*CLI> sip set debug on SIP Debugging enabled *CLI> rtp set debug on RTP Debugging Enabled *CLI> <--- SIP read from UDP:192.168.70.10:2488 ---> INVITE sip:409@asterisk.phone.lab403.neis.khabarovsk.su SIP/2.0 Via: SIP/2.0/UDP 192.168.70.10:2488;branch=z9hG4bK-d8754z-55ab472ac5bbf4fb-1---d8754z-;rport Max-Forwards: 70 Contact: To: From: "XLite_PM10";tag=ea0a4c5e Call-ID: MzQwNTFlYjczOTgzMDlhZDc2ZWFlNTcyZjNhNmY1ODM. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces User-Agent: X-Lite 4 release 4.0 stamp 58832 Content-Length: 709 v=0 o=- 12942239636841929 1 IN IP4 192.168.70.10 s=CounterPath X-Lite 4.0 c=IN IP4 192.168.70.10 t=0 0 a=ice-ufrag:b87c84 a=ice-pwd:5fb4d12e15ed523478a7d5a7bce4cd14 m=audio 54758 RTP/AVP 107 6 98 8 3 101 a=rtpmap:107 BV32/16000 a=rtpmap:98 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=candidate:1 1 UDP 659136 192.168.70.10 54758 typ host a=candidate:1 2 UDP 659134 192.168.70.10 54759 typ host m=video 65162 RTP/AVP 34 115 a=rtpmap:34 H263/90000 a=fmtp:34 QCIF=2;CIF=3 a=rtpmap:115 H263-1998/90000 a=fmtp:115 QCIF=2;CIF=3;I=1;J=1;T=1 a=sendrecv a=candidate:1 1 UDP 659136 192.168.70.10 65162 typ host a=candidate:1 2 UDP 659134 192.168.70.10 65163 typ host <-------------> --- (13 headers 23 lines) --- == Using UDPTL TOS bits 184 == Using UDPTL CoS mark 5 Sending to 192.168.70.10:2488 (no NAT) Using INVITE request as basis request - MzQwNTFlYjczOTgzMDlhZDc2ZWFlNTcyZjNhNmY1ODM. Found peer '410' for '410' from 192.168.70.10:2488 <--- Reliably Transmitting (no NAT) to 192.168.70.10:2488 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.70.10:2488;branch=z9hG4bK-d8754z-55ab472ac5bbf4fb-1---d8754z-;received=192.168.70.10;rport=2488 From: "XLite_PM10";tag=ea0a4c5e To: ;tag=as42e951b1 Call-ID: MzQwNTFlYjczOTgzMDlhZDc2ZWFlNTcyZjNhNmY1ODM. CSeq: 1 INVITE Server: Asterisk PBX SVN--r Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="phone.lab403.neis.khabarovsk.su", nonce="71b609bf" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'MzQwNTFlYjczOTgzMDlhZDc2ZWFlNTcyZjNhNmY1ODM.' in 6400 ms (Method: INVITE) <--- SIP read from UDP:192.168.70.10:2488 ---> ACK sip:409@asterisk.phone.lab403.neis.khabarovsk.su SIP/2.0 Via: SIP/2.0/UDP 192.168.70.10:2488;branch=z9hG4bK-d8754z-55ab472ac5bbf4fb-1---d8754z-;rport Max-Forwards: 70 To: ;tag=as42e951b1 From: "XLite_PM10";tag=ea0a4c5e Call-ID: MzQwNTFlYjczOTgzMDlhZDc2ZWFlNTcyZjNhNmY1ODM. CSeq: 1 ACK Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from UDP:192.168.70.10:2488 ---> INVITE sip:409@asterisk.phone.lab403.neis.khabarovsk.su SIP/2.0 Via: SIP/2.0/UDP 192.168.70.10:2488;branch=z9hG4bK-d8754z-b5d48c674c97ec35-1---d8754z-;rport Max-Forwards: 70 Contact: To: From: "XLite_PM10";tag=ea0a4c5e Call-ID: MzQwNTFlYjczOTgzMDlhZDc2ZWFlNTcyZjNhNmY1ODM. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces User-Agent: X-Lite 4 release 4.0 stamp 58832 Authorization: Digest username="410",realm="phone.lab403.neis.khabarovsk.su",nonce="71b609bf",uri="sip:409@asterisk.phone.lab403.neis.khabarovsk.su",response="80ab0454f89273aa2d23af2875cacf1a",algorithm=MD5 Content-Length: 709 v=0 o=- 12942239636841929 1 IN IP4 192.168.70.10 s=CounterPath X-Lite 4.0 c=IN IP4 192.168.70.10 t=0 0 a=ice-ufrag:b87c84 a=ice-pwd:5fb4d12e15ed523478a7d5a7bce4cd14 m=audio 54758 RTP/AVP 107 6 98 8 3 101 a=rtpmap:107 BV32/16000 a=rtpmap:98 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=candidate:1 1 UDP 659136 192.168.70.10 54758 typ host a=candidate:1 2 UDP 659134 192.168.70.10 54759 typ host m=video 65162 RTP/AVP 34 115 a=rtpmap:34 H263/90000 a=fmtp:34 QCIF=2;CIF=3 a=rtpmap:115 H263-1998/90000 a=fmtp:115 QCIF=2;CIF=3;I=1;J=1;T=1 a=sendrecv a=candidate:1 1 UDP 659136 192.168.70.10 65162 typ host a=candidate:1 2 UDP 659134 192.168.70.10 65163 typ host <-------------> --- (14 headers 23 lines) --- Sending to 192.168.70.10:2488 (no NAT) Using INVITE request as basis request - MzQwNTFlYjczOTgzMDlhZDc2ZWFlNTcyZjNhNmY1ODM. Found peer '410' for '410' from 192.168.70.10:2488 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 Found RTP audio format 107 Found RTP audio format 6 Found RTP audio format 98 Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 101 Found audio description format BV32 for ID 107 Found audio description format iLBC for ID 98 Found audio description format telephone-event for ID 101 Found RTP video format 34 Found RTP video format 115 Found video description format H263 for ID 34 Found video description format H263-1998 for ID 115 Capabilities: us - (g723|gsm|alaw|lpc10|g729|speex|ilbc|h261|h263|h263p|h264), peer - audio=(gsm|alaw|adpcm|ilbc)/video=(h263|h263p)/text=(nothing), combined - (gsm|alaw|ilbc|h263|h263p) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.70.10:54758 Peer video RTP is at port 192.168.70.10:65162 Peer doesn't provide T.140 Looking for 409 in phones (domain asterisk.phone.lab403.neis.khabarovsk.su) list_route: hop: <--- Transmitting (no NAT) to 192.168.70.10:2488 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.70.10:2488;branch=z9hG4bK-d8754z-b5d48c674c97ec35-1---d8754z-;received=192.168.70.10;rport=2488 From: "XLite_PM10";tag=ea0a4c5e To: Call-ID: MzQwNTFlYjczOTgzMDlhZDc2ZWFlNTcyZjNhNmY1ODM. CSeq: 2 INVITE Server: Asterisk PBX SVN--r Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> -- Executing [409@phones:1] Gosub("SIP/410-00000000", "409,stdexten(SIP/409)") in new stack -- Executing [409@phones:50000] NoOp("SIP/410-00000000", "Start stdexten") in new stack -- Executing [409@phones:50001] Set("SIP/410-00000000", "LOCAL(ext)=409") in new stack -- Executing [409@phones:50002] Set("SIP/410-00000000", "LOCAL(dev)=SIP/409") in new stack -- Executing [409@phones:50003] Set("SIP/410-00000000", "LOCAL(cntx)=") in new stack -- Executing [409@phones:50004] Set("SIP/410-00000000", "LOCAL(mbx)="409"""") in new stack -- Executing [409@phones:50005] Dial("SIP/410-00000000", "SIP/409,30") in new stack == Using UDPTL TOS bits 184 == Using UDPTL CoS mark 5 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 Audio is at 5060 Video is at 192.168.70.51:5060 Adding codec 100004 (alaw) to SDP Adding codec 100009 (g729) to SDP Adding codec 100001 (g723) to SDP Adding codec 100011 (ilbc) to SDP Adding codec 100002 (gsm) to SDP Adding codec 100008 (lpc10) to SDP Adding codec 100010 (speex) to SDP Adding video codec 200004 (h264) to SDP Adding video codec 200003 (h263p) to SDP Adding video codec 200002 (h263) to SDP Adding video codec 200001 (h261) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.70.9:41692: INVITE sip:409@192.168.70.9:41692;rinstance=fa41035e9e5da8ea SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK6c1ddfc6 Max-Forwards: 70 From: "PM10 Soft-Phone" ;tag=as2db312a1 To: Contact: Call-ID: 586148e5479a959b1042d76e3ed77c7c@192.168.70.51:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX SVN--r Date: Tue, 15 Feb 2011 10:33:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Remote-Party-ID: "PM10 Soft-Phone" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 605 v=0 o=root 307637014 307637014 IN IP4 192.168.70.51 s=Asterisk PBX SVN--r c=IN IP4 192.168.70.51 b=CT:8192 t=0 0 m=audio 18358 RTP/AVP 8 18 4 97 3 7 110 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:7 LPC/8000 a=rtpmap:110 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv m=video 12910 RTP/AVP 99 98 34 31 a=rtpmap:99 H264/90000 a=rtpmap:98 h263-1998/90000 a=rtpmap:34 H263/90000 a=rtpmap:31 H261/90000 a=sendrecv --- -- Called 409 Retransmitting #1 (no NAT) to 192.168.70.9:41692: INVITE sip:409@192.168.70.9:41692;rinstance=fa41035e9e5da8ea SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK6c1ddfc6 Max-Forwards: 70 From: "PM10 Soft-Phone" ;tag=as2db312a1 To: Contact: Call-ID: 586148e5479a959b1042d76e3ed77c7c@192.168.70.51:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX SVN--r Date: Tue, 15 Feb 2011 10:33:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Remote-Party-ID: "PM10 Soft-Phone" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 605 v=0 o=root 307637014 307637014 IN IP4 192.168.70.51 s=Asterisk PBX SVN--r c=IN IP4 192.168.70.51 b=CT:8192 t=0 0 m=audio 18358 RTP/AVP 8 18 4 97 3 7 110 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:7 LPC/8000 a=rtpmap:110 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv m=video 12910 RTP/AVP 99 98 34 31 a=rtpmap:99 H264/90000 a=rtpmap:98 h263-1998/90000 a=rtpmap:34 H263/90000 a=rtpmap:31 H261/90000 a=sendrecv --- <--- SIP read from UDP:192.168.70.9:41692 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK6c1ddfc6 To: From: "PM10 Soft-Phone" ;tag=as2db312a1 Call-ID: 586148e5479a959b1042d76e3ed77c7c@192.168.70.51:5060 CSeq: 102 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- <--- SIP read from UDP:192.168.70.9:41692 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK6c1ddfc6 To: From: "PM10 Soft-Phone" ;tag=as2db312a1 Call-ID: 586148e5479a959b1042d76e3ed77c7c@192.168.70.51:5060 CSeq: 102 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- <--- SIP read from UDP:192.168.70.9:41692 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK6c1ddfc6 Contact: To: ;tag=a3dc8a32 From: "PM10 Soft-Phone";tag=as2db312a1 Call-ID: 586148e5479a959b1042d76e3ed77c7c@192.168.70.51:5060 CSeq: 102 INVITE User-Agent: X-Lite 4 release 4.0 stamp 58832 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- -- SIP/409-00000001 is ringing <--- Transmitting (no NAT) to 192.168.70.10:2488 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.70.10:2488;branch=z9hG4bK-d8754z-b5d48c674c97ec35-1---d8754z-;received=192.168.70.10;rport=2488 From: "XLite_PM10";tag=ea0a4c5e To: ;tag=as66ed85cf Call-ID: MzQwNTFlYjczOTgzMDlhZDc2ZWFlNTcyZjNhNmY1ODM. CSeq: 2 INVITE Server: Asterisk PBX SVN--r Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> <--- SIP read from UDP:192.168.70.9:41692 ---> SUBSCRIBE sip:asterisk@192.168.70.51:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.70.9:41692;branch=z9hG4bK-d8754z-f4aa0b50e4858ea3-1---d8754z-;rport Max-Forwards: 70 Contact: To: "XLite_PM9";tag=as3ca3ec76 From: "XLite_PM9";tag=ad9685c2 Call-ID: Yjk1NWI4NjA0Y2Y0MjM0ZWNlMGQwMjk0ZmNiY2YzMTI. CSeq: 3 SUBSCRIBE Expires: 300 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite 4 release 4.0 stamp 58832 Authorization: Digest username="409",realm="phone.lab403.neis.khabarovsk.su",nonce="33967696",uri="sip:asterisk@192.168.70.51:5060",response="622f5be0e4eea21a75d8fb3d1130c4c8",algorithm=MD5 Event: message-summary Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Found peer '409' for '409' from 192.168.70.9:41692 [Feb 15 20:33:58] NOTICE[26717]: chan_sip.c:13792 check_auth: Correct auth, but based on stale nonce received from '"XLite_PM9";tag=ad9685c2' <--- Transmitting (no NAT) to 192.168.70.9:41692 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.70.9:41692;branch=z9hG4bK-d8754z-f4aa0b50e4858ea3-1---d8754z-;received=192.168.70.9;rport=41692 From: "XLite_PM9";tag=ad9685c2 To: "XLite_PM9";tag=as3ca3ec76 Call-ID: Yjk1NWI4NjA0Y2Y0MjM0ZWNlMGQwMjk0ZmNiY2YzMTI. CSeq: 3 SUBSCRIBE Server: Asterisk PBX SVN--r Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="phone.lab403.neis.khabarovsk.su", nonce="079df9e7", stale=true Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'Yjk1NWI4NjA0Y2Y0MjM0ZWNlMGQwMjk0ZmNiY2YzMTI.' in 6400 ms (Method: SUBSCRIBE) <--- SIP read from UDP:192.168.70.9:41692 ---> SUBSCRIBE sip:asterisk@192.168.70.51:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.70.9:41692;branch=z9hG4bK-d8754z-6ebe0c68981be3ae-1---d8754z-;rport Max-Forwards: 70 Contact: To: "XLite_PM9";tag=as3ca3ec76 From: "XLite_PM9";tag=ad9685c2 Call-ID: Yjk1NWI4NjA0Y2Y0MjM0ZWNlMGQwMjk0ZmNiY2YzMTI. CSeq: 4 SUBSCRIBE Expires: 300 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite 4 release 4.0 stamp 58832 Authorization: Digest username="409",realm="phone.lab403.neis.khabarovsk.su",nonce="079df9e7",uri="sip:asterisk@192.168.70.51:5060",response="68470e50b6d52d1a515df339d1e1dd18",algorithm=MD5 Event: message-summary Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Found peer '409' for '409' from 192.168.70.9:41692 Scheduling destruction of SIP dialog 'Yjk1NWI4NjA0Y2Y0MjM0ZWNlMGQwMjk0ZmNiY2YzMTI.' in 310000 ms (Method: SUBSCRIBE) <--- Transmitting (no NAT) to 192.168.70.9:41692 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.70.9:41692;branch=z9hG4bK-d8754z-6ebe0c68981be3ae-1---d8754z-;received=192.168.70.9;rport=41692 From: "XLite_PM9";tag=ad9685c2 To: "XLite_PM9";tag=as3ca3ec76 Call-ID: Yjk1NWI4NjA0Y2Y0MjM0ZWNlMGQwMjk0ZmNiY2YzMTI. CSeq: 4 SUBSCRIBE Server: Asterisk PBX SVN--r Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 300 Contact: ;expires=300 Content-Length: 0 <------------> Reliably Transmitting (no NAT) to 192.168.70.9:41692: NOTIFY sip:409@192.168.70.9:41692 SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK2af13298 Max-Forwards: 70 Route: From: "asterisk" ;tag=as3ca3ec76 To: ;tag=ad9685c2 Contact: Call-ID: Yjk1NWI4NjA0Y2Y0MjM0ZWNlMGQwMjk0ZmNiY2YzMTI. CSeq: 103 NOTIFY User-Agent: Asterisk PBX SVN--r Event: message-summary Content-Type: application/simple-message-summary Subscription-State: active Content-Length: 99 Messages-Waiting: no