part #2
<--- SIP read from UDP:10.10.1.144:5060 ---> INVITE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 10.10.1.144:5060;branch=z9hG4bK.UwYy83FQ9;rport From: <sip:[email protected]>;tag=XY9jSoWko To: sip:[email protected] CSeq: 20 INVITE Call-ID: RYw7fDsLDF Max-Forwards: 70 Supported: replaces, outbound Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Content-Length: 562 Contact: <sip:[email protected]>;+sip.instance="<urn:uuid:c03a376f-325c-4764-a9c0-827fb2a23a79>" User-Agent: Linphone/3.7.0 (belle-sip/1.3.0)
v=0 o=301 4060 2139 IN IP4 192.168.220.16 s=Talk c=IN IP4 192.168.220.16 t=0 0 m=audio 7078 RTP/AVP 9 124 111 110 0 8 101 a=rtpmap:124 opus/48000 a=fmtp:124 useinbandfec=1; usedtx=1 a=rtpmap:111 speex/16000 a=fmtp:111 vbr=on a=rtpmap:110 speex/8000 a=fmtp:110 vbr=on a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 m=video 9078 RTP/AVP 102 98 103 99 a=rtpmap:102 H264/90000 a=fmtp:102 profile-level-id=42801F a=rtpmap:98 H263-1998/90000 a=fmtp:98 CIF=1;QCIF=1 a=rtpmap:103 VP8/90000 a=rtpmap:99 MP4V-ES/90000 a=fmtp:99 profile-level-id=3 <-------------> --- (13 headers 22 lines) --- Sending to 10.10.1.144:5060 (no NAT) Sending to 10.10.1.144:5060 (no NAT) Using INVITE request as basis request - RYw7fDsLDF Found peer '301' for '301' from 10.10.1.144:5060 == Using SIP VIDEO CoS mark 6 == Using SIP RTP CoS mark 5 Found RTP audio format 9 Found RTP audio format 124 Found RTP audio format 111 Found RTP audio format 110 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Found audio description format opus for ID 124 Found audio description format speex for ID 111 Found audio description format speex for ID 110 Found audio description format telephone-event for ID 101 Found RTP video format 102 Found RTP video format 98 Found RTP video format 103 Found RTP video format 99 Found video description format H264 for ID 102 Found video description format H263-1998 for ID 98 Found video description format VP8 for ID 103 Found video description format MP4V-ES for ID 99 Capabilities: us - (ulaw|alaw|g722|h264), peer - audio=(ulaw|alaw|speex|speex16|g722|opus)/video=(h263p|h264|mpeg4|vp8)/text=(nothing), combined - (ulaw|alaw|g722|h264) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.220.16:7078 Peer video RTP is at port 192.168.220.16:9078 Peer doesn't provide T.140 Looking for 306 in LocalSets (domain 10.10.1.201) list_route: route/path hop: <sip:[email protected]> <--- Transmitting (NAT) to 10.10.1.144:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.1.144:5060;branch=z9hG4bK.UwYy83FQ9;received=10.10.1.144;rport=5060 From: <sip:[email protected]>;tag=XY9jSoWko To: sip:[email protected] Call-ID: RYw7fDsLDF CSeq: 20 INVITE Server: Asterisk PBX 12.2.0-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: <sip:[email protected]:5060> Content-Length: 0 <------------> -- Executing [306@LocalSets:1] NoOp("SIP/301-0000000a", "dial 306") in new stack -- Executing [306@LocalSets:3] Dial("SIP/301-0000000a", "SIP/306") in new stack == Using SIP VIDEO CoS mark 6 == Using SIP RTP CoS mark 5 Audio is at 14884 Video is at 10.10.1.201:12672 Adding codec 100012 (g722) to SDP Adding codec 100003 (ulaw) to SDP Adding codec 100004 (alaw) to SDP Adding video codec 200004 (h264) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 10.10.1.145:5062: INVITE sip:[email protected]:5062 SIP/2.0 Via: SIP/2.0/UDP 10.10.1.201:5060;branch=z9hG4bK5ed3b996;rport