*CLI> sip set debug peer 409 SIP Debugging Enabled for IP: 192.168.70.9 *CLI> == Using UDPTL TOS bits 184 == Using UDPTL CoS mark 5 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [409@phones:1] Gosub("SIP/410-00000002", "409,stdexten(SIP/409)") in new stack -- Executing [409@phones:50000] NoOp("SIP/410-00000002", "Start stdexten") in new stack -- Executing [409@phones:50001] Set("SIP/410-00000002", "LOCAL(ext)=409") in new stack -- Executing [409@phones:50002] Set("SIP/410-00000002", "LOCAL(dev)=SIP/409") in new stack -- Executing [409@phones:50003] Set("SIP/410-00000002", "LOCAL(cntx)=") in new stack -- Executing [409@phones:50004] Set("SIP/410-00000002", "LOCAL(mbx)="409"""") in new stack -- Executing [409@phones:50005] Dial("SIP/410-00000002", "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 0x8 (alaw) to SDP Adding codec 0x800 (g726) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x80 (lpc10) to SDP Adding video codec 0x80000 (h263) to SDP Adding video codec 0x100000 (h263p) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.70.9:53110: INVITE sip:409@192.168.70.9:53110;rinstance=8010d2be72d8040b SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK413be7eb Max-Forwards: 70 From: "XLite_PM10" ;tag=as7611b2f1 To: Contact: Call-ID: 7f8bf597504da84f037f36bb4cb22f43@192.168.70.51:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX SVN--r Date: Thu, 28 Apr 2011 09:23:16 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Remote-Party-ID: "XLite_PM10" ;party=calling;privacy=off; screen=no Content-Type: application/sdp Content-Length: 417 v=0 o=root 700541228 700541228 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 19576 RTP/AVP 8 111 3 7 101 a=rtpmap:8 PCMA/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:7 LPC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv m=video 11882 RTP/AVP 34 98 a=rtpmap:34 H263/90000 a=rtpmap:98 h263-1998/90000 a=sendrecv --- -- Called SIP/409 <--- SIP read from UDP:192.168.70.9:53110 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK413be7eb Contact: To: ;tag=9e3937db From: "XLite_PM10";tag=as7611b2f1 Call-ID: 7f8bf597504da84f037f36bb4cb22f43@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/409-00000003 is ringing <--- SIP read from UDP:192.168.70.9:53110 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK413be7eb Contact: To: ;tag=9e3937db From: "XLite_PM10";tag=as7611b2f1 Call-ID: 7f8bf597504da84f037f36bb4cb22f43@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: 655 v=0 o=- 12948456200701673 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:9f9216 a=ice-pwd:3bf8e1baef750f87bfb277cfcb3ecc4f m=audio 61648 RTP/AVP 8 3 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=candidate:1 1 UDP 659136 192.168.70.9 61648 typ host a=candidate:1 2 UDP 659134 192.168.70.9 61649 typ host m=video 54488 RTP/AVP 34 98 a=rtpmap:34 H263/90000 a=fmtp:34 QCIF=1;CIF=1;VGA=1 a=rtpmap:98 H263-1998/90000 a=fmtp:98 QCIF=1;CIF=1;VGA=1;I=1;J=1;T=1 a=sendrecv a=candidate:1 1 UDP 659136 192.168.70.9 54488 typ host a=candidate:1 2 UDP 659134 192.168.70.9 54489 typ host <-------------> --- (12 headers 21 lines) --- Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 101 Found audio description format telephone-event for ID 101 Found RTP video format 34 Found RTP video format 98 Found video description format H263 for ID 34 Found video description format H263-1998 for ID 98 Capabilities: us - 0x3c0f8b (g723|gsm|alaw|g726|lpc10|g729|speex|ilbc|h261|h263|h263p|h264), peer - audio=0xa (gsm|alaw)/video=0x180000 (h263|h263p)/text=0x0 (nothing), combined - 0x18000a (gsm|alaw|h263|h263p) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.70.9:61648 Peer video RTP is at port 192.168.70.9:54488 Peer doesn't provide T.140 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.70.9:53110 Transmitting (no NAT) to 192.168.70.9:53110: ACK sip:409@192.168.70.9:53110;rinstance=8010d2be72d8040b SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK55f2157e