verbose 3 trace -- Executing [013112573@PBX:1] Goto("SIP/ip-in-0a5eee50", "004313112573|1") in new stack -- Goto (PBX,004313112573,1) ....do some numbermanipulation.... -- Goto (PBX,004313112573,22) -- Executing [004313112573@PBX:22] Dial("SIP/ip-in-0a5eee50", "SIP/4313112573@ip-out:5060") in new stack [Mar 25 15:37:04] NOTICE[1444]: chan_sip.c:6678 add_sdp: add audio = 1 add t38 = 0 -- Called 4313112573@ip-out:5060 -- SIP/ip-out:5060-0a5f3230 is ringing -- SIP/ip-out:5060-0a5f3230 answered SIP/ip-in-0a5eee50 [Mar 25 15:37:08] NOTICE[1444]: chan_sip.c:6678 add_sdp: add audio = 1 add t38 = 1 -- Native bridging SIP/ip-in-0a5eee50 and SIP/ip-out:5060-0a5f3230 [Mar 25 15:37:08] NOTICE[1444]: chan_sip.c:6678 add_sdp: add audio = 1 add t38 = 0 [Mar 25 15:37:08] NOTICE[1421]: chan_sip.c:6678 add_sdp: add audio = 1 add t38 = 0 [Mar 25 15:37:08] WARNING[1421]: chan_sip.c:12569 handle_response_invite: RTP re-invite after T38 session not handled yet ! [Mar 25 15:37:40] WARNING[1421]: chan_sip.c:2131 __sip_autodestruct: Autodestruct on dialog '2cf6ffdd63b70b40308d84f75506b793@ip-asterisk' with owner in place (Method: INVITE) [Mar 25 15:37:40] NOTICE[1444]: chan_sip.c:6678 add_sdp: add audio = 1 add t38 = 0 [Mar 25 15:37:40] NOTICE[1444]: chan_sip.c:6678 add_sdp: add audio = 1 add t38 = 0 == Spawn extension (PBX, 004313112573, 22) exited non-zero on 'SIP/ip-in-0a5eee50' SIP Trace calling side <--- SIP read from ip-in:5060 ---> INVITE sip:013112573@ip-asterisk SIP/2.0 Max-Forwards: 50 Via: SIP/2.0/UDP ip-in:5060;rport;branch=z9hG4bK37901929691601021504735 From: ;tag=416386026902121798762838496323 To: Call-ID: 746301659585914062335152852010@ip-in CSeq: 1 INVITE Contact: User-Agent: TELES.VoIPBOX 13.0k 895 Allow: INVITE,ACK,CANCEL,BYE,UPDATE,REGISTER,PRACK,INFO,NOTIFY,REFER Timestamp: 1237992314 Content-Type: application/sdp Content-Length: 381 v=0 o=- 10 1 IN IP4 ip-in s=- c=IN IP4 ip-in t=0 0 m=audio 29000 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=ptime:20 a=sendrecv m=image 29000 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxUdpEC:t38UDPRedundancy a=T38FaxRateManagement:transferredTCF a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 <-------------> --- (13 headers 18 lines) --- Sending to ip-in : 5060 (NAT) Using INVITE request as basis request - 746301659585914062335152852010@ip-in Found peer '4313112574' Found RTP audio format 18 Got T.38 offer in SDP in dialog 746301659585914062335152852010@ip-in Peer audio RTP is at port ip-in:29000 Found audio description format G729 for ID 18 Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port ip-in:29000 Looking for 013112573 in PBX (domain ip-asterisk) list_route: hop: DataNode1*CLI> <--- Transmitting (no NAT) to ip-in:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP ip-in:5060;branch=z9hG4bK37901929691601021504735;received=ip-in;rport=5060 From: ;tag=416386026902121798762838496323 To: Call-ID: 746301659585914062335152852010@ip-in CSeq: 1 INVITE User-Agent: GlobalCall+T.38 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 25 15:45:14] NOTICE[1458]: chan_sip.c:6678 add_sdp: add audio = 1 add t38 = 0 DataNode1*CLI> <--- Transmitting (no NAT) to ip-in:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP ip-in:5060;branch=z9hG4bK37901929691601021504735;received=ip-in;rport=5060 From: ;tag=416386026902121798762838496323 To: ;tag=as4259bb5a Call-ID: 746301659585914062335152852010@ip-in CSeq: 1 INVITE User-Agent: GlobalCall+T.38 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 25 15:45:18] NOTICE[1458]: chan_sip.c:6678 add_sdp: add audio = 1 add t38 = 1 Audio is at ip-asterisk port 18314 Adding codec 0x100 (g729) to SDP <--- Reliably Transmitting (no NAT) to ip-in:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP ip-in:5060;branch=z9hG4bK37901929691601021504735;received=ip-in;rport=5060 From: ;tag=416386026902121798762838496323 To: ;tag=as4259bb5a Call-ID: 746301659585914062335152852010@ip-in CSeq: 1 INVITE User-Agent: GlobalCall+T.38 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 394 v=0 o=root 1405 1405 IN IP4 ip-asterisk s=session c=IN IP4 ip-asterisk t=0 0 m=audio 18314 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - a=ptime:20 a=sendrecv m=image 12765 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:400 a=T38FaxMaxDatagram:400 a=T38FaxUdpEC:t38UDPRedundancy <------------> [Mar 25 15:45:18] NOTICE[1458]: chan_sip.c:6678 add_sdp: add audio = 1 add t38 = 0 DataNode1*CLI> <--- SIP read from ip-in:5060 ---> ACK sip:013112573@ip-asterisk SIP/2.0 Max-Forwards: 50 Via: SIP/2.0/UDP ip-in:5060;rport;branch=z9hG4bK69542171947666734707063 From: ;tag=416386026902121798762838496323 To: ;tag=as4259bb5a Contact: User-Agent: TELES.VoIPBOX 13.0k 895 Call-ID: 746301659585914062335152852010@ip-in CSeq: 1 ACK Allow: INVITE,ACK,CANCEL,BYE,UPDATE,REGISTER,PRACK,INFO,NOTIFY,REFER Content-Length: 0 <-------------> --- (11 headers 0 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to ip-in, port 5060 [Mar 25 15:45:18] NOTICE[1421]: chan_sip.c:6678 add_sdp: add audio = 1 add t38 = 0 Audio is at ip-asterisk port 18314 Adding codec 0x100 (g729) to SDP Reliably Transmitting (no NAT) to ip-in:5060: INVITE sip:33@ip-in SIP/2.0 Via: SIP/2.0/UDP ip-asterisk:5060;branch=z9hG4bK3367517f;rport From: ;tag=as4259bb5a To: ;tag=416386026902121798762838496323 Contact: Call-ID: 746301659585914062335152852010@ip-in CSeq: 102 INVITE User-Agent: GlobalCall+T.38 Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 205 v=0 o=root 1405 1406 IN IP4 ip-out s=session c=IN IP4 ip-out t=0 0 m=audio 16426 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Mar 25 15:45:18] WARNING[1421]: chan_sip.c:12569 handle_response_invite: RTP re-invite after T38 session not handled yet ! DataNode1*CLI> <--- SIP read from ip-in:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP ip-asterisk:5060;branch=z9hG4bK3367517f;rport From: ;tag=as4259bb5a To: ;tag=416386026902121798762838496323 Call-ID: 746301659585914062335152852010@ip-in CSeq: 102 INVITE Allow: INVITE,ACK,CANCEL,BYE,UPDATE,REGISTER,PRACK,INFO,NOTIFY,REFER Contact: User-Agent: TELES.VoIPBOX 13.0k 895 Content-Type: application/sdp Content-Length: 164 v=0 o=- 10 1 IN IP4 ip-in s=- c=IN IP4 ip-in t=0 