Message-Account: sip:voicemail@192.168.70.51:5060 Voice-Message: 0/0 (0/0) --- <--- SIP read from UDP:192.168.70.9:41692 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK2af13298 Contact: To: ;tag=ad9685c2 From: "asterisk";tag=as3ca3ec76 Call-ID: Yjk1NWI4NjA0Y2Y0MjM0ZWNlMGQwMjk0ZmNiY2YzMTI. CSeq: 103 NOTIFY User-Agent: X-Lite 4 release 4.0 stamp 58832 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Got RTP packet from 192.168.70.9:57598 (type 08, seq 006203, ts 2946700, len 000160) Got RTP packet from 192.168.70.9:57598 (type 08, seq 006204, ts 2946860, len 000160) Got RTP packet from 192.168.70.9:57598 (type 08, seq 006205, ts 2947020, len 000160) Got RTP packet from 192.168.70.9:57598 (type 08, seq 006206, ts 2947180, len 000160) Got RTP packet from 192.168.70.9:57598 (type 08, seq 006207, ts 2947340, len 000160) Got RTP packet from 192.168.70.9:57598 (type 08, seq 006208, ts 2947500, len 000160) <--- SIP read from UDP:192.168.70.9:41692 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK6c1ddfc6 Contact: To: ;tag=a3dc8a32 From: "PM10 Soft-Phone";tag=as2db312a1 Call-ID: 586148e5479a959b1042d76e3ed77c7c@192.168.70.51:5060 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces User-Agent: X-Lite 4 release 4.0 stamp 58832 Content-Length: 681 v=0 o=- 12942239641798258 1 IN IP4 192.168.70.9 s=CounterPath X-Lite 4.0 c=IN IP4 192.168.70.9 t=0 0 a=ice-ufrag:8043e5 a=ice-pwd:5f2af35950be008cce660153e07db8df m=audio 57598 RTP/AVP 8 97 3 101 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=candidate:1 1 UDP 659136 192.168.70.9 57598 typ host a=candidate:1 2 UDP 659134 192.168.70.9 57599 typ host m=video 55192 RTP/AVP 98 34 a=rtpmap:98 H263-1998/90000 a=fmtp:98 QCIF=1;CIF=1;VGA=1;I=1;J=1;T=1 a=rtpmap:34 H263/90000 a=fmtp:34 QCIF=1;CIF=1;VGA=1 a=sendrecv a=candidate:1 1 UDP 659136 192.168.70.9 55192 typ host a=candidate:1 2 UDP 659134 192.168.70.9 55193 typ host <-------------> --- (12 headers 22 lines) --- Found RTP audio format 8 Found RTP audio format 97 Found RTP audio format 3 Found RTP audio format 101 Found audio description format iLBC for ID 97 Found audio description format telephone-event for ID 101 Found RTP video format 98 Found RTP video format 34 Found video description format H263-1998 for ID 98 Found video description format H263 for ID 34 Capabilities: us - (g723|gsm|alaw|lpc10|g729|speex|ilbc|h261|h263|h263p|h264), peer - audio=(gsm|alaw|ilbc)/video=(h263|h263p)/text=(nothing), combined - (gsm|alaw|ilbc|h263|h263p) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.70.9:57598 Peer video RTP is at port 192.168.70.9:55192 Peer doesn't provide T.140 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.70.9:41692 Transmitting (no NAT) to 192.168.70.9:41692: ACK sip:409@192.168.70.9:41692;rinstance=fa41035e9e5da8ea SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK6fc44739 Max-Forwards: 70 From: "PM10 Soft-Phone" ;tag=as2db312a1 To: ;tag=a3dc8a32 Contact: Call-ID: 586148e5479a959b1042d76e3ed77c7c@192.168.70.51:5060 CSeq: 102 ACK User-Agent: Asterisk PBX SVN--r Content-Length: 0 --- -- SIP/409-00000001 answered SIP/410-00000000 Audio is at 5060 Video is at 192.168.70.51:5060 Adding codec 100004 (alaw) to SDP Adding codec 100011 (ilbc) to SDP Adding codec 100002 (gsm) to SDP Adding video codec 200003 (h263p) to SDP Adding video codec 200002 (h263) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 192.168.70.10:2488 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.70.10:2488;branch=z9hG4bK-d8754z-b5d48c674c97ec35-1---d8754z-;received=192.168.70.10;rport=2488 From: "XLite_PM10";tag=ea0a4c5e To: ;tag=as66ed85cf Call-ID: MzQwNTFlYjczOTgzMDlhZDc2ZWFlNTcyZjNhNmY1ODM. CSeq: 2 INVITE Server: Asterisk PBX SVN--r Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Remote-Party-ID: "PM9 Soft-Phone" ;party=called;privacy=off;screen=no Content-Type: application/sdp Content-Length: 412 v=0 o=root 2004846078 2004846078 IN IP4 192.168.70.51 s=Asterisk PBX SVN--r c=IN IP4 192.168.70.51 b=CT:8192 t=0 0 m=audio 18702 RTP/AVP 8 98 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv m=video 16512 RTP/AVP 115 34 a=rtpmap:115 h263-1998/90000 a=rtpmap:34 H263/90000 a=sendrecv <------------> -- Remotely bridging SIP/410-00000000 and SIP/409-00000001 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.70.9:41692 Audio is at 5060 Video is at 192.168.70.51:5060 Adding codec 100004 (alaw) to SDP Adding codec 100011 (ilbc) to SDP Adding codec 100002 (gsm) to SDP Adding video codec 200003 (h263p) to SDP Adding video codec 200002 (h263) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.70.9:41692: INVITE sip:409@192.168.70.9:41692;rinstance=fa41035e9e5da8ea SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK29c15b98 Max-Forwards: 70 From: "PM10 Soft-Phone" ;tag=as2db312a1 To: ;tag=a3dc8a32 Contact: Call-ID: 586148e5479a959b1042d76e3ed77c7c@192.168.70.51:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX SVN--r Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Remote-Party-ID: "PM10 Soft-Phone" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 408 v=0 o=root 307637014 307637015 IN IP4 192.168.70.10 s=Asterisk PBX SVN--r c=IN IP4 192.168.70.10 b=CT:8192 t=0 0 m=audio 54758 RTP/AVP 8 97 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv m=video 65162 RTP/AVP 98 34 a=rtpmap:98 h263-1998/90000 a=rtpmap:34 H263/90000 a=sendrecv --- Sent RTP P2P packet to 192.168.70.10:54758 (type 08, len 000160) Sent RTP P2P packet to 192.168.70.10:54758 (type 08, len 000160) Sent RTP P2P packet to 192.168.70.10:54758 (type 08, len 000160) Sent RTP P2P packet to 192.168.70.10:54758 (type 08, len 000160) Sent RTP P2P packet to 192.168.70.10:54758 (type 08, len 000160) Retransmitting #1 (no NAT) to 192.168.70.10:2488: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.70.10:2488;branch=z9hG4bK-d8754z-b5d48c674c97ec35-1---d8754z-;received=192.168.70.10;rport=2488 From: "XLite_PM10";tag=ea0a4c5e To: ;tag=as66ed85cf Call-ID: MzQwNTFlYjczOTgzMDlhZDc2ZWFlNTcyZjNhNmY1ODM. CSeq: 2 INVITE Server: Asterisk PBX SVN--r Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Remote-Party-ID: "PM9 Soft-Phone" ;party=called;privacy=off;screen=no Content-Type: application/sdp Content-Length: 412 v=0 o=root 2004846078 2004846078 IN IP4 192.168.70.51 s=Asterisk PBX SVN--r c=IN IP4 192.168.70.51 b=CT:8192 t=0 0 m=audio 18702 RTP/AVP 8 98 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv m=video 16512 RTP/AVP 115 34 a=rtpmap:115 h263-1998/90000 a=rtpmap:34 H263/90000 a=sendrecv --- Sent RTP P2P packet to 192.168.70.10:54758 (type 08, len 000160) Retransmitting #1 (no NAT) to 192.168.70.9:41692: INVITE sip:409@192.168.70.9:41692;rinstance=fa41035e9e5da8ea SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK29c15b98 Max-Forwards: 70 From: "PM10 Soft-Phone" ;tag=as2db312a1 To: ;tag=a3dc8a32 Contact: Call-ID: 586148e5479a959b1042d76e3ed77c7c@192.168.70.51:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX SVN--r Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Remote-Party-ID: "PM10 Soft-Phone" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 408 v=0 o=root 307637014 307637015 IN IP4 192.168.70.10 s=Asterisk PBX SVN--r c=IN IP4 192.168.70.10 b=CT:8192 t=0 0 m=audio 54758 RTP/AVP 8 97 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv m=video 65162 RTP/AVP 98 34 a=rtpmap:98 h263-1998/90000 a=rtpmap:34 H263/90000 a=sendrecv --- Sent RTP P2P packet to 192.168.70.9:57598 (type 08, len 000160) Sent RTP P2P packet to 192.168.70.10:54758 (type 08, len 000160) Sent RTP P2P packet to 192.168.70.9:57598 (type 08, len 000160) Sent RTP P2P packet to 192.168.70.10:54758 (type 08, len 000160) Sent RTP P2P packet to 192.168.70.9:57598 (type 08, len 000160) Sent RTP P2P packet to 192.168.70.9:57598 (type 08, len 000160) Sent RTP P2P packet to 192.168.70.10:54758 (type 08, len 000160) Sent RTP P2P packet to 192.168.70.9:57598 (type 08, len 000160) Sent RTP P2P packet to 192.168.70.9:57598 (type 08, len 000160) Sent RTP P2P packet to 192.168.70.9:57598 (type 08, len 000160) Sent RTP P2P packet to 192.168.70.9:57598 (type 08, len 000160) Sent RTP P2P packet to 192.168.70.9:57598 (type 08, len 000160) Retransmitting #2 (no NAT) to 192.168.70.10:2488: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.70.10:2488;branch=z9hG4bK-d8754z-b5d48c674c97ec35-1---d8754z-;received=192.168.70.10;rport=2488 From: "XLite_PM10";tag=ea0a4c5e To: ;tag=as66ed85cf Call-ID: MzQwNTFlYjczOTgzMDlhZDc2ZWFlNTcyZjNhNmY1ODM. CSeq: 2 INVITE Server: Asterisk PBX SVN--r Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Remote-Party-ID: "PM9 Soft-Phone" ;party=called;privacy=off;screen=no Content-Type: application/sdp Content-Length: 412 v=0 o=root 2004846078 2004846078 IN IP4 192.168.70.51 s=Asterisk PBX SVN--r c=IN IP4 192.168.70.51 b=CT:8192 t=0 0 m=audio 18702 RTP/AVP 8 98 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv m=video 16512 RTP/AVP 115 34 a=rtpmap:115 h263-1998/90000 a=rtpmap:34 H263/90000 a=sendrecv --- Retransmitting #2 (no NAT) to 192.168.70.9:41692: INVITE sip:409@192.168.70.9:41692;rinstance=fa41035e9e5da8ea SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK29c15b98 Max-Forwards: 70 From: "PM10 Soft-Phone" ;tag=as2db312a1 To: ;tag=a3dc8a32 Contact: Call-ID: 586148e5479a959b1042d76e3ed77c7c@192.168.70.51:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX SVN--r Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Remote-Party-ID: "PM10 Soft-Phone" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 408 v=0 o=root 307637014 307637015 IN IP4 192.168.70.10 s=Asterisk PBX SVN--r c=IN IP4 192.168.70.10 b=CT:8192 t=0 0 m=audio 54758 RTP/AVP 8 97 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv m=video 65162 RTP/AVP 98 34 a=rtpmap:98 h263-1998/90000 a=rtpmap:34 H263/90000 a=sendrecv --- Sent RTP P2P packet to 192.168.70.9:57598 (type 08, len 000160) Sent RTP P2P packet to 192.168.70.9:57598 (type 08, len 000160) <--- SIP read from UDP:192.168.70.9:41692 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK29c15b98 Contact: To: ;tag=a3dc8a32 From: "PM10 Soft-Phone";tag=as2db312a1 Call-ID: 586148e5479a959b1042d76e3ed77c7c@192.168.70.51:5060 CSeq: 103 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces User-Agent: X-Lite 4 release 4.0 stamp 58832 Content-Length: 681 v=0 o=- 12942239641798258 2 IN IP4 192.168.70.9 s=CounterPath X-Lite 4.0 c=IN IP4 192.168.70.9 t=0 0 a=ice-ufrag:8043e5 a=ice-pwd:5f2af35950be008cce660153e07db8df m=audio 57598 RTP/AVP 8 97 3 101 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=candidate:1 1 UDP 659136 192.168.70.9 57598 typ host a=candidate:1 2 UDP 659134 192.168.70.9 57599 typ host m=video 55192 RTP/AVP 98 34 a=rtpmap:98 H263-1998/90000 a=fmtp:98 QCIF=1;CIF=1;VGA=1;I=1;J=1;T=1 a=rtpmap:34 H263/90000 a=fmtp:34 QCIF=1;CIF=1;VGA=1 a=sendrecv a=candidate:1 1 UDP 659136 192.168.70.9 55192 typ host a=candidate:1 2 UDP 659134 192.168.70.9 55193 typ host <-------------> --- (12 headers 22 lines) --- Found RTP audio format 8 Found RTP audio format 97 Found RTP audio format 3 Found RTP audio format 101 Found audio description format iLBC for ID 97 Found audio description format telephone-event for ID 101 Found RTP video format 98 Found RTP video format 34 Found video description format H263-1998 for ID 98 Found video description format H263 for ID 34 Capabilities: us - (g723|gsm|alaw|lpc10|g729|speex|ilbc|h261|h263|h263p|h264), peer - audio=(gsm|alaw|ilbc)/video=(h263|h263p)/text=(nothing), combined - (gsm|alaw|ilbc|h263|h263p) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.70.9:57598 Peer video RTP is at port 192.168.70.9:55192 Peer doesn't provide T.140 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.70.9:41692 Transmitting (no NAT) to 192.168.70.9:41692: ACK sip:409@192.168.70.9:41692;rinstance=fa41035e9e5da8ea SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK0cd667a2 Max-Forwards: 70 From: "PM10 Soft-Phone" ;tag=as2db312a1 To: ;tag=a3dc8a32 Contact: Call-ID: 586148e5479a959b1042d76e3ed77c7c@192.168.70.51:5060 CSeq: 103 ACK User-Agent: Asterisk PBX SVN--r Content-Length: 0 --- set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.70.9:41692 Audio is at 5060 Video is at 192.168.70.51:5060 Adding codec 100004 (alaw) to SDP Adding codec 100011 (ilbc) to SDP Adding codec 100002 (gsm) to SDP Adding video codec 200003 (h263p) to SDP Adding video codec 200002 (h263) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.70.9:41692: INVITE sip:409@192.168.70.9:41692;rinstance=fa41035e9e5da8ea SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK1e87b04c Max-Forwards: 70 From: "PM10 Soft-Phone" ;tag=as2db312a1 To: ;tag=a3dc8a32 Contact: Call-ID: 586148e5479a959b1042d76e3ed77c7c@192.168.70.51:5060 CSeq: 104 INVITE User-Agent: Asterisk PBX SVN--r Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Remote-Party-ID: "PM10 Soft-Phone" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 408 v=0 o=root 307637014 307637016 IN IP4 192.168.70.10 s=Asterisk PBX SVN--r c=IN IP4 192.168.70.10 b=CT:8192 t=0 0 m=audio 54758 RTP/AVP 8 97 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv m=video 65162 RTP/AVP 98 34 a=rtpmap:98 h263-1998/90000 a=rtpmap:34 