Max-Forwards: 70 From: "XLite_PM10" ;tag=as7611b2f1 To: ;tag=9e3937db Contact: Call-ID: 7f8bf597504da84f037f36bb4cb22f43@192.168.70.51:5060 CSeq: 102 ACK User-Agent: Asterisk PBX SVN--r Content-Length: 0 --- -- SIP/409-00000003 answered SIP/410-00000002 -- Remotely bridging SIP/410-00000002 and SIP/409-00000003 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.70.9:53110 Audio is at 5060 Video is at 192.168.70.51:5060 Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding video codec 0x80000 (h263) to SDP Adding video codec 0x100000 (h263p) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.70.9:53110: INVITE sip:409@192.168.70.9:53110;rinstance=8010d2be72d8040b SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK07b0b175 Max-Forwards: 70 From: "XLite_PM10" ;tag=as7611b2f1 To: ;tag=9e3937db Contact: Call-ID: 7f8bf597504da84f037f36bb4cb22f43@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: "XLite_PM10" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 363 v=0 o=root 700541228 700541229 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 55948 RTP/AVP 8 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv m=video 53684 RTP/AVP 34 98 a=rtpmap:34 H263/90000 a=rtpmap:98 h263-1998/90000 a=sendrecv --- <--- SIP read from UDP:192.168.70.9:53110 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK07b0b175 Contact: To: ;tag=9e3937db From: "XLite_PM10";tag=as7611b2f1 Call-ID: 7f8bf597504da84f037f36bb4cb22f43@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: 655 v=0 o=- 12948456200701673 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:9f9216 a=ice-pwd:3bf8e1baef750f87bfb277cfcb3ecc4f m=audio 61648 RTP/AVP 8 3 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=candidate:1 1 UDP 659136 192.168.70.9 61648 typ host a=candidate:1 2 UDP 659134 192.168.70.9 61649 typ host m=video 54488 RTP/AVP 34 98 a=rtpmap:34 H263/90000 a=fmtp:34 QCIF=1;CIF=1;VGA=1 a=rtpmap:98 H263-1998/90000 a=fmtp:98 QCIF=1;CIF=1;VGA=1;I=1;J=1;T=1 a=sendrecv a=candidate:1 1 UDP 659136 192.168.70.9 54488 typ host a=candidate:1 2 UDP 659134 192.168.70.9 54489 typ host <-------------> --- (12 headers 21 lines) --- Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 101 Found audio description format telephone-event for ID 101 Found RTP video format 34 Found RTP video format 98 Found video description format H263 for ID 34 Found video description format H263-1998 for ID 98 Capabilities: us - 0x3c0f8b (g723|gsm|alaw|g726|lpc10|g729|speex|ilbc|h261|h263|h263p|h264), peer - audio=0xa (gsm|alaw)/video=0x180000 (h263|h263p)/text=0x0 (nothing), combined - 0x18000a (gsm|alaw|h263|h263p) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.70.9:61648 Peer video RTP is at port 192.168.70.9:54488 Peer doesn't provide T.140 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.70.9:53110 Transmitting (no NAT) to 192.168.70.9:53110: ACK sip:409@192.168.70.9:53110;rinstance=8010d2be72d8040b SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK0508053f Max-Forwards: 70 From: "XLite_PM10" ;tag=as7611b2f1 To: ;tag=9e3937db Contact: Call-ID: 7f8bf597504da84f037f36bb4cb22f43@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 192.168.70.9:53110 Audio is at 5060 Video is at 192.168.70.51:5060 Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding video codec 0x80000 (h263) to SDP Adding video codec 0x100000 (h263p) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.70.9:53110: INVITE sip:409@192.168.70.9:53110;rinstance=8010d2be72d8040b SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK0cb80c89 Max-Forwards: 70 From: "XLite_PM10" ;tag=as7611b2f1 To: ;tag=9e3937db Contact: Call-ID: 7f8bf597504da84f037f36bb4cb22f43@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: "XLite_PM10" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 363 v=0 o=root 700541228 700541230 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 55948 RTP/AVP 8 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv m=video 53684 