DataNode1*CLI> sip set debug ip ip-teles-a SIP Debugging Enabled for IP: ip-teles-a DataNode1*CLI> DataNode1*CLI> DataNode1*CLI> DataNode1*CLI> DataNode1*CLI> DataNode1*CLI> DataNode1*CLI> <--- SIP read from ip-teles-a:5060 ---> INVITE sip:016163090708@ip-asterisk SIP/2.0 Max-Forwards: 50 Via: SIP/2.0/UDP ip-teles-a:5060;rport;branch=z9hG4bK32941660233581811307235 From: ;tag=335617924839647731749698036654 To: Call-ID: 811442636910458216084781704695@ip-teles-a CSeq: 1 INVITE Contact: User-Agent: TELES.VoIPBOX 14.0a 905 Allow: INVITE,ACK,CANCEL,BYE,UPDATE,REGISTER,PRACK,INFO,NOTIFY,REFER Timestamp: 1238078467 Content-Type: application/sdp Content-Length: 380 v=0 o=- 4 1 IN IP4 ip-teles-a s=- c=IN IP4 ip-teles-a t=0 0 m=audio 29000 RTP/AVP 18 a=sendrecv a=ptime:20 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no 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-teles-a : 5060 (NAT) Using INVITE request as basis request - 811442636910458216084781704695@ip-teles-a Found peer '4313112574' Found RTP audio format 18 Got T.38 offer in SDP in dialog 811442636910458216084781704695@ip-teles-a Peer audio RTP is at port ip-teles-a: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-teles-a:29000 Looking for 016163090708 in PBX (domain ip-asterisk) list_route: hop: DataNode1*CLI> <--- Transmitting (no NAT) to ip-teles-a:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP ip-teles-a:5060;branch=z9hG4bK32941660233581811307235;received=ip-teles-a;rport=5060 From: ;tag=335617924839647731749698036654 To: Call-ID: 811442636910458216084781704695@ip-teles-a CSeq: 1 INVITE User-Agent: GlobalCall+T.38 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Executing [016163090708@PBX:1] Goto("SIP/ip-teles-a-18b18310", "004316163090708|1") in new stack -- Goto (PBX,004316163090708,1) -- Executing [004316163090708@PBX:1] Dial("SIP/ip-teles-a-18b18310", "SIP/61054316163090708@ip-teles-b|90") in new stack -- Called 61054316163090708@ip-teles-b -- SIP/ip-teles-b-18b67550 is making progress passing it to SIP/ip-teles-a-18b18310 Audio is at ip-asterisk port 13260 Adding codec 0x100 (g729) to SDP DataNode1*CLI> <--- Transmitting (no NAT) to ip-teles-a:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP ip-teles-a:5060;branch=z9hG4bK32941660233581811307235;received=ip-teles-a;rport=5060 From: ;tag=335617924839647731749698036654 To: ;tag=as77bd760e Call-ID: 811442636910458216084781704695@ip-teles-a 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: 395 v=0 o=root 18769 18769 IN IP4 ip-asterisk s=session c=IN IP4 ip-asterisk t=0 0 m=audio 13260 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - a=ptime:20 a=sendrecv m=image 8578 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:400 a=T38FaxMaxDatagram:400 a=T38FaxUdpEC:t38UDPRedundancy <------------> set_destination: Parsing for address/port to send to set_destination: set destination to ip-teles-a, port 5060 Audio is at ip-asterisk port 13260 Adding codec 0x100 (g729) to SDP Reliably Transmitting (no NAT) to ip-teles-a:5060: INVITE sip:33@ip-teles-a SIP/2.0 Via: SIP/2.0/UDP ip-asterisk:5060;branch=z9hG4bK7ae43c16;rport From: ;tag=as77bd760e To: ;tag=335617924839647731749698036654 Contact: Call-ID: 811442636910458216084781704695@ip-teles-a 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: 211 v=0 o=root 18769 18770 IN IP4 ip-asterisk s=session c=IN IP4 ip-asterisk t=0 0 m=audio 13260 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- DataNode1*CLI> <--- SIP read from ip-teles-a:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP ip-asterisk:5060;branch=z9hG4bK7ae43c16;rport From: ;tag=as77bd760e To: ;tag=335617924839647731749698036654 Call-ID: 811442636910458216084781704695@ip-teles-a CSeq: 102 INVITE Allow: INVITE,ACK,CANCEL,BYE,UPDATE,REGISTER,PRACK,INFO,NOTIFY,REFER Contact: User-Agent: TELES.VoIPBOX 14.0a 905 Content-Type: application/sdp Content-Length: 379 =0taNode1*CLI> o=- 4 1 IN IP4 ip-teles-a s=- c=IN IP4 ip-teles-a t=0 0 m=audio 29000 RTP/AVP 18 a=sendrecv a=ptime:20 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no m=image 29000 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxUdpEC:t38UDPRedundancy a=T38FaxRateManagement:transferredTCF a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 <-------------> --- (11 headers 18 lines) --- Found RTP audio format 18 Got T.38 offer in SDP in dialog 811442636910458216084781704695@ip-teles-a Peer audio RTP is at port ip-teles-a: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-teles-a:29000 set_destination: Parsing for address/port to send to set_destination: set destination to ip-teles-a, port 5060 Transmitting (no NAT) to ip-teles-a:5060: ACK sip:33@ip-teles-a SIP/2.0 Via: SIP/2.0/UDP ip-asterisk:5060;branch=z9hG4bK1429cb60;rport From: ;tag=as77bd760e To: ;tag=335617924839647731749698036654 Contact: Call-ID: 811442636910458216084781704695@ip-teles-a CSeq: 102 ACK User-Agent: GlobalCall+T.38 Max-Forwards: 70 Content-Length: 0 --- -- SIP/ip-teles-b-18b67550 is ringing DataNode1*CLI> <--- Transmitting (no NAT) to ip-teles-a:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP ip-asterisk:5060;branch=z9hG4bK7ae43c16;received=ip-teles-a;rport=5060 From: ;tag=as77bd760e To: ;tag=335617924839647731749698036654 Call-ID: 811442636910458216084781704695@ip-teles-a CSeq: 102 INVITE User-Agent: GlobalCall+T.38 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- SIP/ip-teles-b-18b67550 is making progress passing it to SIP/ip-teles-a-18b18310 -- SIP/ip-teles-b-18b67550 answered SIP/ip-teles-a-18b18310 Audio is at ip-asterisk port 13260 Adding codec 0x100 (g729) to SDP DataNode1*CLI> <--- Reliably Transmitting (no NAT) to ip-teles-a:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP ip-asterisk:5060;branch=z9hG4bK7ae43c16;received=ip-teles-a;rport=5060 From: ;tag=as77bd760e To: ;tag=335617924839647731749698036654 Call-ID: 811442636910458216084781704695@ip-teles-a CSeq: 102 INVITE User-Agent: GlobalCall+T.38 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 211 v=0 o=root 18769 18771 IN IP4 ip-asterisk s=session c=IN IP4 ip-asterisk t=0 0 m=audio 13260 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> -- Native bridging SIP/ip-teles-a-18b18310 and SIP/ip-teles-b-18b67550 DataNode1*CLI> <--- SIP read from ip-teles-a:5060 ---> ACK sip:016163090708@ip-asterisk SIP/2.0 Max-Forwards: 50 Via: SIP/2.0/UDP ip-teles-a:5060;rport;branch=z9hG4bK23121256480105426795910 From: ;tag=as77bd760e To: ;tag=335617924839647731749698036654 User-Agent: TELES.VoIPBOX 14.0a 905 Call-ID: 811442636910458216084781704695@ip-teles-a CSeq: 102 ACK Allow: INVITE,ACK,CANCEL,BYE,UPDATE,REGISTER,PRACK,INFO,NOTIFY,REFER Content-Length: 0 <-------------> --- (10 headers 0 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to ip-teles-a, port 5060 Audio is at ip-asterisk port 13260 Adding codec 0x100 (g729) to SDP Reliably Transmitting (no NAT) to ip-teles-a:5060: INVITE sip:33@ip-teles-a SIP/2.0 Via: SIP/2.0/UDP ip-asterisk:5060;branch=z9hG4bK4b94c39d;rport From: ;tag=as77bd760e To: ;tag=335617924839647731749698036654 Contact: Call-ID: 811442636910458216084781704695@ip-teles-a 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: 211 v=0 o=root 18769 18772 IN IP4 ip-teles-b s=session c=IN IP4 ip-teles-b t=0 0 m=audio 29000 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- DataNode1*CLI> <--- SIP read from ip-teles-a:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP ip-asterisk:5060;branch=z9hG4bK4b94c39d;rport From: ;tag=as77bd760e To: ;tag=335617924839647731749698036654 Call-ID: 811442636910458216084781704695@ip-teles-a CSeq: 103 INVITE Allow: INVITE,ACK,CANCEL,BYE,UPDATE,REGISTER,PRACK,INFO,NOTIFY,REFER Contact: User-Agent: TELES.VoIPBOX 14.0a 905 Content-Type: application/sdp Content-Length: 163 =0taNode1*CLI> o=- 4 1 IN IP4 ip-teles-a s=- c=IN IP4 ip-teles-a t=0 0 m=audio 29000 RTP/AVP 18 a=sendrecv a=ptime:20 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no <-------------> --- (11 headers 10 lines) --- Found RTP audio format 18 Peer audio RTP is at port ip-teles-a: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-teles-a:29000 set_destination: Parsing for address/port to send to set_destination: set destination to ip-teles-a, port 5060 Transmitting (no NAT) to ip-teles-a:5060: ACK sip:33@ip-teles-a SIP/2.0 Via: SIP/2.0/UDP ip-asterisk:5060;branch=z9hG4bK1078b3e9;rport From: ;tag=as77bd760e To: ;tag=335617924839647731749698036654 Contact: Call-ID: 811442636910458216084781704695@ip-teles-a 