan_sip.c: Reliably Transmitting (no NAT) to 172.17.15.156:5060: INVITE sip:31@172.17.15.156:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 172.17.15.245:5060;branch=z9hG4bK0ccdb8c2 Max-Forwards: 70 From: ;tag=as4f55130d To: "Ismailov Rafael" ;tag=0007509810c8074a054c0ae5-3b5b42ec Contact: Call-ID: 00075098-10c8002c-08ff8f74-2de12d44@172.17.15.156 CSeq: 102 INVITE User-Agent: Asterisk PBX SVN-trunk-r251310 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Remote-Party-ID: "Test1" ;party=called;privacy=off;screen=no Content-Type: application/sdp Content-Length: 275 v=0 o=root 1390242811 1390242812 IN IP4 172.17.15.245 s=Asterisk PBX SVN-trunk-r251310 c=IN IP4 172.17.15.245 t=0 0 m=audio 17262 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Mar 10 18:40:48] VERBOSE[11989] chan_sip.c: <--- SIP read from UDP:172.17.15.156:52470 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.17.15.245:5060;branch=z9hG4bK0ccdb8c2 From: ;tag=as4f55130d To: "Ismailov Rafael" ;tag=0007509810c8074a054c0ae5-3b5b42ec Call-ID: 00075098-10c8002c-08ff8f74-2de12d44@172.17.15.156 Date: Wed, 10 Mar 2010 16:40:53 GMT CSeq: 102 INVITE Server: Cisco-CP7960G/8.0 Contact: Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE Remote-Party-ID: "Ismailov Rafael" ;party=calling;id-type=subscriber;privacy=off;screen=yes Supported: replaces,join,norefersub Content-Length: 206 Content-Type: application/sdp Content-Disposition: session;handling=optional v=0 o=Cisco-SIPUA 3144 1 IN IP4 172.17.15.156 s=SIP Call t=0 0 m=audio 28866 RTP/AVP 0 101 c=IN IP4 172.17.15.156 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv <-------------> [Mar 10 18:40:48] VERBOSE[11989] chan_sip.c: --- (15 headers 10 lines) --- [Mar 10 18:40:48] VERBOSE[11989] chan_sip.c: Found RTP audio format 0 [Mar 10 18:40:48] VERBOSE[11989] chan_sip.c: Found RTP audio format 101 [Mar 10 18:40:48] VERBOSE[11989] chan_sip.c: Found audio description format PCMU for ID 0 [Mar 10 18:40:48] VERBOSE[11989] chan_sip.c: Found audio description format telephone-event for ID 101 [Mar 10 18:40:48] VERBOSE[11989] chan_sip.c: Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [Mar 10 18:40:48] VERBOSE[11989] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Mar 10 18:40:48] VERBOSE[11989] chan_sip.c: Peer audio RTP is at port 172.17.15.156:28866 [Mar 10 18:40:48] VERBOSE[11989] chan_sip.c: set_destination: Parsing for address/port to send to [Mar 10 18:40:48] VERBOSE[11989] chan_sip.c: set_destination: set destination to 172.17.15.156, port 5060 [Mar 10 18:40:48] VERBOSE[11989] chan_sip.c: Transmitting (no NAT) to 172.17.15.156:5060: ACK sip:31@172.17.15.156:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 172.17.15.245:5060;branch=z9hG4bK18b9fad2 Max-Forwards: 70 From: ;tag=as4f55130d To: "Ismailov Rafael" ;tag=0007509810c8074a054c0ae5-3b5b42ec Contact: Call-ID: 00075098-10c8002c-08ff8f74-2de12d44@172.17.15.156 CSeq: 102 ACK User-Agent: Asterisk PBX SVN-trunk-r251310 Content-Length: 0 --- [Mar 10 18:40:52] VERBOSE[11989] chan_sip.c: <--- SIP read from UDP:172.17.15.156:50239 ---> INVITE sip:37@172.17.15.245 SIP/2.0 Via: SIP/2.0/UDP 172.17.15.156:5060;branch=z9hG4bK390f581d