0 m=audio 29000 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=ptime:20 a=sendrecv <-------------> --- (11 headers 10 lines) --- Found RTP audio format 18 Peer audio RTP is at port ip-in:29000 Found audio description format G729 for ID 18 Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port ip-in:29000 set_destination: Parsing for address/port to send to set_destination: set destination to ip-in, port 5060 Transmitting (no NAT) to ip-in:5060: ACK sip:33@ip-in SIP/2.0 Via: SIP/2.0/UDP ip-asterisk:5060;branch=z9hG4bK696223e2;rport From: ;tag=as4259bb5a To: ;tag=416386026902121798762838496323 Contact: Call-ID: 746301659585914062335152852010@ip-in CSeq: 102 ACK User-Agent: GlobalCall+T.38 Max-Forwards: 70 Content-Length: 0 --- [Mar 25 15:45:38] WARNING[1421]: res_config_mysql.c:360 update_mysql: MySQL RealTime: Failed to query database. Check debug for more info. [Mar 25 15:45:50] WARNING[1421]: chan_sip.c:2131 __sip_autodestruct: Autodestruct on dialog '6ef46e7849df834c40101e0a51c40288@ip-asterisk' with owner in place (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to ip-in, port 5060 [Mar 25 15:45:50] NOTICE[1458]: chan_sip.c:6678 add_sdp: add audio = 1 add t38 = 0 Audio is at ip-asterisk port 18314 Adding codec 0x100 (g729) to SDP Reliably Transmitting (no NAT) to ip-in:5060: INVITE sip:33@ip-in SIP/2.0 Via: SIP/2.0/UDP ip-asterisk:5060;branch=z9hG4bK043c2951;rport From: ;tag=as4259bb5a To: ;tag=416386026902121798762838496323 Contact: Call-ID: 746301659585914062335152852010@ip-in CSeq: 103 INVITE User-Agent: GlobalCall+T.38 Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 209 v=0 o=root 1405 1407 IN IP4 ip-asterisk s=session c=IN IP4 ip-asterisk t=0 0 m=audio 18314 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Mar 25 15:45:50] NOTICE[1458]: chan_sip.c:6678 add_sdp: add audio = 1 add t38 = 0 Scheduling destruction of SIP dialog '746301659585914062335152852010@ip-in' in 32000 ms (Method: ACK) DataNode1*CLI> <--- SIP read from ip-in:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP ip-asterisk:5060;branch=z9hG4bK043c2951;rport From: ;tag=as4259bb5a To: ;tag=416386026902121798762838496323 Call-ID: 746301659585914062335152852010@ip-in CSeq: 103 INVITE Allow: INVITE,ACK,CANCEL,BYE,UPDATE,REGISTER,PRACK,INFO,NOTIFY,REFER Contact: User-Agent: TELES.VoIPBOX 13.0k 895 Content-Type: application/sdp Content-Length: 164 v=0 o=- 10 1 IN IP4 ip-in s=- c=IN IP4 ip-in t=0 0 m=audio 29000 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=ptime:20 a=sendrecv <-------------> --- (11 headers 10 lines) --- Found RTP audio format 18 Peer audio RTP is at port ip-in:29000 Found audio description format G729 for ID 18 Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port ip-in:29000 set_destination: Parsing for address/port to send to set_destination: set destination to ip-in, port 5060 Transmitting (no NAT) to ip-in:5060: ACK sip:33@ip-in SIP/2.0 Via: SIP/2.0/UDP ip-asterisk:5060;branch=z9hG4bK2d564040;rport From: ;tag=as4259bb5a To: ;tag=416386026902121798762838496323 Contact: Call-ID: 746301659585914062335152852010@ip-in CSeq: 103 