Max-Forwards: 70 From: <sip:[email protected]>;tag=as7052b518 To: <sip:[email protected]:5062> Contact: <sip:[email protected]:5060> Call-ID: [email protected]:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 12.2.0-rc1 Date: Wed, 07 May 2014 12:02:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 433 v=0 o=root 286204086 286204086 IN IP4 10.10.1.201 s=Asterisk PBX 12.2.0-rc1 c=IN IP4 10.10.1.201 b=CT:384 t=0 0 m=audio 14884 RTP/AVP 9 0 8 101 a=rtpmap:9 G722/8000 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=maxptime:150 a=sendrecv m=video 12672 RTP/AVP 99 a=rtpmap:99 H264/90000 a=fmtp:99 profile-level-id=42801F a=sendrecv --- -- Called SIP/306 <--- SIP read from UDP:10.10.1.145:5062 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.1.201:5060;branch=z9hG4bK5ed3b996;rport=5060 From: <sip:[email protected]>;tag=as7052b518 To: <sip:[email protected]:5062> Call-ID: [email protected]:5060 CSeq: 102 INVITE Supported: replaces, path, timer, eventlist User-Agent: Grandstream GXV3175v2 1.0.1.55 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from UDP:10.10.1.145:5062 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.10.1.201:5060;branch=z9hG4bK5ed3b996;rport=5060 From: <sip:[email protected]>;tag=as7052b518 To: <sip:[email protected]:5062>;tag=647474019 Call-ID: [email protected]:5060 CSeq: 102 INVITE Contact: <sip:[email protected]:5062> Supported: replaces, path, timer, eventlist User-Agent: Grandstream GXV3175v2 1.0.1.55 Allow-Events: talk, hold Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 <-------------> --- (12 headers 0 lines) --- list_route: route/path hop: <sip:[email protected]:5062> -- SIP/306-0000000b is ringing <--- Transmitting (NAT) to 10.10.1.144:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.10.1.144:5060;branch=z9hG4bK.UwYy83FQ9;received=10.10.1.144;rport=5060 From: <sip:[email protected]>;tag=XY9jSoWko To: sip:[email protected];tag=as1221478d Call-ID: RYw7fDsLDF CSeq: 20 INVITE Server: Asterisk PBX 12.2.0-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: <sip:[email protected]:5060> Content-Length: 0 <------------> <--- SIP read from UDP:10.10.1.144:5060 ---> <-------------> <--- SIP read from UDP:10.10.1.145:5062 ---> <-------------> Reliably Transmitting (NAT) to 10.10.1.146:34602: OPTIONS sip:[email protected]:34602 SIP/2.0 Via: SIP/2.0/UDP 10.10.1.201:5060;branch=z9hG4bK5ee0f102;rport Max-Forwards: 70 From: "asterisk" <sip:[email protected]>;tag=as28d2335b To: <sip:[email protected]:34602> Contact: <sip:[email protected]:5060> Call-ID: [email protected]:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 12.2.0-rc1 Date: Wed, 07 May 2014 12:03:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- Retransmitting #1 (NAT) to 10.10.1.146:34602: OPTIONS sip:[email protected]:34602 SIP/2.0 Via: SIP/2.0/UDP 10.10.1.201:5060;branch=z9hG4bK5ee0f102;rport Max-Forwards: 70 From: "asterisk" <sip:[email protected]>;tag=as28d2335b To: <sip:[email protected]:34602> Contact: <sip:[email protected]:5060> Call-ID: [email protected]:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 12.2.0-rc1 Date: Wed, 07 May 2014 12:03:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:10.10.1.145:5062 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.1.201:5060;branch=z9hG4bK5ed3b996;rport=5060 From: <sip:[email protected]>;tag=as7052b518 To: <sip:[email protected]:5062>;tag=647474019 Call-ID: [email protected]:5060 