RTP/AVP 34 98 a=rtpmap:34 H263/90000 a=rtpmap:98 h263-1998/90000 a=sendrecv --- <--- SIP read from UDP:192.168.70.9:53110 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK0cb80c89 Contact: To: ;tag=9e3937db From: "XLite_PM10";tag=as7611b2f1 Call-ID: 7f8bf597504da84f037f36bb4cb22f43@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: 655 v=0 o=- 12948456200701673 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:9f9216 a=ice-pwd:3bf8e1baef750f87bfb277cfcb3ecc4f m=audio 61648 RTP/AVP 8 3 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=candidate:1 1 UDP 659136 192.168.70.9 61648 typ host a=candidate:1 2 UDP 659134 192.168.70.9 61649 typ host m=video 54488 RTP/AVP 34 98 a=rtpmap:34 H263/90000 a=fmtp:34 QCIF=1;CIF=1;VGA=1 a=rtpmap:98 H263-1998/90000 a=fmtp:98 QCIF=1;CIF=1;VGA=1;I=1;J=1;T=1 a=sendrecv a=candidate:1 1 UDP 659136 192.168.70.9 54488 typ host a=candidate:1 2 UDP 659134 192.168.70.9 54489 typ host <-------------> --- (12 headers 21 lines) --- Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 101 Found audio description format telephone-event for ID 101 Found RTP video format 34 Found RTP video format 98 Found video description format H263 for ID 34 Found video description format H263-1998 for ID 98 Capabilities: us - 0x3c0f8b (g723|gsm|alaw|g726|lpc10|g729|speex|ilbc|h261|h263|h263p|h264), peer - audio=0xa (gsm|alaw)/video=0x180000 (h263|h263p)/text=0x0 (nothing), combined - 0x18000a (gsm|alaw|h263|h263p) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.70.9:61648 Peer video RTP is at port 192.168.70.9:54488 Peer doesn't provide T.140 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.70.9:53110 Transmitting (no NAT) to 192.168.70.9:53110: ACK sip:409@192.168.70.9:53110;rinstance=8010d2be72d8040b SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK1a9bceca Max-Forwards: 70 From: "XLite_PM10" ;tag=as7611b2f1 To: ;tag=9e3937db Contact: Call-ID: 7f8bf597504da84f037f36bb4cb22f43@192.168.70.51:5060 CSeq: 104 ACK User-Agent: Asterisk PBX SVN--r Content-Length: 0 --- set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.70.9:53110 Audio is at 5060 Video is at 192.168.70.51:5060 Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding video codec 0x80000 (h263) to SDP Adding video codec 0x100000 (h263p) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.70.9:53110: INVITE sip:409@192.168.70.9:53110;rinstance=8010d2be72d8040b SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK1e62714a Max-Forwards: 70 From: "XLite_PM10" ;tag=as7611b2f1 To: ;tag=9e3937db Contact: Call-ID: 7f8bf597504da84f037f36bb4cb22f43@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: "XLite_PM10" ;party=calling;privacy=off; screen=no Content-Type: application/sdp Content-Length: 363 v=0 o=root 700541228 700541231 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 19576 RTP/AVP 8 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv m=video 11882 RTP/AVP 34 98 a=rtpmap:34 H263/90000 a=rtpmap:98 h263-1998/90000 a=sendrecv --- Scheduling destruction of SIP dialog '7f8bf597504da84f037f36bb4cb22f43@192.168.70.51:5060' in 32000 ms (Method: INVITE) == Spawn extension (phones, 409, 50005) exited non-zero on 'SIP/410-00000002' <--- SIP read from UDP:192.168.70.9:53110 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK1e62714a Contact: To: ;tag=9e3937db From: "XLite_PM10";tag=as7611b2f1 Call-ID: 7f8bf597504da84f037f36bb4cb22f43@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: 655 v=0 o=- 12948456200701673 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:9f9216 a=ice-pwd:3bf8e1baef750f87bfb277cfcb3ecc4f m=audio 61648 RTP/AVP 8 3 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=candidate:1 1 UDP 659136 192.168.70.9 61648 typ host a=candidate:1 2 UDP 659134 192.168.70.9 61649 typ host m=video 54488 RTP/AVP 34 98 a=rtpmap:34 H263/90000 a=fmtp:34 QCIF=1;CIF=1;VGA=1 a=rtpmap:98 H263-1998/90000 a=fmtp:98 QCIF=1;CIF=1;VGA=1;I=1;J=1;T=1 a=sendrecv a=candidate:1 1 UDP 659136 