ACK User-Agent: GlobalCall+T.38 Max-Forwards: 70 Content-Length: 0 --- set_destination: Parsing for address/port to send to set_destination: set destination to ip-in, port 5060 Reliably Transmitting (no NAT) to ip-in:5060: BYE sip:33@ip-in SIP/2.0 Via: SIP/2.0/UDP ip-asterisk:5060;branch=z9hG4bK4ca7486b;rport From: ;tag=as4259bb5a To: ;tag=416386026902121798762838496323 Call-ID: 746301659585914062335152852010@ip-in CSeq: 104 BYE User-Agent: GlobalCall+T.38 Max-Forwards: 70 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- Scheduling destruction of SIP dialog '746301659585914062335152852010@ip-in' in 32000 ms (Method: ACK) DataNode1*CLI> <--- SIP read from ip-in:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP ip-asterisk:5060;branch=z9hG4bK4ca7486b;rport From: ;tag=as4259bb5a To: ;tag=416386026902121798762838496323 Call-ID: 746301659585914062335152852010@ip-in CSeq: 104 BYE Allow: INVITE,ACK,CANCEL,BYE,UPDATE,REGISTER,PRACK,INFO,NOTIFY,REFER Contact: User-Agent: TELES.VoIPBOX 13.0k 895 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '746301659585914062335152852010@ip-in' Method: ACK DataNode1*CLI> SIP Trace called side Really destroying SIP dialog 'c83c630f-9b4a12e5@92.61.51.210' Method: REGISTER [Mar 25 15:55:57] NOTICE[1480]: chan_sip.c:6678 add_sdp: add audio = 1 add t38 = 0 Audio is at 79.170.208.149 port 17202 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 92.61.51.210:5060: INVITE sip:4313112573@92.61.51.210 SIP/2.0 Via: SIP/2.0/UDP 79.170.208.149:5060;branch=z9hG4bK3ef669fe;rport From: "431311257433" ;tag=as637892c8 To: Contact: Call-ID: 5a9564f2289bd1d206d0207b4f709495@79.170.208.149 CSeq: 102 INVITE User-Agent: GlobalCall+T.38 Max-Forwards: 70 Remote-Party-ID: "431311257433" ;privacy=off;screen=yes Date: Wed, 25 Mar 2009 14:55:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 265 v=0 o=root 1405 1405 IN IP4 79.170.208.149 s=session c=IN IP4 79.170.208.149 t=0 0 m=audio 17202 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- DataNode1*CLI> <--- SIP read from 92.61.51.210:5060 ---> SIP/2.0 100 Trying To: From: "431311257433" ;tag=as637892c8 Call-ID: 5a9564f2289bd1d206d0207b4f709495@79.170.208.149 CSeq: 102 INVITE Via: SIP/2.0/UDP 79.170.208.149:5060;branch=z9hG4bK3ef669fe Server: Linksys/SPA2102-5.1.9 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- DataNode1*CLI> <--- SIP read from 92.61.51.210:5060 ---> SIP/2.0 180 Ringing To: ;tag=f20d007e52df9fe1i0 From: "431311257433" ;tag=as637892c8 Call-ID: 5a9564f2289bd1d206d0207b4f709495@79.170.208.149 CSeq: 102 INVITE Via: SIP/2.0/UDP 79.170.208.149:5060;branch=z9hG4bK3ef669fe Server: Linksys/SPA2102-5.1.9 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- DataNode1*CLI> <--- SIP read from 92.61.51.210:5060 ---> SIP/2.0 200 OK To: ;tag=f20d007e52df9fe1i0 From: "431311257433" ;tag=as637892c8 Call-ID: 5a9564f2289bd1d206d0207b4f709495@79.170.208.149 CSeq: 102 INVITE Via: SIP/2.0/UDP 79.170.208.149:5060;branch=z9hG4bK3ef669fe Contact: 4313112573 Server: Linksys/SPA2102-5.1.9 Content-Length: 315 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces Content-Type: application/sdp v=0 o=- 59679688 59679688 IN IP4 92.61.51.210 s=- c=IN IP4 92.61.51.210 t=0 0 m=audio 16434 RTP/AVP 18 102 101 96 a=rtpmap:18 G729a/8000 a=fmtp:18 annexb=no a=rtpmap:102 NSE/8000 a=fmtp:102 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:96 X-nt-inforeq/8000 a=ptime:40 a=sendrecv <-------------> --- (12 headers 15 lines) --- Found RTP audio format 18 Found RTP audio format 102 Found RTP audio format 101 Found RTP audio format 96 Peer audio RTP is at port 92.61.51.210:16434 Found audio description format G729a for ID 18 Found unknown media description format NSE for ID 102 Found audio description format telephone-event for ID 101 Found unknown media description format X-nt-inforeq for ID 96 Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 92.61.51.210:16434 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 92.61.51.210, port 5060 Transmitting (no NAT) to 92.61.51.210:5060: ACK sip:4313112573@92.61.51.210:5060 SIP/2.0 Via: SIP/2.0/UDP 79.170.208.149:5060;branch=z9hG4bK221e9d10;rport From: "431311257433" ;tag=as637892c8 To: ;tag=f20d007e52df9fe1i0 Contact: Call-ID: 5a9564f2289bd1d206d0207b4f709495@79.170.208.149 CSeq: 102 ACK User-Agent: GlobalCall+T.38 Max-Forwards: 70 Remote-Party-ID: "431311257433" ;privacy=off;screen=yes Content-Length: 0 --- [Mar 25 15:56:01] NOTICE[1480]: chan_sip.c:6678 add_sdp: add audio = 1 add t38 = 1 set_destination: Parsing for address/port to send to set_destination: set destination to 92.61.51.210, port 5060 [Mar 25 15:56:01] NOTICE[1480]: chan_sip.c:6678 add_sdp: add audio = 1 add t38 = 0 Audio is at 79.170.208.149 port 17202 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 92.61.51.210:5060: INVITE sip:4313112573@92.61.51.210:5060 SIP/2.0 Via: SIP/2.0/UDP 79.170.208.149:5060;branch=z9hG4bK2d9771f0;rport From: "431311257433" ;tag=as637892c8 To: ;tag=f20d007e52df9fe1i0 Contact: Call-ID: 5a9564f2289bd1d206d0207b4f709495@79.170.208.149 CSeq: 103 INVITE User-Agent: GlobalCall+T.38 Max-Forwards: 70 Remote-Party-ID: "431311257433" ;privacy=off;screen=yes Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 261 v=0 o=root 1405 1406 IN IP4 92.61.51.212 s=session c=IN IP4 92.61.51.212 t=0 0 m=audio 29000 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Mar 25 15:56:01] NOTICE[1421]: chan_sip.c:6678 add_sdp: add audio = 1 add t38 = 0 DataNode1*CLI> <--- SIP read from 92.61.51.210:5060 ---> SIP/2.0 200 OK To: ;tag=f20d007e52df9fe1i0 From: "431311257433" ;tag=as637892c8 Call-ID: 5a9564f2289bd1d206d0207b4f709495@79.170.208.149 CSeq: 103 INVITE Via: SIP/2.0/UDP 79.170.208.149:5060;branch=z9hG4bK2d9771f0 Contact: 4313112573 Server: Linksys/SPA2102-5.1.9 Content-Length: 342 Content-Type: application/sdp v=0 o=- 59679999 59679999 IN IP4 92.61.51.210 s=- c=IN IP4 92.61.51.210 t=0 0 m=audio 16434 RTP/AVP 18 102 101 96 a=rtpmap:18 G729a/8000 a=fmtp:18 annexb=no a=rtpmap:102 NSE/8000 a=fmtp:102 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:96 X-nt-inforeq/8000 a=ptime:40 a=sendrecv a=silenceSupp:off - - - - <-------------> --- (10 headers 16 lines) --- Found RTP audio format 18 Found RTP audio