CSeq: 102 INVITE Contact: <sip:[email protected]:5062> Supported: replaces, path, timer, eventlist User-Agent: Grandstream GXV3175v2 1.0.1.55 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Type: application/sdp Content-Length: 447 v=0 o=306 8000 8000 IN IP4 10.10.1.145 s=SIP Call c=IN IP4 10.10.1.145 t=0 0 m=audio 45382 RTP/AVP 9 0 8 101 a=sendrecv a=rtpmap:9 G722/8000 a=ptime:20 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=silenceSupp:off - - - - m=video 55076 RTP/AVP 99 b=AS:320 a=sendrecv a=rtpmap:99 H264/90000 a=fmtp:99 profile-level-id=42801F; sprop-parameter-sets=Z0KADJWgUH5A,aM4Ecg==; max-br=320 <-------------> --- (12 headers 19 lines) --- Found RTP audio format 9 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Found audio description format G722 for ID 9 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Found RTP video format 99 Found video description format H264 for ID 99 Capabilities: us - (ulaw|alaw|g722|h264), peer - audio=(ulaw|alaw|g722)/video=(h264)/text=(nothing), combined - (ulaw|alaw|g722|h264) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 10.10.1.145:45382 Peer video RTP is at port 10.10.1.145:55076 Peer doesn't provide T.140 list_route: route/path hop: <sip:[email protected]:5062> Transmitting (NAT) to 10.10.1.145:5062: ACK sip:[email protected]:5062 SIP/2.0 Via: SIP/2.0/UDP 10.10.1.201:5060;branch=z9hG4bK4ff8e080;rport Max-Forwards: 70 From: <sip:[email protected]>;tag=as7052b518 To: <sip:[email protected]:5062>;tag=647474019 Contact: <sip:[email protected]:5060> Call-ID: [email protected]:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 12.2.0-rc1 Content-Length: 0 --- -- SIP/306-0000000b answered SIP/301-0000000a Audio is at 10078 Video is at 10.10.1.201:16536 Adding codec 100012 (g722) to SDP Adding codec 100003 (ulaw) to SDP Adding codec 100004 (alaw) to SDP Adding video codec 200004 (h264) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (NAT) to 10.10.1.144:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.1.144:5060;branch=z9hG4bK.UwYy83FQ9;received=10.10.1.144;rport=5060 From: <sip:[email protected]>;tag=XY9jSoWko To: sip:[email protected];tag=as1221478d Call-ID: RYw7fDsLDF CSeq: 20 INVITE Server: Asterisk PBX 12.2.0-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: <sip:[email protected]:5060> Content-Type: application/sdp Content-Length: 438 v=0 o=root 1048282658 1048282658 IN IP4 10.10.1.201 s=Asterisk PBX 12.2.0-rc1 c=IN IP4 10.10.1.201 b=CT:384 t=0 0 m=audio 10078 RTP/AVP 9 0 8 101 a=rtpmap:9 G722/8000 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=maxptime:150 a=sendrecv m=video 16536 RTP/AVP 102 a=rtpmap:102 H264/90000 a=fmtp:102 profile-level-id=42801F a=sendrecv <------------> -- Channel SIP/301-0000000a joined 'simple_bridge' basic-bridge <057aca53-4ef0-4568-b312-ff6e032da9d3> <--- SIP read from UDP:10.10.1.144:5060 ---> ACK sip:[email protected]:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.1.144:5060;rport;branch=z9hG4bK.KNApcdgKs From: <sip:[email protected]>;tag=XY9jSoWko To: <sip:[email protected]>;tag=as1221478d CSeq: 20 ACK Call-ID: RYw7fDsLDF Max-Forwards: 70 <-------------> --- (7 headers 0 lines) --- <--- SIP read from UDP:10.10.1.144:5060 ---> PUBLISH sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 10.10.1.144:5060;branch=z9hG4bK.AZJ6xMUEY;rport From: <sip:[email protected]>;tag=TOdTkgRTc To: sip:[email protected] CSeq: 27 PUBLISH Call-ID: LErqEXzaBt Max-Forwards: 70 Supported: replaces, outbound Event: presence Expires: 3600 User-Agent: Linphone/3.7.0 (belle-sip/1.3.0) Content-Type: application/pidf+xml Content-Length: 514 <?xml version="1.0" encoding="UTF-8"?