From: "Ismailov Rafael" ;tag=0007509810c8074a054c0ae5-3b5b42ec To: ;tag=as4f55130d Call-ID: 00075098-10c8002c-08ff8f74-2de12d44@172.17.15.156 Max-Forwards: 70 Date: Wed, 10 Mar 2010 16:40:58 GMT CSeq: 103 INVITE User-Agent: Cisco-CP7960G/8.0 Contact: Accept: application/sdp Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE Remote-Party-ID: "Ismailov Rafael" ;party=calling;id-type=subscriber;privacy=off;screen=yes Supported: replaces,join,norefersub Authorization: Digest username="31",realm="asterisk",uri="sip:37@172.17.15.245",response="074d57968d8620bcf577c25492561512",nonce="17d9f8bf",algorithm=MD5 Content-Length: 277 Content-Type: application/sdp Content-Disposition: session;handling=optional v=0 o=Cisco-SIPUA 3144 2 IN IP4 172.17.15.156 s=SIP Call t=0 0 m=audio 28866 RTP/AVP 0 8 18 101 c=IN IP4 172.17.15.156 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendonly <-------------> [Mar 10 18:40:52] VERBOSE[11989] chan_sip.c: --- (18 headers 13 lines) --- [Mar 10 18:40:52] VERBOSE[11989] chan_sip.c: Sending to 172.17.15.156 : 5060 (no NAT) [Mar 10 18:40:52] VERBOSE[11989] chan_sip.c: Found RTP audio format 0 [Mar 10 18:40:52] VERBOSE[11989] chan_sip.c: Found RTP audio format 8 [Mar 10 18:40:52] VERBOSE[11989] chan_sip.c: Found RTP audio format 18 [Mar 10 18:40:52] VERBOSE[11989] chan_sip.c: Found RTP audio format 101 [Mar 10 18:40:52] VERBOSE[11989] chan_sip.c: Found audio description format PCMU for ID 0 [Mar 10 18:40:52] VERBOSE[11989] chan_sip.c: Found audio description format PCMA for ID 8 [Mar 10 18:40:52] VERBOSE[11989] chan_sip.c: Found audio description format G729 for ID 18 [Mar 10 18:40:52] VERBOSE[11989] chan_sip.c: Found audio description format telephone-event for ID 101 [Mar 10 18:40:52] VERBOSE[11989] chan_sip.c: Capabilities: us - 0x4 (ulaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [Mar 10 18:40:52] VERBOSE[11989] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Mar 10 18:40:52] VERBOSE[11989] chan_sip.c: Peer audio RTP is at port 172.17.15.156:28866 [Mar 10 18:40:52] VERBOSE[11989] chan_sip.c: <--- Transmitting (no NAT) to 172.17.15.156:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.17.15.156:5060;branch=z9hG4bK390f581d;received=172.17.15.156 From: "Ismailov Rafael" ;tag=0007509810c8074a054c0ae5-3b5b42ec To: ;tag=as4f55130d Call-ID: 00075098-10c8002c-08ff8f74-2de12d44@172.17.15.156 CSeq: 103 INVITE Server: Asterisk PBX SVN-trunk-r251310 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 <------------> [Mar 10 18:40:52] VERBOSE[11989] chan_sip.c: Audio is at 172.17.15.245 port 17262 [Mar 10 18:40:52] VERBOSE[11989] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Mar 10 18:40:52] VERBOSE[11989] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Mar 10 18:40:52] VERBOSE[11989] chan_sip.c: <--- Reliably Transmitting (no NAT) to 172.17.15.156:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.17.15.156:5060;branch=z9hG4bK390f581d;received=172.17.15.156 From: "Ismailov Rafael" ;tag=0007509810c8074a054c0ae5-3b5b42ec To: ;tag=as4f55130d Call-ID: 00075098-10c8002c-08ff8f74-2de12d44@172.17.15.156 CSeq: 103 INVITE Server: Asterisk PBX SVN-trunk-r251310 