format 102 Found RTP audio format 101 Found RTP audio format 96 Peer audio RTP is at port 92.61.51.210:16434 Found audio description format G729a for ID 18 Found unknown media description format NSE for ID 102 Found audio description format telephone-event for ID 101 Found unknown media description format X-nt-inforeq for ID 96 Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 92.61.51.210:16434 [Mar 25 15:56:01] WARNING[1421]: chan_sip.c:12569 handle_response_invite: RTP re-invite after T38 session not handled yet ! Scheduling destruction of SIP dialog '5a9564f2289bd1d206d0207b4f709495@79.170.208.149' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 92.61.51.210, port 5060 Transmitting (no NAT) to 92.61.51.210:5060: ACK sip:4313112573@92.61.51.210:5060 SIP/2.0 Via: SIP/2.0/UDP 79.170.208.149:5060;branch=z9hG4bK0b9cb731;rport From: "431311257433" ;tag=as637892c8 To: ;tag=f20d007e52df9fe1i0 Contact: Call-ID: 5a9564f2289bd1d206d0207b4f709495@79.170.208.149 CSeq: 103 ACK User-Agent: GlobalCall+T.38 Max-Forwards: 70 Remote-Party-ID: "431311257433" ;privacy=off;screen=yes Content-Length: 0 --- DataNode1*CLI> <--- SIP read from 92.61.51.210:5060 ---> REGISTER sip:79.170.208.149 SIP/2.0 Via: SIP/2.0/UDP 92.61.51.210:5060;branch=z9hG4bK-fde00f74 From: 4313112573 ;tag=18ad9458190eb9fo0 To: 4313112573 Call-ID: c83c630f-9b4a12e5@92.61.51.210 CSeq: 35560 REGISTER Max-Forwards: 70 Authorization: Digest username="4313112573",realm="asterisk",nonce="016502fc",uri="sip:79.170.208.149",algorithm=MD5,response="60faf2cf074f7b03da4bd27ff99e5715" Contact: 4313112573 ;expires=60 User-Agent: Linksys/SPA2102-5.1.9 Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces <-------------> --- (13 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 92.61.51.210 : 5060 (no NAT) DataNode1*CLI> <--- Transmitting (no NAT) to 92.61.51.210:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 92.61.51.210:5060;branch=z9hG4bK-fde00f74;received=92.61.51.210 From: 4313112573 ;tag=18ad9458190eb9fo0 To: 4313112573 Call-ID: c83c630f-9b4a12e5@92.61.51.210 CSeq: 35560 REGISTER User-Agent: GlobalCall+T.38 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> DataNode1*CLI> <--- Transmitting (no NAT) to 92.61.51.210:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 92.61.51.210:5060;branch=z9hG4bK-fde00f74;received=92.61.51.210 From: 4313112573 ;tag=18ad9458190eb9fo0 To: 4313112573 ;tag=as45889c15 Call-ID: c83c630f-9b4a12e5@92.61.51.210 CSeq: 35560 REGISTER User-Agent: GlobalCall+T.38 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="021232b4" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'c83c630f-9b4a12e5@92.61.51.210' in 32000 ms (Method: REGISTER) DataNode1*CLI> <--- SIP read from 92.61.51.210:5060 ---> REGISTER sip:79.170.208.149 SIP/2.0 Via: SIP/2.0/UDP 92.61.51.210:5060;branch=z9hG4bK-28641a8d From: 4313112573 ;tag=18ad9458190eb9fo0 To: 4313112573 Call-ID: c83c630f-9b4a12e5@92.61.51.210 CSeq: 35561 REGISTER Max-Forwards: 70 Authorization: Digest username="4313112573",realm="asterisk",nonce="021232b4",uri="sip:79.170.208.149",algorithm=MD5,response="a6ee9389cd89a6f59921e39d56bd3b5f" Contact: 