> <presence xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" entity="sip:[email protected]" xmlns="urn:ietf:params:xml:ns:pidf"><tuple id="yhk46v"><status><basic>open</basic></status><contact priority="0.8">sip:[email protected]</contact><timestamp>2014-05-07T12:03:06Z</timestamp></tuple><dm:person id="8rrehq"><rpid:activities><rpid:on-the-phone/></rpid:activities><timestamp>2014-05-07T12:03:06Z</timestamp></dm:person></presence> <-------------> --- (13 headers 2 lines) --- Sending to 10.10.1.144:5060 (no NAT) <--- Transmitting (no NAT) to 10.10.1.144:5060 ---> SIP/2.0 489 Bad Event Via: SIP/2.0/UDP 10.10.1.144:5060;branch=z9hG4bK.AZJ6xMUEY;received=10.10.1.144;rport=5060 From: <sip:[email protected]>;tag=TOdTkgRTc To: sip:[email protected];tag=as54e8d3b1 Call-ID: LErqEXzaBt CSeq: 27 PUBLISH Server: Asterisk PBX 12.2.0-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> Really destroying SIP dialog 'LErqEXzaBt' Method: PUBLISH -- Channel SIP/306-0000000b joined 'simple_bridge' basic-bridge <057aca53-4ef0-4568-b312-ff6e032da9d3> > Bridge 057aca53-4ef0-4568-b312-ff6e032da9d3: switching from simple_bridge technology to native_rtp > 0x7f84dc00df30 -- Probation passed - setting RTP source address to 10.10.1.144:7078 > 0x7f84dc00df30 -- Probation passed - setting RTP source address to 10.10.1.144:7078 Retransmitting #2 (NAT) to 10.10.1.146:34602: OPTIONS sip:[email protected]:34602 SIP/2.0 Via: SIP/2.0/UDP 10.10.1.201:5060;branch=z9hG4bK5ee0f102;rport Max-Forwards: 70 From: "asterisk" <sip:[email protected]>;tag=as28d2335b To: <sip:[email protected]:34602> Contact: <sip:[email protected]:5060> Call-ID: [email protected]:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 12.2.0-rc1 Date: Wed, 07 May 2014 12:03:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- > 0x7f84dc010ef0 -- Probation passed - setting RTP source address to 10.10.1.144:9078 > 0x7f84e8004660 -- Probation passed - setting RTP source address to 10.10.1.145:45382 > 0x7f84dc010ef0 -- Probation passed - setting RTP source address to 10.10.1.144:9078 > 0x7f84e800c180 -- Probation passed - setting RTP source address to 10.10.1.145:55076 Retransmitting #3 (NAT) to 10.10.1.146:34602: OPTIONS sip:[email protected]:34602 SIP/2.0 Via: SIP/2.0/UDP 10.10.1.201:5060;branch=z9hG4bK5ee0f102;rport Max-Forwards: 70 From: "asterisk" <sip:[email protected]>;tag=as28d2335b To: <sip:[email protected]:34602> Contact: <sip:[email protected]:5060> Call-ID: [email protected]:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 12.2.0-rc1 Date: Wed, 07 May 2014 12:03:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- Retransmitting #4 (NAT) to 10.10.1.146:34602: OPTIONS sip:[email protected]:34602 SIP/2.0 Via: SIP/2.0/UDP 10.10.1.201:5060;branch=z9hG4bK5ee0f102;rport Max-Forwards: 70 From: "asterisk" <sip:[email protected]>;tag=as28d2335b To: <sip:[email protected]:34602> Contact: <sip:[email protected]:5060> Call-ID: [email protected]:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 12.2.0-rc1 Date: Wed, 07 May 2014 12:03:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS <--- SIP read from UDP:10.10.1.145:5062 ---> BYE sip:[email