H263/90000 a=sendrecv --- Sent RTP P2P packet to 192.168.70.9:57598 (type 08, len 000160) Sent RTP P2P packet to 192.168.70.9:57598 (type 08, len 000160) Sent RTP P2P packet to 192.168.70.9:57598 (type 08, len 000160) <--- SIP read from UDP:192.168.70.10:2488 ---> ACK sip:409@192.168.70.51:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.70.10:2488;branch=z9hG4bK-d8754z-731d39a473e32698-1---d8754z-;rport Max-Forwards: 70 Contact: To: ;tag=as66ed85cf From: "XLite_PM10";tag=ea0a4c5e Call-ID: MzQwNTFlYjczOTgzMDlhZDc2ZWFlNTcyZjNhNmY1ODM. CSeq: 2 ACK User-Agent: X-Lite 4 release 4.0 stamp 58832 Authorization: Digest username="410",realm="phone.lab403.neis.khabarovsk.su",nonce="71b609bf",uri="sip:409@asterisk.phone.lab403.neis.khabarovsk.su",response="80ab0454f89273aa2d23af2875cacf1a",algorithm=MD5 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.70.10:2488 Audio is at 5060 Video is at 192.168.70.51:5060 Adding codec 100004 (alaw) to SDP Adding codec 100011 (ilbc) to SDP Adding codec 100002 (gsm) to SDP Adding video codec 200003 (h263p) to SDP Adding video codec 200002 (h263) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.70.10:2488: INVITE sip:410@192.168.70.10:2488 SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK3151223b;rport Max-Forwards: 70 From: ;tag=as66ed85cf To: "XLite_PM10";tag=ea0a4c5e Contact: Call-ID: MzQwNTFlYjczOTgzMDlhZDc2ZWFlNTcyZjNhNmY1ODM. CSeq: 102 INVITE User-Agent: Asterisk PBX SVN--r Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Remote-Party-ID: "PM9 Soft-Phone" ;party=called;privacy=off;screen=no Content-Type: application/sdp Content-Length: 410 v=0 o=root 2004846078 2004846079 IN IP4 192.168.70.9 s=Asterisk PBX SVN--r c=IN IP4 192.168.70.9 b=CT:8192 t=0 0 m=audio 57598 RTP/AVP 8 98 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv m=video 55192 RTP/AVP 115 34 a=rtpmap:115 h263-1998/90000 a=rtpmap:34 H263/90000 a=sendrecv --- Sent RTP P2P packet to 192.168.70.9:57598 (type 08, len 000160) <--- SIP read from UDP:192.168.70.10:2488 ---> ACK sip:409@192.168.70.51:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.70.10:2488;branch=z9hG4bK-d8754z-731d39a473e32698-1---d8754z-;rport Max-Forwards: 70 Contact: To: ;tag=as66ed85cf From: "XLite_PM10";tag=ea0a4c5e Call-ID: MzQwNTFlYjczOTgzMDlhZDc2ZWFlNTcyZjNhNmY1ODM. CSeq: 2 ACK User-Agent: X-Lite 4 release 4.0 stamp 58832 Authorization: Digest username="410",realm="phone.lab403.neis.khabarovsk.su",nonce="71b609bf",uri="sip:409@asterisk.phone.lab403.neis.khabarovsk.su",response="80ab0454f89273aa2d23af2875cacf1a",algorithm=MD5 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Sent RTP P2P packet to 192.168.70.9:57598 (type 08, len 000160) Retransmitting #1 (no NAT) to 192.168.70.9:41692: INVITE sip:409@192.168.70.9:41692;rinstance=fa41035e9e5da8ea SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK1e87b04c Max-Forwards: 70 From: "PM10 Soft-Phone" ;tag=as2db312a1 To: ;tag=a3dc8a32 Contact: Call-ID: 586148e5479a959b1042d76e3ed77c7c@192.168.70.51:5060 CSeq: 104 INVITE User-Agent: Asterisk PBX SVN--r Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Remote-Party-ID: "PM10 Soft-Phone" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 408 v=0 o=root 307637014 307637016 IN IP4 192.168.70.10 s=Asterisk PBX SVN--r c=IN IP4 192.168.70.10 b=CT:8192 t=0 0 m=audio 54758 RTP/AVP 8 97 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv m=video 65162 RTP/AVP 98 34 a=rtpmap:98 h263-1998/90000 a=rtpmap:34 H263/90000 a=sendrecv --- Sent RTP P2P packet to 192.168.70.9:57598 (type 08, len 000160) Sent RTP P2P packet to 192.168.70.9:57598 (type 08, len 000160) Sent RTP P2P packet to 192.168.70.9:57598 (type 08, len 000160) Retransmitting #1 (no NAT) to 192.168.70.10:2488: INVITE sip:410@192.168.70.10:2488 SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK3151223b;rport Max-Forwards: 70 From: ;tag=as66ed85cf To: "XLite_PM10";tag=ea0a4c5e Contact: Call-ID: MzQwNTFlYjczOTgzMDlhZDc2ZWFlNTcyZjNhNmY1ODM. CSeq: 102 INVITE User-Agent: Asterisk PBX SVN--r Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Remote-Party-ID: "PM9 Soft-Phone" ;party=called;privacy=off;screen=no Content-Type: application/sdp Content-Length: 410 v=0 o=root 2004846078 2004846079 IN IP4 192.168.70.9 s=Asterisk PBX SVN--r c=IN IP4 192.168.70.9 b=CT:8192 t=0 0 m=audio 57598 RTP/AVP 8 98 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv m=video 55192 RTP/AVP 115 34 a=rtpmap:115 h263-1998/90000 a=rtpmap:34 H263/90000 a=sendrecv --- Sent RTP P2P packet to 192.168.70.9:57598 (type 08, len 000160) Sent RTP P2P packet to 192.168.70.9:57598 (type 08, len 000160) Sent RTP P2P packet to 192.168.70.9:57598 (type 08, len 000160) Got RTP packet from 192.168.70.10:65162 (type 126, seq 000865, ts 000000, len 000004) [Feb 15 20:34:02] NOTICE[26725]: res_rtp_asterisk.c:2190 ast_rtp_read: Unknown RTP codec 126 received from '192.168.70.10:65162' Got RTP packet from 192.168.70.10:65162 (type 126, seq 000865, ts 000000, len 000004) [Feb 15 20:34:02] NOTICE[26725]: res_rtp_asterisk.c:2190 ast_rtp_read: Unknown RTP codec 126 received from '192.168.70.10:65162' Got RTP packet from 192.168.70.10:65162 (type 126, seq 000865, ts 000000, len 000004) [Feb 15 20:34:02] NOTICE[26725]: res_rtp_asterisk.c:2190 ast_rtp_read: Unknown RTP codec 126 received from '192.168.70.10:65162' <--- SIP read from UDP:192.168.70.9:41692 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK1e87b04c Contact: To: ;tag=a3dc8a32 From: "PM10 Soft-Phone";tag=as2db312a1 Call-ID: 586148e5479a959b1042d76e3ed77c7c@192.168.70.51:5060 CSeq: 104 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces User-Agent: X-Lite 4 release 4.0 stamp 58832 Content-Length: 681 v=0 o=- 12942239641798258 3 IN IP4 192.168.70.9 s=CounterPath X-Lite 4.0 c=IN IP4 192.168.70.9 t=0 0 a=ice-ufrag:8043e5 a=ice-pwd:5f2af35950be008cce660153e07db8df m=audio 57598 RTP/AVP 8 97 3 101 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=candidate:1 1 UDP 659136 192.168.70.9 57598 typ host a=candidate:1 2 UDP 659134 192.168.70.9 57599 typ host m=video 55192 RTP/AVP 98 34 a=rtpmap:98 H263-1998/90000 a=fmtp:98 QCIF=1;CIF=1;VGA=1;I=1;J=1;T=1 a=rtpmap:34 H263/90000 a=fmtp:34 QCIF=1;CIF=1;VGA=1 a=sendrecv a=candidate:1 1 UDP 659136 192.168.70.9 55192 typ host a=candidate:1 2 UDP 659134 192.168.70.9 55193 typ host <-------------> --- (12 headers 22 lines) --- Found RTP audio format 8 Found RTP audio format 97 Found RTP audio format 3 Found RTP audio format 101 Found audio description format iLBC for ID 97 Found audio description format telephone-event for ID 101 