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 275 v=0 o=root 1390242811 1390242813 IN IP4 172.17.15.245 s=Asterisk PBX SVN-trunk-r251310 c=IN IP4 172.17.15.245 t=0 0 m=audio 17262 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=recvonly <------------> [Mar 10 18:40:52] VERBOSE[11989] chan_sip.c: <--- SIP read from UDP:172.17.15.156:50239 ---> ACK sip:37@172.17.15.245 SIP/2.0 Via: SIP/2.0/UDP 172.17.15.156:5060;branch=z9hG4bK09ed60a8 From: "Ismailov Rafael" ;tag=0007509810c8074a054c0ae5-3b5b42ec To: ;tag=as4f55130d Call-ID: 00075098-10c8002c-08ff8f74-2de12d44@172.17.15.156 Max-Forwards: 70 Date: Wed, 10 Mar 2010 16:40:58 GMT CSeq: 103 ACK User-Agent: Cisco-CP7960G/8.0 Authorization: Digest username="31",realm="asterisk",uri="sip:37@172.17.15.245",response="074d57968d8620bcf577c25492561512",nonce="17d9f8bf",algorithm=MD5 Remote-Party-ID: "Ismailov Rafael" ;party=calling;id-type=subscriber;privacy=off;screen=yes Content-Length: 0 <-------------> [Mar 10 18:40:52] VERBOSE[11989] chan_sip.c: --- (12 headers 0 lines) --- [Mar 10 18:40:52] VERBOSE[11989] chan_sip.c: <--- SIP read from UDP:172.17.15.156:50239 ---> REFER sip:38@172.17.15.245 SIP/2.0 Via: SIP/2.0/UDP 172.17.15.156:5060;branch=z9hG4bK5d46ef8f From: ;tag=0007509810c807492a7f72c6-2ab209aa To: "Test" ;tag=as331324f7 Call-ID: 2e2f170d0ef48834251c4eee0b9610f2@172.17.15.245 Max-Forwards: 70 Date: Wed, 10 Mar 2010 16:40:58 GMT CSeq: 102 REFER User-Agent: Cisco-CP7960G/8.0 Contact: Remote-Party-ID: "Ismailov Rafael" ;party=called;id-type=subscriber;privacy=off;screen=yes Refer-To: Referred-By: Content-Length: 0 <-------------> [Mar 10 18:40:52] VERBOSE[11989] chan_sip.c: --- (14 headers 0 lines) --- [Mar 10 18:40:52] VERBOSE[11989] chan_sip.c: Call 2e2f170d0ef48834251c4eee0b9610f2@172.17.15.245 got a SIP call transfer from caller: (REFER)! [Mar 10 18:40:52] VERBOSE[11989] chan_sip.c: SIP transfer to extension 37@DLPN_To_Town by 31@172.17.15.245 [Mar 10 18:40:52] VERBOSE[11989] chan_sip.c: <--- Transmitting (no NAT) to 172.17.15.156:5060 ---> SIP/2.0 202 Accepted Via: SIP/2.0/UDP 172.17.15.156:5060;branch=z9hG4bK5d46ef8f;received=172.17.15.156 From: ;tag=0007509810c807492a7f72c6-2ab209aa To: "Test" ;tag=as331324f7 Call-ID: 2e2f170d0ef48834251c4eee0b9610f2@172.17.15.245 CSeq: 102 REFER Server: Asterisk PBX SVN-trunk-r251310 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 <------------> [Mar 10 18:40:52] VERBOSE[11989] chan_sip.c: set_destination: Parsing for address/port to send to [Mar 10 18:40:52] VERBOSE[11989] chan_sip.c: set_destination: set destination to 172.17.15.156, port 5060 [Mar 10 18:40:52] VERBOSE[11989] chan_sip.c: Reliably Transmitting (no NAT) to 172.17.15.156:5060: NOTIFY sip:31@172.17.15.156:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 172.17.15.245:5060;branch=z9hG4bK2960b4ec Max-Forwards: 70 From: "Test" ;tag=as331324f7 To: ;tag=0007509810c807492a7f72c6-2ab209aa Contact: Call-ID: 2e2f170d0ef48834251c4eee0b9610f2@172.17.15.245 CSeq: 103 NOTIFY User-Agent: Asterisk PBX SVN-trunk-r251310 Event: refer;id=102 Subscription-state: terminated;reason=noresource Content-Type: message/sipfrag;version=2.