Asterisk SVN-branch-1.4-r51360, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark Spencer Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= <--- SIP read from 10.0.0.1:5060 ---> INVITE sip:1234@mydomain.tld SIP/2.0 Record-Route: Via: SIP/2.0/UDP 10.0.0.1;branch=z9hG4bKd619.0869877.0 Via: SIP/2.0/UDP 10.2.0.1:5242;branch=z9hG4bK-c71e2cd6;rport=5242 From: ivoc ;tag=561074982381cbd6o0 To: Call-ID: 78a70d40-bcb7cb4e@10.2.0.1 CSeq: 102 INVITE Max-Forwards: 69 Contact: ivoc Expires: 240 User-Agent: Linksys/SPA2102-3.3.6 Content-Length: 262 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp X-Custom-Duration: 7200000 v=0 o=- 1066255 1066255 IN IP4 10.2.0.1 s=- c=IN IP4 10.2.0.1 t=0 0 m=audio 16474 RTP/AVP 18 100 101 a=rtpmap:18 G729a/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> --- (17 headers 13 lines) --- Sending to 10.0.0.1 : 5060 (no NAT) Using INVITE request as basis request - 78a70d40-bcb7cb4e@10.2.0.1 Found peer 'ser' Found RTP audio format 18 Found RTP audio format 100 Found RTP audio format 101 Peer audio RTP is at port 10.2.0.1:16474 Found description format G729a for ID 18 Found description format NSE for ID 100 Got unsupported a:fmtp in SDP offer Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x50a (gsm|alaw|g729|ilbc), 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 10.2.0.1:16474 Looking for 1234 in pre-process (domain mydomain.tld) list_route: hop: <--- Transmitting (NAT) to 10.0.0.1:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.0.1;branch=z9hG4bKd619.0869877.0;received=10.0.0.1 Via: SIP/2.0/UDP 10.2.0.1:5242;branch=z9hG4bK-c71e2cd6;rport=5242 From: ivoc ;tag=561074982381cbd6o0 To: Call-ID: 78a70d40-bcb7cb4e@10.2.0.1 CSeq: 102 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Executing [1234@pre-process:1] SIPAddHeader("SIP/mydomain.tld-081dc908", "X-Custom-Test3: ") in new stack -- Executing [1234@pre-process:2] SIPAddHeader("SIP/mydomain.tld-081dc908", "X-Custom-Network: class4") in new stack -- Executing [1234@pre-process:3] Dial("SIP/mydomain.tld-081dc908", "SIP/1234@ser") in new stack Audio is at 10.0.0.2 port 18980 Adding codec 0x100 (g729) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x400 (ilbc) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 10.0.0.1:5060: INVITE sip:1234@10.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK7c59b748;rport From: "ivoc" ;tag=as60c8dc97 To: Contact: Call-ID: 2e0535e624774ff447e9f60400cbe294@mydomain.tld CSeq: 102 INVITE User-Agent: Asterisk Max-Forwards: 70 Remote-Party-ID: "ivoc" ;privacy=off;screen=no Date: Tue, 23 Jan 2007 15:49:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-Custom-Network: class4 X-Custom-Test3: Content-Type: application/sdp Content-Length: 359 v=0 o=root 11811 11811 IN IP4 10.0.0.2 s=session c=IN IP4 10.0.0.2 t=0 0 m=audio 18980 RTP/AVP 18 8 3 97 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 1234@ser a-node2*CLI> <--- SIP read from 10.0.0.1:5060 ---> SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK7c59b748;rport=5060 From: "ivoc" ;tag=as60c8dc97 To: Call-ID: 2e0535e624774ff447e9f60400cbe294@mydomain.tld CSeq: 102 INVITE Server: OpenSer (1.1.0-notls (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- a-node2*CLI> <--- SIP read from 10.0.0.1:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK7c59b748;rport=5060 From: "ivoc" ;tag=as60c8dc97 To: ;tag=18D57F78-7E8 Date: Tue, 23 Jan 2007 15:49:18 GMT Call-ID: 2e0535e624774ff447e9f60400cbe294@mydomain.tld Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER Allow-Events: telephone-event Contact: Record-Route: Content-Disposition: session;handling=required Content-Type: application/sdp Content-Length: 299 v=0 o=CiscoSystemsSIP-GW-UserAgent 936 9294 IN IP4 10.3.0.1 s=SIP Call c=IN IP4 10.3.0.1 t=0 0 m=audio 17240 RTP/AVP 18 101 c=IN IP4 10.3.0.1 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=silenceSupp:off - - - - <-------------> --- (15 headers 13 lines) --- Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 10.3.0.1:17240 Found description format G729 for ID 18 Got unsupported a:fmtp in SDP offer Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x50a (gsm|alaw|g729|ilbc), 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 10.3.0.1:17240 -- SIP/ser-08228110 is making progress passing it to SIP/mydomain.tld-081dc908 Audio is at 10.0.0.2 port 11536 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Transmitting (NAT) to 10.0.0.1:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.0.0.1;branch=z9hG4bKd619.0869877.0;received=10.0.0.1 Via: SIP/2.0/UDP 10.2.0.1:5242;branch=z9hG4bK-c71e2cd6;rport=5242 From: ivoc ;tag=561074982381cbd6o0 To: ;tag=as77f9b275 Call-ID: 78a70d40-bcb7cb4e@10.2.0.1 CSeq: 102 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 267 v=0 o=root 11811 11811 IN IP4 10.0.0.2 s=session c=IN IP4 10.0.0.2 t=0 0 m=audio 11536 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 <------------> a-node2*CLI> <--- SIP read from 10.0.0.