Found RTP video format 98 Found RTP video format 34 Found video description format H263-1998 for ID 98 Found video description format H263 for ID 34 Capabilities: us - (g723|gsm|alaw|lpc10|g729|speex|ilbc|h261|h263|h263p|h264), peer - audio=(gsm|alaw|ilbc)/video=(h263|h263p)/text=(nothing), combined - (gsm|alaw|ilbc|h263|h263p) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.70.9:57598 Peer video RTP is at port 192.168.70.9:55192 Peer doesn't provide T.140 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.70.9:41692 Transmitting (no NAT) to 192.168.70.9:41692: ACK sip:409@192.168.70.9:41692;rinstance=fa41035e9e5da8ea SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK32265658 Max-Forwards: 70 From: "PM10 Soft-Phone" ;tag=as2db312a1 To: ;tag=a3dc8a32 Contact: Call-ID: 586148e5479a959b1042d76e3ed77c7c@192.168.70.51:5060 CSeq: 104 ACK User-Agent: Asterisk PBX SVN--r Content-Length: 0 --- <--- SIP read from UDP:192.168.70.9:41692 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK1e87b04c Contact: To: ;tag=a3dc8a32 From: "PM10 Soft-Phone";tag=as2db312a1 Call-ID: 586148e5479a959b1042d76e3ed77c7c@192.168.70.51:5060 CSeq: 104 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces User-Agent: X-Lite 4 release 4.0 stamp 58832 Content-Length: 681 v=0 o=- 12942239641798258 3 IN IP4 192.168.70.9 s=CounterPath X-Lite 4.0 c=IN IP4 192.168.70.9 t=0 0 a=ice-ufrag:8043e5 a=ice-pwd:5f2af35950be008cce660153e07db8df m=audio 57598 RTP/AVP 8 97 3 101 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=candidate:1 1 UDP 659136 192.168.70.9 57598 typ host a=candidate:1 2 UDP 659134 192.168.70.9 57599 typ host m=video 55192 RTP/AVP 98 34 a=rtpmap:98 H263-1998/90000 a=fmtp:98 QCIF=1;CIF=1;VGA=1;I=1;J=1;T=1 a=rtpmap:34 H263/90000 a=fmtp:34 QCIF=1;CIF=1;VGA=1 a=sendrecv a=candidate:1 1 UDP 659136 192.168.70.9 55192 typ host a=candidate:1 2 UDP 659134 192.168.70.9 55193 typ host <-------------> --- (12 headers 22 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.70.9:41692 Transmitting (no NAT) to 192.168.70.9:41692: ACK sip:409@192.168.70.9:41692;rinstance=fa41035e9e5da8ea SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK51f834da Max-Forwards: 70 From: "PM10 Soft-Phone" ;tag=as2db312a1 To: ;tag=a3dc8a32 Contact: Call-ID: 586148e5479a959b1042d76e3ed77c7c@192.168.70.51:5060 CSeq: 104 ACK User-Agent: Asterisk PBX SVN--r Content-Length: 0 --- <--- SIP read from UDP:192.168.70.10:2488 ---> ACK sip:409@192.168.70.51:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.70.10:2488;branch=z9hG4bK-d8754z-731d39a473e32698-1---d8754z-;rport Max-Forwards: 70 Contact: To: ;tag=as66ed85cf From: "XLite_PM10";tag=ea0a4c5e Call-ID: MzQwNTFlYjczOTgzMDlhZDc2ZWFlNTcyZjNhNmY1ODM. CSeq: 2 ACK User-Agent: X-Lite 4 release 4.0 stamp 58832 Authorization: Digest username="410",realm="phone.lab403.neis.khabarovsk.su",nonce="71b609bf",uri="sip:409@asterisk.phone.lab403.neis.khabarovsk.su",response="80ab0454f89273aa2d23af2875cacf1a",algorithm=MD5 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from UDP:192.168.70.10:2488 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK3151223b;rport=5060 Contact: To: "XLite_PM10";tag=ea0a4c5e From: ;tag=as66ed85cf Call-ID: MzQwNTFlYjczOTgzMDlhZDc2ZWFlNTcyZjNhNmY1ODM. CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces User-Agent: X-Lite 4 release 4.0 stamp 58832 Content-Length: 678 v=0 o=- 12942239636841929 2 IN IP4 192.168.70.10 s=CounterPath X-Lite 4.0 c=IN IP4 192.168.70.10 t=0 0 a=ice-ufrag:b87c84 a=ice-pwd:5fb4d12e15ed523478a7d5a7bce4cd14 m=audio 54758 RTP/AVP 8 98 3 101 a=rtpmap:98 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=candidate:1 1 UDP 659136 192.168.70.10 54758 typ host a=candidate:1 2 UDP 659134 192.168.70.10 54759 typ host m=video 65162 RTP/AVP 115 34 a=rtpmap:115 H263-1998/90000 a=fmtp:115 QCIF=2;CIF=3;I=1;J=1;T=1 a=rtpmap:34 H263/90000 a=fmtp:34 QCIF=2;CIF=3 a=sendrecv a=candidate:1 1 UDP 659136 192.168.70.10 65162 typ host a=candidate:1 2 UDP 659134 192.168.70.10 65163 typ host <-------------> --- (12 headers 22 lines) --- Found RTP audio format 8 Found RTP audio format 98 Found RTP audio format 3 Found RTP audio format 101 Found audio description format iLBC for ID 98 Found audio description format telephone-event for ID 101 Found RTP video format 115 Found RTP video format 34 Found video description format H263-1998 for ID 115 Found video description format H263 for ID 34 Capabilities: us - (g723|gsm|alaw|lpc10|g729|speex|ilbc|h261|h263|h263p|h264), peer - audio=(gsm|alaw|ilbc)/video=(h263|h263p)/text=(nothing), combined - (gsm|alaw|ilbc|h263|h263p) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.70.10:54758 Peer video RTP is at port 192.168.70.10:65162 Peer doesn't provide T.140 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.70.10:2488 Transmitting (no NAT) to 192.168.70.10:2488: ACK sip:410@192.168.70.10:2488 SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK73fd8142;rport Max-Forwards: 70 From: ;tag=as66ed85cf To: "XLite_PM10";tag=ea0a4c5e Contact: Call-ID: MzQwNTFlYjczOTgzMDlhZDc2ZWFlNTcyZjNhNmY1ODM. CSeq: 102 ACK User-Agent: Asterisk PBX SVN--r Content-Length: 0 --- set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.70.9:41692 Audio is at 5060 Video is at 192.168.70.51:5060 Adding codec 100004 (alaw) to SDP Adding codec 100011 (ilbc) to SDP Adding codec 100002 (gsm) to SDP Adding video codec 200003 (h263p) to SDP Adding video codec 200002 (h263) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.70.9:41692: INVITE sip:409@192.168.70.9:41692;rinstance=fa41035e9e5da8ea SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK1367d200 Max-Forwards: 70 From: "PM10 Soft-Phone" ;tag=as2db312a1 To: ;tag=a3dc8a32 Contact: Call-ID: 586148e5479a959b1042d76e3ed77c7c@192.168.70.51:5060 CSeq: 105 INVITE User-Agent: Asterisk PBX SVN--r Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Remote-Party-ID: "PM10 Soft-Phone" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 408 v=0 o=root 307637014 307637017 IN IP4 192.168.70.10 s=Asterisk PBX SVN--r c=IN IP4 192.168.70.10 b=CT:8192 t=0 0 m=audio 54758 RTP/AVP 8 97 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv m=video 65162 RTP/AVP 98 34 a=rtpmap:98 h263-1998/90000 a=rtpmap:34 H263/90000 a=sendrecv --- Retransmitting #1 (no NAT) to 192.168.70.9:41692: INVITE sip:409@192.168.70.9:41692;rinstance=fa41035e9e5da8ea SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK1367d200 Max-Forwards: 70 From: "PM10 Soft-Phone" ;tag=as2db312a1 To: ;tag=a3dc8a32 Contact: Call-ID: 586148e5479a959b1042d76e3ed77c7c@192.168.70.51:5060 