192.168.70.9 54488 typ host a=candidate:1 2 UDP 659134 192.168.70.9 54489 typ host <-------------> --- (12 headers 21 lines) --- Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 101 Found audio description format telephone-event for ID 101 Found RTP video format 34 Found RTP video format 98 Found video description format H263 for ID 34 Found video description format H263-1998 for ID 98 Capabilities: us - 0x3c0f8b (g723|gsm|alaw|g726|lpc10|g729|speex|ilbc|h261|h263|h263p|h264), peer - audio=0xa (gsm|alaw)/video=0x180000 (h263|h263p)/text=0x0 (nothing), combined - 0x18000a (gsm|alaw|h263|h263p) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.70.9:61648 Peer video RTP is at port 192.168.70.9:54488 Peer doesn't provide T.140 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.70.9:53110 Transmitting (no NAT) to 192.168.70.9:53110: ACK sip:409@192.168.70.9:53110;rinstance=8010d2be72d8040b SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK445df3dc Max-Forwards: 70 From: "XLite_PM10" ;tag=as7611b2f1 To: ;tag=9e3937db Contact: Call-ID: 7f8bf597504da84f037f36bb4cb22f43@192.168.70.51:5060 CSeq: 105 ACK User-Agent: Asterisk PBX SVN--r Content-Length: 0 --- set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.70.9:53110 Reliably Transmitting (no NAT) to 192.168.70.9:53110: BYE sip:409@192.168.70.9:53110;rinstance=8010d2be72d8040b SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK1e7bf9f2 Max-Forwards: 70 From: "XLite_PM10" ;tag=as7611b2f1 To: ;tag=9e3937db Call-ID: 7f8bf597504da84f037f36bb4cb22f43@192.168.70.51:5060 CSeq: 106 BYE User-Agent: Asterisk PBX SVN--r X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- Scheduling destruction of SIP dialog '7f8bf597504da84f037f36bb4cb22f43@192.168.70.51:5060' in 32000 ms (Method: INVITE) <--- SIP read from UDP:192.168.70.9:53110 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK1e7bf9f2 Contact: To: ;tag=9e3937db From: "XLite_PM10";tag=as7611b2f1 Call-ID: 7f8bf597504da84f037f36bb4cb22f43@192.168.70.51:5060 CSeq: 106 BYE User-Agent: X-Lite 4 release 4.0 stamp 58832 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '7f8bf597504da84f037f36bb4cb22f43@192.168.70.51:5060' Method: INVITE <--- SIP read from UDP:192.168.70.9:53110 ---> SUBSCRIBE sip:asterisk@192.168.70.51:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.70.9:53110;branch=z9hG4bK-d8754z-25bdf34677472c31-1---d8754z-;rport Max-Forwards: 70 Contact: To: "XLite_PM9";tag=as056c909a From: "XLite_PM9";tag=9d6dcfff Call-ID: YWY3MDdmYTk5OTA5N2I0NzYyYzM5ZmJiMjljZmI4OTM. CSeq: 5 SUBSCRIBE Expires: 0 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",non ce="5592f4d2",uri="sip:asterisk@192.168.70.51:5060",response="a8194732a7c531980d 9700e53a9f32f2",algorithm=MD5 Event: message-summary Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Found peer '409' for '409' from 192.168.70.9:53110 [Apr 28 20:24:33] NOTICE[26054]: chan_sip.c:13764 check_auth: Correct auth, but based on stale nonce received from '"XLite_PM9";tag=9d6dcfff' <--- Transmitting (no NAT) to 192.168.70.9:53110 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.70.9:53110;branch=z9hG4bK-d8754z-25bdf34677472c31-1---d8754z-;received=192.168.70.9;rport=53110 From: "XLite_PM9";tag=9d6dcfff To: "XLite_PM9";tag=as056c909a Call-ID: YWY3MDdmYTk5OTA5N2I0NzYyYzM5ZmJiMjljZmI4OTM. CSeq: 5 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="29caf32a", stale=true Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'YWY3MDdmYTk5OTA5N2I0NzYyYzM5ZmJiMjljZmI4OTM.' in 32000 ms (Method: SUBSCRIBE) <--- SIP read from UDP:192.168.70.9:53110 ---> REGISTER sip:asterisk.phone.lab403.neis.khabarovsk.su SIP/2.0 Via: SIP/2.0/UDP 192.168.70.9:53110;branch=z9hG4bK-d8754z-9c73ca963cd7a9ea-1---d8754z-;rport Max-Forwards: 70 Contact: ;expires=0 To: "XLite_PM9" From: "XLite_PM9";tag=9075c239 Call-ID: Y2JmNWZiNDkzNTQ0N2U5MzZhODA2N2Y4MTM3ZDFiYzU. CSeq: 3 REGISTER 