protected]:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.1.145:5062;branch=z9hG4bK651537380;rport From: <sip:[email protected]:5062>;tag=647474019 To: <sip:[email protected]>;tag=as7052b518 Call-ID: [email protected]:5060 CSeq: 103 BYE Contact: <sip:[email protected]:5062> Max-Forwards: 70 Supported: replaces, path, timer, eventlist User-Agent: Grandstream GXV3175v2 1.0.1.55 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Sending to 10.10.1.145:5062 (NAT) Scheduling destruction of SIP dialog '[email protected]:5060' in 6400 ms (Method: BYE) <--- Transmitting (NAT) to 10.10.1.145:5062 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.1.145:5062;branch=z9hG4bK651537380;received=10.10.1.145;rport=5062 From: <sip:[email protected]:5062>;tag=647474019 To: <sip:[email protected]>;tag=as7052b518 Call-ID: [email protected]:5060 CSeq: 103 BYE Server: Asterisk PBX 12.2.0-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> -- Channel SIP/306-0000000b left 'native_rtp' basic-bridge <057aca53-4ef0-4568-b312-ff6e032da9d3> -- Channel SIP/301-0000000a left 'native_rtp' basic-bridge <057aca53-4ef0-4568-b312-ff6e032da9d3> == Spawn extension (LocalSets, 306, 3) exited non-zero on 'SIP/301-0000000a' Scheduling destruction of SIP dialog 'RYw7fDsLDF' in 6400 ms (Method: ACK) Reliably Transmitting (NAT) to 10.10.1.144:5060: BYE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 10.10.1.201:5060;branch=z9hG4bK30d73897;rport Max-Forwards: 70 From: sip:[email protected];tag=as1221478d To: <sip:[email protected]>;tag=XY9jSoWko Call-ID: RYw7fDsLDF CSeq: 102 BYE User-Agent: Asterisk PBX 12.2.0-rc1 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- <--- SIP read from UDP:10.10.1.144:5060 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 10.10.1.201:5060;branch=z9hG4bK30d73897;rport From: <sip:[email protected]>;tag=as1221478d To: <sip:[email protected]>;tag=XY9jSoWko Call-ID: RYw7fDsLDF CSeq: 102 BYE User-Agent: Linphone/3.7.0 (belle-sip/1.3.0) Supported: replaces, outbound <-------------> --- (8 headers 0 lines) --- SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog 'RYw7fDsLDF' Method: ACK <--- SIP read from UDP:10.10.1.144:5060 ---> PUBLISH sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 10.10.1.144:5060;branch=z9hG4bK.uy~NP6rxW;rport From: <sip:[email protected]>;tag=TOdTkgRTc To: sip:[email protected] CSeq: 28 PUBLISH Call-ID: LErqEXzaBt Max-Forwards: 70 Supported: replaces, outbound Event: presence Expires: 3600 User-Agent: Linphone/3.7.0 (belle-sip/1.3.0) Content-Type: application/pidf+xml Content-Length: 381 <?xml version="1.0" encoding="UTF-8"?> <presence xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" entity="sip:[email protected]" xmlns="urn:ietf:params:xml:ns:pidf"><tuple id="nqcpwx"><status><basic>open</basic></status><contact priority="0.8">sip:[email protected]</contact><timestamp>2014-05-07T11:55:33Z</timestamp></tuple></presence> <-------------> --- (13 headers 2 lines) --- Sending to 10.10.1.144:5060 (no NAT) <--- Transmitting (no NAT) to 10.10.1.144:5060 ---> SIP/2.0 489 Bad Event Via: SIP/2.0/UDP 10.10.1.144:5060;branch=z9hG4bK.uy~NP6rxW;received=10.10.1.144;rport=5060 From: <sip:[email protected]>;tag=TOdTkgRTc To: sip:[email protected];tag=as0a7851b1 Call-ID: LErqEXzaBt CSeq: 28 PUBLISH Server: Asterisk PBX 12.2.0-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> Really destroying SIP dialog 'LErqEXzaBt' Method: PUBLISH <--- SIP read from UDP:10.10.1.144:5060 ---> <-------------> Reliably Transmitting (NAT) to 10.10.1.146:34602: OPTIONS sip:[email protected]:34602 SIP/2.0 Via: SIP/2.0/UDP 10.10.1.201:5060;branch=z9hG4bK2952aad8;rport Max-Forwards: 70 From: "asterisk" <sip:[email protected]>;tag=as2e98e406 To: <sip:[email protected]:34602> Contact: <sip:[email protected]:5060> Call-ID: [email protected]:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 12.2.0-rc1 Date: Wed, 07 May 2014 12:03:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- Retransmitting #1 (NAT) to 10.10.1.146:34602: OPTIONS sip:[email protected]:34602 SIP/2.0 Via: SIP/2.0/UDP 10.10.1.201:5060;branch=z9hG4bK2952aad8;rport Max-Forwards: 70 From: "asterisk" <sip:[email protected]>;tag=as2e98e406 To: <sip:[email protected]:34602> Contact: <sip:[email protected]:5060> Call-ID: [email protected]:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 12.2.0-rc1 Date: Wed, 07 May 2014 12:03:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- Really destroying SIP dialog '[email protected]:5060' Method: BYE Retransmitting #2 (NAT) to 10.10.1.146:34602: OPTIONS sip:[email protected]:34602 SIP/2.0 Via: SIP/2.0/UDP 10.10.1.201:5060;branch=z9hG4bK2952aad8;rport Max-Forwards: 70 From: "asterisk" <sip:[email protected]>;tag=as2e98e406 To: <sip:[email protected]:34602> Contact: <sip:[email protected]:5060> Call-ID: [email protected]:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 12.2.0-rc1 Date: Wed, 07 May 2014 12:03:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- Retransmitting #3 (NAT) to 10.10.1.146:34602: OPTIONS sip:[email protected]:34602 SIP/2.0 Via: SIP/2.0/UDP 10.10.1.201:5060;branch=z9hG4bK2952aad8;rport Max-Forwards: 70 From: "asterisk" <sip:[email protected]>;tag=as2e98e406 To: <sip:[email protected]:34602> Contact: <sip:[email protected]:5060> Call-ID: [email protected]:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 12.2.0-rc1 Date: Wed, 07 May 2014 12:03:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- Retransmitting #4 (NAT) to 10.10.1.146:34602: OPTIONS sip:[email protected]:34602 SIP/2.0 Via: SIP/2.0/UDP 10.10.1.201:5060;branch=z9hG4bK2952aad8;rport Max-Forwards: 70 From: "asterisk" <sip:[email protected]>;tag=as2e98e406 To: <sip:[email protected]:34602> Contact: <sip:[email protected]:5060> Call-ID: [email protected]:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 12.2.0-rc1 Date: Wed, 07 May 2014 12:03:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS <--- SIP read from UDP:10.10.1.144:5060 ---> <-------------> <--- SIP read from UDP:10.10.1.145:5062 ---> <-------------> Reliably Transmitting (NAT) to 10.10.1.145:5062: OPTIONS sip:[email protected]:5062 SIP/2.0 Via: SIP/2.0/UDP 10.10.1.201:5060;branch=z9hG4bK1bab16d3;rport Max-Forwards: 70 From: "asterisk" <sip:[email protected]>;tag=as0733973c To: <sip:[email protected]:5062> Contact: <sip:[email protected]:5060> Call-ID: [email protected]:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 12.2.0-rc1 Date: Wed, 07 May 2014 12:03:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:10.10.1.145:5062 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.1.201:5060;branch=z9hG4bK1bab16d3;rport=5060 From: "asterisk" <sip:[email protected]>;tag=as0733973c To: <sip:[email protected]:5062>;tag=1217888556 Call-ID: [email protected]:5060 CSeq: 102 OPTIONS Supported: replaces, path, timer, eventlist