CSeq: 105 INVITE User-Agent: Asterisk PBX SVN--r Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Remote-Party-ID: "PM10 Soft-Phone" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 408 v=0 o=root 307637014 307637017 IN IP4 192.168.70.10 s=Asterisk PBX SVN--r c=IN IP4 192.168.70.10 b=CT:8192 t=0 0 m=audio 54758 RTP/AVP 8 97 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv m=video 65162 RTP/AVP 98 34 a=rtpmap:98 h263-1998/90000 a=rtpmap:34 H263/90000 a=sendrecv --- <--- SIP read from UDP:192.168.70.9:41692 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK1367d200 Contact: To: ;tag=a3dc8a32 From: "PM10 Soft-Phone";tag=as2db312a1 Call-ID: 586148e5479a959b1042d76e3ed77c7c@192.168.70.51:5060 CSeq: 105 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces User-Agent: X-Lite 4 release 4.0 stamp 58832 Content-Length: 681 v=0 o=- 12942239641798258 4 IN IP4 192.168.70.9 s=CounterPath X-Lite 4.0 c=IN IP4 192.168.70.9 t=0 0 a=ice-ufrag:8043e5 a=ice-pwd:5f2af35950be008cce660153e07db8df m=audio 57598 RTP/AVP 8 97 3 101 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=candidate:1 1 UDP 659136 192.168.70.9 57598 typ host a=candidate:1 2 UDP 659134 192.168.70.9 57599 typ host m=video 55192 RTP/AVP 98 34 a=rtpmap:98 H263-1998/90000 a=fmtp:98 QCIF=1;CIF=1;VGA=1;I=1;J=1;T=1 a=rtpmap:34 H263/90000 a=fmtp:34 QCIF=1;CIF=1;VGA=1 a=sendrecv a=candidate:1 1 UDP 659136 192.168.70.9 55192 typ host a=candidate:1 2 UDP 659134 192.168.70.9 55193 typ host <-------------> --- (12 headers 22 lines) --- Found RTP audio format 8 Found RTP audio format 97 Found RTP audio format 3 Found RTP audio format 101 Found audio description format iLBC for ID 97 Found audio description format telephone-event for ID 101 Found RTP video format 98 Found RTP video format 34 Found video description format H263-1998 for ID 98 Found video description format H263 for ID 34 Capabilities: us - (g723|gsm|alaw|lpc10|g729|speex|ilbc|h261|h263|h263p|h264), peer - audio=(gsm|alaw|ilbc)/video=(h263|h263p)/text=(nothing), combined - (gsm|alaw|ilbc|h263|h263p) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.70.9:57598 Peer video RTP is at port 192.168.70.9:55192 Peer doesn't provide T.140 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.70.9:41692 Transmitting (no NAT) to 192.168.70.9:41692: ACK sip:409@192.168.70.9:41692;rinstance=fa41035e9e5da8ea SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK76811942 Max-Forwards: 70 From: "PM10 Soft-Phone" ;tag=as2db312a1 To: ;tag=a3dc8a32 Contact: Call-ID: 586148e5479a959b1042d76e3ed77c7c@192.168.70.51:5060 CSeq: 105 ACK User-Agent: Asterisk PBX SVN--r Content-Length: 0 --- <--- SIP read from UDP:192.168.70.9:41692 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK1367d200 Contact: To: ;tag=a3dc8a32 From: "PM10 Soft-Phone";tag=as2db312a1 Call-ID: 586148e5479a959b1042d76e3ed77c7c@192.168.70.51:5060 CSeq: 105 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces User-Agent: X-Lite 4 release 4.0 stamp 58832 Content-Length: 681 v=0 o=- 12942239641798258 4 IN IP4 192.168.70.9 s=CounterPath X-Lite 4.0 c=IN IP4 192.168.70.9 t=0 0 a=ice-ufrag:8043e5 a=ice-pwd:5f2af35950be008cce660153e07db8df m=audio 57598 RTP/AVP 8 97 3 101 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=candidate:1 1 UDP 659136 192.168.70.9 57598 typ host a=candidate:1 2 UDP 659134 192.168.70.9 57599 typ host m=video 55192 RTP/AVP 98 34 a=rtpmap:98 H263-1998/90000 a=fmtp:98 QCIF=1;CIF=1;VGA=1;I=1;J=1;T=1 a=rtpmap:34 H263/90000 a=fmtp:34 QCIF=1;CIF=1;VGA=1 a=sendrecv a=candidate:1 1 UDP 659136 192.168.70.9 55192 typ host a=candidate:1 2 UDP 659134 192.168.70.9 55193 typ host <-------------> --- (12 headers 22 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.70.9:41692 Transmitting (no NAT) to 192.168.70.9:41692: ACK sip:409@192.168.70.9:41692;rinstance=fa41035e9e5da8ea SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK21b5fe90 Max-Forwards: 70 From: "PM10 Soft-Phone" ;tag=as2db312a1 To: ;tag=a3dc8a32 Contact: Call-ID: 586148e5479a959b1042d76e3ed77c7c@192.168.70.51:5060 CSeq: 105 ACK User-Agent: Asterisk PBX SVN--r Content-Length: 0 <--- SIP read from UDP:192.168.70.10:2488 ---> BYE sip:409@192.168.70.51:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.70.10:2488;branch=z9hG4bK-d8754z-1f3dbc588a741f3c-1---d8754z-;rport Max-Forwards: 70 Contact: To: ;tag=as66ed85cf From: "XLite_PM10";tag=ea0a4c5e Call-ID: MzQwNTFlYjczOTgzMDlhZDc2ZWFlNTcyZjNhNmY1ODM. CSeq: 3 BYE User-Agent: X-Lite 4 release 4.0 stamp 58832 Authorization: Digest username="410",realm="phone.lab403.neis.khabarovsk.su",nonce="71b609bf",uri="sip:409@192.168.70.51:5060",response="277d83f9a44e262ec14373b64a6815bb",algorithm=MD5 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Sending to 192.168.70.10:2488 (no NAT) Scheduling destruction of SIP dialog 'MzQwNTFlYjczOTgzMDlhZDc2ZWFlNTcyZjNhNmY1ODM.' in 6400 ms (Method: BYE) <--- Transmitting (no NAT) to 192.168.70.10:2488 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.70.10:2488;branch=z9hG4bK-d8754z-1f3dbc588a741f3c-1---d8754z-;received=192.168.70.10;rport=2488 From: "XLite_PM10";tag=ea0a4c5e To: ;tag=as66ed85cf Call-ID: MzQwNTFlYjczOTgzMDlhZDc2ZWFlNTcyZjNhNmY1ODM. CSeq: 3 BYE Server: Asterisk PBX SVN--r Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.70.9:41692 Audio is at 5060 Video is at 192.168.70.51:5060 Adding codec 100004 (alaw) to SDP Adding codec 100011 (ilbc) to SDP Adding codec 100002 (gsm) to SDP Adding video codec 200003 (h263p) to SDP Adding video codec 200002 (h263) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.70.9:41692: INVITE sip:409@192.168.70.9:41692;rinstance=fa41035e9e5da8ea SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK77186844 Max-Forwards: 70 From: "PM10 Soft-Phone" ;tag=as2db312a1 To: ;tag=a3dc8a32 Contact: Call-ID: 586148e5479a959b1042d76e3ed77c7c@192.168.70.51:5060 CSeq: 106 INVITE User-Agent: Asterisk PBX SVN--r Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Remote-Party-ID: "PM10 Soft-Phone" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 408 v=0 o=root 307637014 307637018 IN IP4 192.168.70.51 s=Asterisk PBX SVN--r c=IN IP4 192.168.70.51 b=CT:8192 t=0 0 m=audio 18358 RTP/AVP 8 97 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv m=video 12910 RTP/AVP 98 34 a=rtpmap:98 h263-1998/90000 a=rtpmap:34 H263/90000 a=sendrecv --- Scheduling destruction of SIP dialog '586148e5479a959b1042d76e3ed77c7c@192.168.70.51:5060' in 6400 ms (Method: INVITE) == Spawn extension (phones, 409, 50005) exited non-zero on 'SIP/410-00000000' Retransmitting #1 (no NAT) to 