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 49 SIP/2.0 481 Call leg/transaction does not exist --- [Mar 10 18:40:52] VERBOSE[11989] chan_sip.c: <--- SIP read from UDP:172.17.15.156:52471 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.17.15.245:5060;branch=z9hG4bK2960b4ec From: "Test" ;tag=as331324f7 To: ;tag=0007509810c807492a7f72c6-2ab209aa Call-ID: 2e2f170d0ef48834251c4eee0b9610f2@172.17.15.245 Date: Wed, 10 Mar 2010 16:40:58 GMT CSeq: 103 NOTIFY Content-Length: 0 <-------------> [Mar 10 18:40:52] VERBOSE[11989] chan_sip.c: --- (8 headers 0 lines) --- [Mar 10 18:40:52] VERBOSE[11989] chan_sip.c: SIP Response message for INCOMING dialog NOTIFY arrived [Mar 10 18:40:57] VERBOSE[11989] chan_sip.c: <--- SIP read from UDP:172.17.17.72:5060 ---> BYE sip:31@172.17.15.245 SIP/2.0 Via: SIP/2.0/UDP 172.17.17.72:5060;branch=z9hG4bK7fc97b45596ec78e From: ;tag=fea228a24b360836 To: "Ismailov Rafael" ;tag=as79a80205 Supported: path Call-ID: 6844c204626ee5714c77c2433a82d5da@172.17.15.245 CSeq: 64632 BYE User-Agent: Grandstream GXP1200 1.1.6.16 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 <-------------> [Mar 10 18:40:57] VERBOSE[11989] chan_sip.c: --- (11 headers 0 lines) --- [Mar 10 18:40:57] VERBOSE[11989] chan_sip.c: Sending to 172.17.17.72 : 5060 (no NAT) [Mar 10 18:40:57] VERBOSE[11989] chan_sip.c: <--- Transmitting (no NAT) to 172.17.17.72:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.17.17.72:5060;branch=z9hG4bK7fc97b45596ec78e;received=172.17.17.72 From: ;tag=fea228a24b360836 To: "Ismailov Rafael" ;tag=as79a80205 Call-ID: 6844c204626ee5714c77c2433a82d5da@172.17.15.245 CSeq: 64632 BYE Server: Asterisk PBX SVN-trunk-r251310 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> [Mar 10 18:40:57] VERBOSE[11995] chan_sip.c: Scheduling destruction of SIP dialog '00075098-10c8002c-08ff8f74-2de12d44@172.17.15.156' in 7232 ms (Method: ACK) [Mar 10 18:40:57] VERBOSE[11995] chan_sip.c: set_destination: Parsing for address/port to send to [Mar 10 18:40:57] VERBOSE[11995] chan_sip.c: set_destination: set destination to 172.17.15.156, port 5060 [Mar 10 18:40:57] VERBOSE[11995] chan_sip.c: Reliably Transmitting (no NAT) to 172.17.15.156:5060: BYE sip:31@172.17.15.156:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 172.17.15.245:5060;branch=z9hG4bK402b6d38 Max-Forwards: 70 From: ;tag=as4f55130d To: "Ismailov Rafael" ;tag=0007509810c8074a054c0ae5-3b5b42ec Call-ID: 00075098-10c8002c-08ff8f74-2de12d44@172.17.15.156 CSeq: 103 BYE User-Agent: Asterisk PBX SVN-trunk-r251310 Proxy-Authorization: Digest username="31", realm="asterisk", algorithm=MD5, uri="172.17.15.245", nonce="", response="523688cc209a8e88d22eef4d8f38290c" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Mar 10 18:40:57] VERBOSE[11989] chan_sip.c: <--- SIP read from UDP:172.17.15.156:50239 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.17.15.245:5060;branch=z9hG4bK402b6d38 From: ;tag=as4f55130d To: "Ismailov Rafael" ;tag=0007509810c8074a054c0ae5-3b5b42ec Call-ID: 00075098-10c8002c-08ff8f74-2de12d44@172.17.15.156 Date: Wed, 10 Mar 2010 16:41:03 GMT CSeq: 103 BYE Server: Cisco-CP7960G/8.0 Content-Length: 0 <-------------> [Mar 