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="556752f9",uri="sip:asterisk.phone.lab403.neis.khabarovsk.su",response="e03fcf64241149fd19ece7953715c2d6",algorithm=MD5 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Sending to 192.168.70.9:53110 (no NAT) <--- Transmitting (no NAT) to 192.168.70.9:53110 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.70.9:53110;branch=z9hG4bK-d8754z-9c73ca963cd7a9ea-1---d8754z-;received=192.168.70.9;rport=53110 From: "XLite_PM9";tag=9075c239 To: "XLite_PM9";tag=as0009ae93 Call-ID: Y2JmNWZiNDkzNTQ0N2U5MzZhODA2N2Y4MTM3ZDFiYzU. CSeq: 3 REGISTER 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="482f92b4" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'Y2JmNWZiNDkzNTQ0N2U5MzZhODA2N2Y4MTM3ZDFiYzU.' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:192.168.70.9:53110 ---> SUBSCRIBE sip:asterisk@192.168.70.51:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.70.9:53110;branch=z9hG4bK-d8754z-68301f911c55ce78-1---d8754z-;rport Max-Forwards: 70 Contact: To: "XLite_PM9";tag=as056c909a From: "XLite_PM9";tag=9d6dcfff Call-ID: YWY3MDdmYTk5OTA5N2I0NzYyYzM5ZmJiMjljZmI4OTM. CSeq: 6 SUBSCRIBE Expires: 0 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="29caf32a",uri="sip:asterisk@192.168.70.51:5060",response="1f3dbbe2291bf90476f7a813e3da7866",algorithm=MD5 Event: message-summary Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Found peer '409' for '409' from 192.168.70.9:53110 <--- Transmitting (no NAT) to 192.168.70.9:53110 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.70.9:53110;branch=z9hG4bK-d8754z-68301f911c55ce78-1---d8754z-;received=192.168.70.9;rport=53110 From: "XLite_PM9";tag=9d6dcfff To: "XLite_PM9";tag=as056c909a Call-ID: YWY3MDdmYTk5OTA5N2I0NzYyYzM5ZmJiMjljZmI4OTM. CSeq: 6 SUBSCRIBE Server: Asterisk PBX SVN--r Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 0 Content-Length: 0 <------------> Reliably Transmitting (no NAT) to 192.168.70.9:53110: NOTIFY sip:409@192.168.70.9:53110 SIP/2.0 Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK203dba0f Max-Forwards: 70 Route: From: "asterisk" ;tag=as056c909a To: ;tag=9d6dcfff Contact: Call-ID: YWY3MDdmYTk5OTA5N2I0NzYyYzM5ZmJiMjljZmI4OTM. CSeq: 104 NOTIFY User-Agent: Asterisk PBX SVN--r Event: message-summary Content-Type: application/simple-message-summary Subscription-State: terminated;reason=timeout 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:192.168.70.9:53110 ---> REGISTER sip:asterisk.phone.lab403.neis.khabarovsk.su SIP/2.0 Via: SIP/2.0/UDP 192.168.70.9:53110;branch=z9hG4bK-d8754z-bedd80e21958b9b1-1---d8754z-;rport Max-Forwards: 70 Contact: ;expires=0 To: "XLite_PM9" From: "XLite_PM9";tag=9075c239 Call-ID: Y2JmNWZiNDkzNTQ0N2U5MzZhODA2N2Y4MTM3ZDFiYzU. CSeq: 4 REGISTER 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="482f92b4",uri="sip:asterisk.phone.lab403.neis.khabarovsk.su",response="003e1e7ca1160d22f242bc202b8d3ac1",algorithm=MD5 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Sending to 192.168.70.9:53110 (no NAT) -- Unregistered SIP '409' <--- Transmitting (no NAT) to 192.168.70.9:53110 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.70.9:53110;branch=z9hG4bK-d8754z-bedd80e21958b9b1-1---d 8754z-;received=192.168.70.9;rport=53110 From: "XLite_PM9";tag=9075c239 To: "XLite_PM9";tag=as0009ae93 Call-ID: Y2JmNWZiNDkzNTQ0N2U5MzZhODA2N2Y4MTM3ZDFiYzU. CSeq: 4 REGISTER Server: Asterisk PBX SVN--r Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 0 Date: Thu, 28 Apr 2011 09:24:33 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'Y2JmNWZiNDkzNTQ0N2U5MzZhODA2N2Y4MTM3ZDFiYzU.' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:192.168.70.9:53110 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.70.51:5060;branch=z9hG4bK203dba0f Contact: To: ;tag=9d6dcfff From: "asterisk";tag=as056c909a Call-ID: YWY3MDdmYTk5OTA5N2I0NzYyYzM5ZmJiMjljZmI4OTM. CSeq: 104 NOTIFY User-Agent: X-Lite 4 release 4.0 stamp 58832 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog 'YWY3MDdmYTk5OTA5N2I0NzYyYzM5ZmJiMjljZmI4OTM.' Method: SUBSCRIBE