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-teles-a, port 5060 Audio is at ip-asterisk port 13260 Adding codec 0x100 (g729) to SDP Reliably Transmitting (no NAT) to ip-teles-a:5060: INVITE sip:33@ip-teles-a SIP/2.0 Via: SIP/2.0/UDP ip-asterisk:5060;branch=z9hG4bK374b0d3f;rport From: ;tag=as77bd760e To: ;tag=335617924839647731749698036654 Contact: Call-ID: 811442636910458216084781704695@ip-teles-a CSeq: 104 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: 211 v=0 o=root 18769 18773 IN IP4 ip-asterisk s=session c=IN IP4 ip-asterisk t=0 0 m=audio 13260 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- == Spawn extension (PBX, 004316163090708, 1) exited non-zero on 'SIP/ip-teles-a-18b18310' Scheduling destruction of SIP dialog '811442636910458216084781704695@ip-teles-a' in 32000 ms (Method: ACK) DataNode1*CLI> <--- SIP read from ip-teles-a:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP ip-asterisk:5060;branch=z9hG4bK374b0d3f;rport From: ;tag=as77bd760e To: ;tag=335617924839647731749698036654 Call-ID: 811442636910458216084781704695@ip-teles-a CSeq: 104 INVITE Allow: INVITE,ACK,CANCEL,BYE,UPDATE,REGISTER,PRACK,INFO,NOTIFY,REFER Contact: User-Agent: TELES.VoIPBOX 14.0a 905 Content-Type: application/sdp Content-Length: 163 =0taNode1*CLI> o=- 4 1 IN IP4 ip-teles-a s=- c=IN IP4 ip-teles-a t=0 0 m=audio 29000 RTP/AVP 18 a=sendrecv a=ptime:20 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no <-------------> --- (11 headers 10 lines) --- Found RTP audio format 18 Peer audio RTP is at port ip-teles-a: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-teles-a:29000 set_destination: Parsing for address/port to send to set_destination: set destination to ip-teles-a, port 5060 Transmitting (no NAT) to ip-teles-a:5060: ACK sip:33@ip-teles-a SIP/2.0 Via: SIP/2.0/UDP ip-asterisk:5060;branch=z9hG4bK05d2461f;rport From: ;tag=as77bd760e To: ;tag=335617924839647731749698036654 Contact: Call-ID: 811442636910458216084781704695@ip-teles-a CSeq: 104 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-teles-a, port 5060 Reliably Transmitting (no NAT) to ip-teles-a:5060: BYE sip:33@ip-teles-a SIP/2.0 Via: SIP/2.0/UDP ip-asterisk:5060;branch=z9hG4bK78691c85;rport From: ;tag=as77bd760e To: ;tag=335617924839647731749698036654 Call-ID: 811442636910458216084781704695@ip-teles-a CSeq: 105 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 '811442636910458216084781704695@ip-teles-a' in 32000 ms (Method: ACK) DataNode1*CLI> <--- SIP read from ip-teles-a:5060 ---> CANCEL sip:016163090708@ip-asterisk SIP/2.0 Max-Forwards: 50 Via: SIP/2.0/UDP ip-teles-a:5060;rport;branch=z9hG4bK32941660233581811307235 From: ;tag=335617924839647731749698036654 To: Contact: User-Agent: TELES.VoIPBOX 14.0a 905 Call-ID: 811442636910458216084781704695@ip-teles-a CSeq: 1 CANCEL Allow: INVITE,ACK,CANCEL,BYE,UPDATE,REGISTER,PRACK,INFO,NOTIFY,REFER Content-Length: 0 <-------------> --- (11 headers 0 lines) --- DataNode1*CLI> <--- Transmitting (no NAT) to ip-teles-a:5060 ---> SIP/2.0 503 Server error Via: SIP/2.0/UDP ip-teles-a:5060;branch=z9hG4bK32941660233581811307235;received=ip-teles-a;rport=5060 From: ;tag=335617924839647731749698036654 To: ;tag=335617924839647731749698036654 Call-ID: 811442636910458216084781704695@ip-teles-a CSeq: 1 CANCEL User-Agent: GlobalCall+T.38 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> Retransmitting #1 (no NAT) to ip-teles-a:5060: BYE sip:33@ip-teles-a SIP/2.0 Via: SIP/2.0/UDP ip-asterisk:5060;branch=z9hG4bK78691c85;rport From: ;tag=as77bd760e To: ;tag=335617924839647731749698036654 Call-ID: 811442636910458216084781704695@ip-teles-a CSeq: 105 BYE User-Agent: GlobalCall+T.38 Max-Forwards: 70 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- DataNode1*CLI> <--- SIP read from ip-teles-a:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP ip-asterisk:5060;branch=z9hG4bK78691c85;rport From: ;tag=as77bd760e To: ;tag=335617924839647731749698036654 Call-ID: 811442636910458216084781704695@ip-teles-a CSeq: 105 BYE Allow: INVITE,ACK,CANCEL,BYE,UPDATE,REGISTER,PRACK,INFO,NOTIFY,REFER Contact: User-Agent: TELES.VoIPBOX 14.0a 905 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '811442636910458216084781704695@ip-teles-a' Method: CANCEL DataNode1*CLI>