10 18:40:57] VERBOSE[11989] chan_sip.c: --- (9 headers 0 lines) --- [Mar 10 18:40:57] VERBOSE[11989] chan_sip.c: SIP Response message for INCOMING dialog BYE arrived [Mar 10 18:40:57] VERBOSE[11989] chan_sip.c: Really destroying SIP dialog '6844c204626ee5714c77c2433a82d5da@172.17.15.245' Method: BYE [Mar 10 18:40:57] VERBOSE[11989] chan_sip.c: Really destroying SIP dialog '00075098-10c8002c-08ff8f74-2de12d44@172.17.15.156' Method: ACK [Mar 10 18:40:59] VERBOSE[11989] chan_sip.c: <--- SIP read from UDP:172.17.15.166:5062 ---> BYE sip:31@172.17.15.245 SIP/2.0 Via: SIP/2.0/UDP 172.17.15.166:5062;branch=z9hG4bKa16e983c70f28fc8 From: "Ismailov Rafael" ;tag=04720d5e2e970e75 To: ;tag=as00a805ed Supported: path Call-ID: 0cac0c52f9273627@172.17.15.166 CSeq: 25529 BYE User-Agent: Grandstream GXP1200 1.2.1.4 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 <-------------> [Mar 10 18:40:59] VERBOSE[11989] chan_sip.c: --- (11 headers 0 lines) --- [Mar 10 18:40:59] VERBOSE[11989] chan_sip.c: Sending to 172.17.15.166 : 5062 (no NAT) [Mar 10 18:40:59] VERBOSE[11989] chan_sip.c: <--- Transmitting (no NAT) to 172.17.15.166:5062 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.17.15.166:5062;branch=z9hG4bKa16e983c70f28fc8;received=172.17.15.166 From: "Ismailov Rafael" ;tag=04720d5e2e970e75 To: ;tag=as00a805ed Call-ID: 0cac0c52f9273627@172.17.15.166 CSeq: 25529 BYE Server: Asterisk PBX SVN-trunk-r251310 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> [Mar 10 18:40:59] VERBOSE[11994] chan_sip.c: Scheduling destruction of SIP dialog '2e2f170d0ef48834251c4eee0b9610f2@172.17.15.245' in 7232 ms (Method: REFER) [Mar 10 18:40:59] VERBOSE[11994] chan_sip.c: set_destination: Parsing for address/port to send to [Mar 10 18:40:59] VERBOSE[11994] chan_sip.c: set_destination: set destination to 172.17.15.156, port 5060 [Mar 10 18:40:59] VERBOSE[11994] chan_sip.c: Reliably Transmitting (no NAT) to 172.17.15.156:5060: BYE sip:31@172.17.15.156:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 172.17.15.245:5060;branch=z9hG4bK1a7cc135 Max-Forwards: 70 From: "Test" ;tag=as331324f7 To: ;tag=0007509810c807492a7f72c6-2ab209aa Call-ID: 2e2f170d0ef48834251c4eee0b9610f2@172.17.15.245 CSeq: 104 BYE User-Agent: Asterisk PBX SVN-trunk-r251310 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Mar 10 18:40:59] VERBOSE[11989] chan_sip.c: <--- SIP read from UDP:172.17.15.156:50239 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.17.15.245:5060;branch=z9hG4bK1a7cc135 From: "Test" ;tag=as331324f7 To: ;tag=0007509810c807492a7f72c6-2ab209aa Call-ID: 2e2f170d0ef48834251c4eee0b9610f2@172.17.15.245 Date: Wed, 10 Mar 2010 16:41:04 GMT CSeq: 104 BYE Server: Cisco-CP7960G/8.0 Content-Length: 0 <-------------> [Mar 10 18:40:59] VERBOSE[11989] chan_sip.c: --- (9 headers 0 lines) --- [Mar 10 18:40:59] VERBOSE[11989] chan_sip.c: SIP Response message for INCOMING dialog BYE arrived [Mar 10 18:40:59] VERBOSE[11989] chan_sip.c: Really destroying SIP dialog '0cac0c52f9273627@172.17.15.166' Method: BYE [Mar 10 18:40:59] VERBOSE[11989] chan_sip.c: Really destroying SIP dialog '2e2f170d0ef48834251c4eee0b9610f2@172.17.15.245' Method: REFER