4313112573 ;expires=60 User-Agent: Linksys/SPA2102-5.1.9 Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces <-------------> --- (13 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 92.61.51.210 : 5060 (no NAT) DataNode1*CLI> <--- Transmitting (no NAT) to 92.61.51.210:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 92.61.51.210:5060;branch=z9hG4bK-28641a8d;received=92.61.51.210 From: 4313112573 ;tag=18ad9458190eb9fo0 To: 4313112573 Call-ID: c83c630f-9b4a12e5@92.61.51.210 CSeq: 35561 REGISTER User-Agent: GlobalCall+T.38 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> Reliably Transmitting (no NAT) to 92.61.51.210:5060: OPTIONS sip:4313112573@92.61.51.210:5060 SIP/2.0 Via: SIP/2.0/UDP 79.170.208.149:5060;branch=z9hG4bK2d7b1585;rport From: "" ;tag=as6ac55100 To: Contact: Call-ID: 15c4a9d64d86f3115587c81778f0d068@79.170.208.149 CSeq: 102 OPTIONS User-Agent: GlobalCall+T.38 Max-Forwards: 70 Date: Wed, 25 Mar 2009 14:56:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Mar 25 15:56:17] WARNING[1421]: res_config_mysql.c:360 update_mysql: MySQL RealTime: Failed to query database. Check debug for more info. <--- Transmitting (no NAT) to 92.61.51.210:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.61.51.210:5060;branch=z9hG4bK-28641a8d;received=92.61.51.210 From: 4313112573 ;tag=18ad9458190eb9fo0 To: 4313112573 ;tag=as45889c15 Call-ID: c83c630f-9b4a12e5@92.61.51.210 CSeq: 35561 REGISTER User-Agent: GlobalCall+T.38 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: ;expires=60 Date: Wed, 25 Mar 2009 14:56:17 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'c83c630f-9b4a12e5@92.61.51.210' in 32000 ms (Method: REGISTER) DataNode1*CLI> <--- SIP read from 92.61.51.210:5060 ---> SIP/2.0 200 OK To: ;tag=fee3d98c5fa247fbi0 From: "" ;tag=as6ac55100 Call-ID: 15c4a9d64d86f3115587c81778f0d068@79.170.208.149 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 79.170.208.149:5060;branch=z9hG4bK2d7b1585 Server: Linksys/SPA2102-5.1.9 Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '15c4a9d64d86f3115587c81778f0d068@79.170.208.149' Method: OPTIONS [Mar 25 15:56:33] WARNING[1421]: chan_sip.c:2131 __sip_autodestruct: Autodestruct on dialog '5a9564f2289bd1d206d0207b4f709495@79.170.208.149' with owner in place (Method: INVITE) [Mar 25 15:56:33] NOTICE[1480]: chan_sip.c:6678 add_sdp: add audio = 1 add t38 = 0 set_destination: Parsing for address/port to send to set_destination: set destination to 92.61.51.210, port 5060 [Mar 25 15:56:33] NOTICE[1480]: chan_sip.c:6678 add_sdp: add audio = 1 add t38 = 0 Audio is at 79.170.208.149 port 17202 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 92.61.51.210:5060: INVITE sip:4313112573@92.61.51.210:5060 SIP/2.0 Via: SIP/2.0/UDP 79.170.208.149:5060;branch=z9hG4bK000c7e12;rport From: "431311257433" ;tag=as637892c8 To: ;tag=f20d007e52df9fe1i0 Contact: Call-ID: 5a9564f2289bd1d206d0207b4f709495@79.170.208.149 CSeq: 104 INVITE User-Agent: GlobalCall+T.38 Max-Forwards: 70 Remote-Party-ID: "431311257433" ;privacy=off;screen=yes Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 