User-Agent: Grandstream GXV3175v2 1.0.1.55 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS Reliably Transmitting (NAT) to 10.10.1.146:34602: OPTIONS sip:[email protected]:34602 SIP/2.0 Via: SIP/2.0/UDP 10.10.1.201:5060;branch=z9hG4bK69e9074b;rport Max-Forwards: 70 From: "asterisk" <sip:[email protected]>;tag=as2d557dca To: <sip:[email protected]:34602> Contact: <sip:[email protected]:5060> Call-ID: [email protected]:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 12.2.0-rc1 Date: Wed, 07 May 2014 12:03:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- Retransmitting #1 (NAT) to 10.10.1.146:34602: OPTIONS sip:[email protected]:34602 SIP/2.0 Via: SIP/2.0/UDP 10.10.1.201:5060;branch=z9hG4bK69e9074b;rport Max-Forwards: 70 From: "asterisk" <sip:[email protected]>;tag=as2d557dca To: <sip:[email protected]:34602> Contact: <sip:[email protected]:5060> Call-ID: [email protected]:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 12.2.0-rc1 Date: Wed, 07 May 2014 12:03:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- Reliably Transmitting (NAT) to 10.10.1.144:5060: OPTIONS sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 10.10.1.201:5060;branch=z9hG4bK716d9273;rport Max-Forwards: 70 From: "asterisk" <sip:[email protected]>;tag=as7d31d5fe To: <sip:[email protected]> Contact: <sip:[email protected]:5060> Call-ID: [email protected]:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 12.2.0-rc1 Date: Wed, 07 May 2014 12:03:34 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:10.10.1.144:5060 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 10.10.1.201:5060;branch=z9hG4bK716d9273;rport From: "asterisk" <sip:[email protected]>;tag=as7d31d5fe To: <sip:[email protected]>;tag=slu9L Call-ID: [email protected]:5060 CSeq: 102 OPTIONS <-------------> --- (6 headers 0 lines) --- Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS <--- SIP read from UDP:10.10.1.144:5060 ---> <-------------> Retransmitting #2 (NAT) to 10.10.1.146:34602: OPTIONS sip:[email protected]:34602 SIP/2.0 Via: SIP/2.0/UDP 10.10.1.201:5060;branch=z9hG4bK69e9074b;rport Max-Forwards: 70 From: "asterisk" <sip:[email protected]>;tag=as2d557dca To: <sip:[email protected]:34602> Contact: <sip:[email protected]:5060> Call-ID: [email protected]:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 12.2.0-rc1 Date: Wed, 07 May 2014 12:03:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- Retransmitting #3 (NAT) to 10.10.1.146:34602: OPTIONS sip:[email protected]:34602 SIP/2.0 Via: SIP/2.0/UDP 10.10.1.201:5060;branch=z9hG4bK69e9074b;rport Max-Forwards: 70 From: "asterisk" <sip:[email protected]>;tag=as2d557dca To: <sip:[email protected]:34602> Contact: <sip:[email protected]:5060> Call-ID: [email protected]:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 12.2.0-rc1 Date: Wed, 07 May 2014 12:03:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- Retransmitting #4 (NAT) to 10.10.1.146:34602: OPTIONS sip:[email protected]:34602 SIP/2.0 Via: SIP/2.0/UDP 10.10.1.201:5060;branch=z9hG4bK69e9074b;rport Max-Forwards: 70 From: "asterisk" <sip:[email protected]>;tag=as2d557dca To: <sip:[email protected]:34602> Contact: <sip:[email protected]:5060> Call-ID: [email protected]:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 12.2.0-rc1 Date: Wed, 07 May 2014 12:03:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
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