192.168.70.9:41692: INVITE sip:409@192.168.70.9:41692;rinstance=fa41035e9e5da8ea SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK77186844 Max-Forwards: 70 From: "PM10 Soft-Phone" ;tag=as2db312a1 To: ;tag=a3dc8a32 Contact: Call-ID: 586148e5479a959b1042d76e3ed77c7c@192.168.70.51:5060 CSeq: 106 INVITE User-Agent: Asterisk PBX SVN--r Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Remote-Party-ID: "PM10 Soft-Phone" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 408 v=0 o=root 307637014 307637018 IN IP4 192.168.70.51 s=Asterisk PBX SVN--r c=IN IP4 192.168.70.51 b=CT:8192 t=0 0 m=audio 18358 RTP/AVP 8 97 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv m=video 12910 RTP/AVP 98 34 a=rtpmap:98 h263-1998/90000 a=rtpmap:34 H263/90000 a=sendrecv --- <--- SIP read from UDP:192.168.70.9:41692 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK77186844 Contact: To: ;tag=a3dc8a32 From: "PM10 Soft-Phone";tag=as2db312a1 Call-ID: 586148e5479a959b1042d76e3ed77c7c@192.168.70.51:5060 CSeq: 106 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces User-Agent: X-Lite 4 release 4.0 stamp 58832 Content-Length: 681 v=0 o=- 12942239641798258 5 IN IP4 192.168.70.9 s=CounterPath X-Lite 4.0 c=IN IP4 192.168.70.9 t=0 0 a=ice-ufrag:8043e5 a=ice-pwd:5f2af35950be008cce660153e07db8df m=audio 57598 RTP/AVP 8 97 3 101 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=candidate:1 1 UDP 659136 192.168.70.9 57598 typ host a=candidate:1 2 UDP 659134 192.168.70.9 57599 typ host m=video 55192 RTP/AVP 98 34 a=rtpmap:98 H263-1998/90000 a=fmtp:98 QCIF=1;CIF=1;VGA=1;I=1;J=1;T=1 a=rtpmap:34 H263/90000 a=fmtp:34 QCIF=1;CIF=1;VGA=1 a=sendrecv a=candidate:1 1 UDP 659136 192.168.70.9 55192 typ host a=candidate:1 2 UDP 659134 192.168.70.9 55193 typ host <-------------> --- (12 headers 22 lines) --- Found RTP audio format 8 Found RTP audio format 97 Found RTP audio format 3 Found RTP audio format 101 Found audio description format iLBC for ID 97 Found audio description format telephone-event for ID 101 Found RTP video format 98 Found RTP video format 34 Found video description format H263-1998 for ID 98 Found video description format H263 for ID 34 Capabilities: us - (g723|gsm|alaw|lpc10|g729|speex|ilbc|h261|h263|h263p|h264), peer - audio=(gsm|alaw|ilbc)/video=(h263|h263p)/text=(nothing), combined - (gsm|alaw|ilbc|h263|h263p) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.70.9:57598 Peer video RTP is at port 192.168.70.9:55192 Peer doesn't provide T.140 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.70.9:41692 Transmitting (no NAT) to 192.168.70.9:41692: ACK sip:409@192.168.70.9:41692;rinstance=fa41035e9e5da8ea SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK1b20455f Max-Forwards: 70 From: "PM10 Soft-Phone" ;tag=as2db312a1 To: ;tag=a3dc8a32 Contact: Call-ID: 586148e5479a959b1042d76e3ed77c7c@192.168.70.51:5060 CSeq: 106 ACK User-Agent: Asterisk PBX SVN--r Content-Length: 0 --- set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.70.9:41692 Reliably Transmitting (no NAT) to 192.168.70.9:41692: BYE sip:409@192.168.70.9:41692;rinstance=fa41035e9e5da8ea SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK3d18f377 Max-Forwards: 70 From: "PM10 Soft-Phone" ;tag=as2db312a1 To: ;tag=a3dc8a32 Call-ID: 586148e5479a959b1042d76e3ed77c7c@192.168.70.51:5060 CSeq: 107 BYE User-Agent: Asterisk PBX SVN--r X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- Scheduling destruction of SIP dialog '586148e5479a959b1042d76e3ed77c7c@192.168.70.51:5060' in 6400 ms (Method: INVITE) <--- SIP read from UDP:192.168.70.9:41692 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK3d18f377 Contact: To: ;tag=a3dc8a32 From: "PM10 Soft-Phone";tag=as2db312a1 Call-ID: 586148e5479a959b1042d76e3ed77c7c@192.168.70.51:5060 CSeq: 107 BYE User-Agent: X-Lite 4 release 4.0 stamp 58832 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '586148e5479a959b1042d76e3ed77c7c@192.168.70.51:5060' Method: INVITE --- (10 headers 0 lines) --- Really destroying SIP dialog '113fa56a3b04a43a6321c086748a3de1@192.168.70.51:5060' Method: OPTIONS Reliably Transmitting (no NAT) to 192.168.70.10:2488: OPTIONS sip:410@192.168.70.10:2488;rinstance=ca5ae0feb00ce545 SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK06be11e6 Max-Forwards: 70 From: "asterisk" ;tag=as16802a7b To: Contact: Call-ID: 5d4a45534cd7ee04662fda1d3a3588d8@192.168.70.51:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN--r Date: Tue, 15 Feb 2011 10:34:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:192.168.70.10:2488 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK06be11e6 Contact: To: ;tag=9945ac09 From: "asterisk";tag=as16802a7b Call-ID: 5d4a45534cd7ee04662fda1d3a3588d8@192.168.70.51:5060 CSeq: 102 OPTIONS Accept: application/sdp Accept-Language: en Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Supported: replaces User-Agent: X-Lite 4 release 4.0 stamp 58832 Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Really destroying SIP dialog '5d4a45534cd7ee04662fda1d3a3588d8@192.168.70.51:5060' Method: OPTIONS Really destroying SIP dialog 'D1B9-57D9-4745997011F3CD79A7E1-081@SipHost' Method: REGISTER Really destroying SIP dialog 'D1B9-57D9-4745997118928F34D32B-082@SipHost' Method: REGISTER Really destroying SIP dialog 'D1B9-57D9-4745997118928F34D32B-083@SipHost' Method: REGISTER Really destroying SIP dialog 'D1B9-57D9-4745997218928F34D32B-084@SipHost' Method: REGISTER Really destroying SIP dialog 'D1B9-57D9-4745997218928F34D32B-085@SipHost' Method: REGISTER Really destroying SIP dialog 'D1B9-57D9-47459972E1C3E56C1892-086@SipHost' Method: REGISTER Reliably Transmitting (no NAT) to 192.168.70.9:41692: OPTIONS sip:409@192.168.70.9:41692;rinstance=fa41035e9e5da8ea SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK668ef9ac Max-Forwards: 70 From: "asterisk" ;tag=as4a29462b To: Contact: Call-ID: 34ef521302c2c03c28455614245a58ec@192.168.70.51:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN--r Date: Tue, 15 Feb 2011 10:35:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:192.168.70.9:41692 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK668ef9ac Contact: To: ;tag=351a7d1d From: "asterisk";tag=as4a29462b Call-ID: 34ef521302c2c03c28455614245a58ec@192.168.70.51:5060 CSeq: 102 OPTIONS Accept: application/sdp Accept-Language: en Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Supported: replaces User-Agent: X-Lite 4 release 4.0 stamp 58832 Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Really destroying SIP dialog '34ef521302c2c03c28455614245a58ec@192.168.70.51:5060' Method: OPTIONS