265 v=0 o=root 1405 1407 IN IP4 79.170.208.149 s=session c=IN IP4 79.170.208.149 t=0 0 m=audio 17202 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- Scheduling destruction of SIP dialog '5a9564f2289bd1d206d0207b4f709495@79.170.208.149' in 32000 ms (Method: INVITE) DataNode1*CLI> <--- SIP read from 92.61.51.210:5060 ---> SIP/2.0 200 OK To: ;tag=f20d007e52df9fe1i0 From: "431311257433" ;tag=as637892c8 Call-ID: 5a9564f2289bd1d206d0207b4f709495@79.170.208.149 CSeq: 104 INVITE Via: SIP/2.0/UDP 79.170.208.149:5060;branch=z9hG4bK000c7e12 Contact: 4313112573 Server: Linksys/SPA2102-5.1.9 Content-Length: 342 Content-Type: application/sdp v=0 o=- 59683204 59683204 IN IP4 92.61.51.210 s=- c=IN IP4 92.61.51.210 t=0 0 m=audio 16434 RTP/AVP 18 102 101 96 a=rtpmap:18 G729a/8000 a=fmtp:18 annexb=no a=rtpmap:102 NSE/8000 a=fmtp:102 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:96 X-nt-inforeq/8000 a=ptime:40 a=sendrecv a=silenceSupp:off - - - - <-------------> --- (10 headers 16 lines) --- Found RTP audio format 18 Found RTP audio format 102 Found RTP audio format 101 Found RTP audio format 96 Peer audio RTP is at port 92.61.51.210:16434 Found audio description format G729a for ID 18 Found unknown media description format NSE for ID 102 Found audio description format telephone-event for ID 101 Found unknown media description format X-nt-inforeq for ID 96 Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 92.61.51.210:16434 set_destination: Parsing for address/port to send to set_destination: set destination to 92.61.51.210, port 5060 Transmitting (no NAT) to 92.61.51.210:5060: ACK sip:4313112573@92.61.51.210:5060 SIP/2.0 Via: SIP/2.0/UDP 79.170.208.149:5060;branch=z9hG4bK0c8f9eef;rport From: "431311257433" ;tag=as637892c8 To: ;tag=f20d007e52df9fe1i0 Contact: Call-ID: 5a9564f2289bd1d206d0207b4f709495@79.170.208.149 CSeq: 104 ACK User-Agent: GlobalCall+T.38 Max-Forwards: 70 Remote-Party-ID: "431311257433" ;privacy=off;screen=yes Content-Length: 0 --- set_destination: Parsing for address/port to send to set_destination: set destination to 92.61.51.210, port 5060 Reliably Transmitting (no NAT) to 92.61.51.210:5060: BYE sip:4313112573@92.61.51.210:5060 SIP/2.0 Via: SIP/2.0/UDP 79.170.208.149:5060;branch=z9hG4bK7f2801d5;rport From: "431311257433" ;tag=as637892c8 To: ;tag=f20d007e52df9fe1i0 Call-ID: 5a9564f2289bd1d206d0207b4f709495@79.170.208.149 CSeq: 105 BYE User-Agent: GlobalCall+T.38 Max-Forwards: 70 Remote-Party-ID: "431311257433" ;privacy=off;screen=yes X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- Scheduling destruction of SIP dialog '5a9564f2289bd1d206d0207b4f709495@79.170.208.149' in 32000 ms (Method: INVITE) DataNode1*CLI> <--- SIP read from 92.61.51.210:5060 ---> SIP/2.0 200 OK To: ;tag=f20d007e52df9fe1i0 From: "431311257433" ;tag=as637892c8 Call-ID: 5a9564f2289bd1d206d0207b4f709495@79.170.208.149 CSeq: 105 BYE Via: SIP/2.0/UDP 79.170.208.149:5060;branch=z9hG4bK7f2801d5 Server: Linksys/SPA2102-5.1.9 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '5a9564f2289bd1d206d0207b4f709495@79.170.208.149' Method: INVITE DataNode1*CLI>