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK7c59b748;rport=5060 From: "ivoc" ;tag=as60c8dc97 To: ;tag=18D57F78-7E8 Date: Tue, 23 Jan 2007 15:49:18 GMT Call-ID: 2e0535e624774ff447e9f60400cbe294@mydomain.tld Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER Allow-Events: telephone-event Contact: Record-Route: Content-Type: application/sdp Content-Length: 299 v=0 o=CiscoSystemsSIP-GW-UserAgent 936 9294 IN IP4 10.3.0.1 s=SIP Call c=IN IP4 10.3.0.1 t=0 0 m=audio 17240 RTP/AVP 18 101 c=IN IP4 10.3.0.1 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=silenceSupp:off - - - - <-------------> --- (14 headers 13 lines) --- Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 10.3.0.1:17240 Found description format G729 for ID 18 Got unsupported a:fmtp in SDP offer Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x50a (gsm|alaw|g729|ilbc), 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 10.3.0.1:17240 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.0.0.1, port 5060 Transmitting (NAT) to 10.0.0.1:5060: ACK sip:1234@10.3.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK5623b0ea;rport Route: From: "ivoc" ;tag=as60c8dc97 To: ;tag=18D57F78-7E8 Contact: Call-ID: 2e0535e624774ff447e9f60400cbe294@mydomain.tld CSeq: 102 ACK User-Agent: Asterisk Max-Forwards: 70 Remote-Party-ID: "ivoc" ;privacy=off;screen=no Content-Length: 0 --- -- SIP/ser-08228110 answered SIP/mydomain.tld-081dc908 Audio is at 10.0.0.2 port 11536 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (NAT) to 10.0.0.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.1;branch=z9hG4bKd619.0869877.0;received=10.0.0.1 Via: SIP/2.0/UDP 10.2.0.1:5242;branch=z9hG4bK-c71e2cd6;rport=5242 Record-Route: From: ivoc ;tag=561074982381cbd6o0 To: ;tag=as77f9b275 Call-ID: 78a70d40-bcb7cb4e@10.2.0.1 CSeq: 102 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 267 v=0 o=root 11811 11812 IN IP4 10.0.0.2 s=session c=IN IP4 10.0.0.2 t=0 0 m=audio 11536 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 <------------> -- Packet2Packet bridging SIP/mydomain.tld-081dc908 and SIP/ser-08228110 a-node2*CLI> <--- SIP read from 10.0.0.1:5060 ---> ACK sip:1234@10.0.0.2:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.1;branch=z9hG4bKd619.0869877.2 Via: SIP/2.0/UDP 10.2.0.1:5242;branch=z9hG4bK-569e427c;rport=5242 From: ivoc ;tag=561074982381cbd6o0 To: ;tag=as77f9b275 Call-ID: 78a70d40-bcb7cb4e@10.2.0.1 CSeq: 102 ACK Max-Forwards: 69 Proxy-Authorization: Digest username="2222",realm="mydomain.tld",nonce="45b62fa93a7c18d121a691d82113b5d7c1d0f497",uri="sip:1234@10.0.0.2:5060",algorithm=MD5,response="cbe23abea8fe32c46d84647446fe9ced" Contact: ivoc User-Agent: Linksys/SPA2102-3.3.6 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- a-node2*CLI> <--- SIP read from 10.0.0.1:5060 ---> INVITE sip:2222@10.0.0.2:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.1;branch=z9hG4bK9a3e.61c2ea42.0 Via: SIP/2.0/UDP 10.3.0.1:5060;branch=z9hG4bK5CB549DA From: ;tag=18D57F78-7E8 To: "ivoc" ;tag=as60c8dc97 Date: Tue, 23 Jan 2007 15:49:27 GMT Call-ID: 2e0535e624774ff447e9f60400cbe294@mydomain.tld Supported: 100rel,timer,replaces Min-SE: 1800 Cisco-Guid: 507747979-2855277019-2520307638-1321745574 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER CSeq: 101 INVITE Max-Forwards: 69 Timestamp: 1169567367 Contact: Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 480 v=0 o=CiscoSystemsSIP-GW-UserAgent 936 9295 IN IP4 10.3.0.1 s=SIP Call c=IN IP4 10.3.0.1 t=0 0 m=image 17240 udptl 101 t38 c=IN IP4 10.3.0.1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=T38FaxVersion:0 a=T38MaxBitRate:7200 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:200 a=T38FaxMaxDatagram:72 a=T38FaxUdpEC:t38UDPRedundancy <-------------> --- (20 headers 19 lines) --- Sending to 10.0.0.1 : 5060 (NAT) Got T.38 offer in SDP in dialog 2e0535e624774ff447e9f60400cbe294@mydomain.tld Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. Callid 2e0535e624774ff447e9f60400cbe294@mydomain.tld Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x50a (gsm|alaw|g729|ilbc), peer - audio=0x0 (nothing)/video=0x0 (nothing), combined - 0x0 (nothing) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) set_destination: Parsing for address/port to send to set_destination: set destination to 10.0.0.1, port 5060 Reliably Transmitting (NAT) to 10.0.0.1:5060: INVITE sip:2222@10.2.0.1:5242 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK00efc4cf;rport Route: From: ;tag=as77f9b275 To: ivoc ;tag=561074982381cbd6o0 Contact: Call-ID: 78a70d40-bcb7cb4e@10.2.0.1 CSeq: 102 INVITE User-Agent: Asterisk Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-info: SIP re-invite (T38 switchover) Content-Type: application/sdp Content-Length: 351 v=0 o=root 11811 11813 IN IP4 10.0.0.2 s=session c=IN IP4 10.0.0.2 t=0 0 m=image 4640 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:7200 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:72 a=T38FaxMaxDatagram:72 a=T38FaxUdpEC:t38UDPRedundancy --- a-node2*CLI> <--- SIP read from 10.0.0.1:5060 ---> SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK00efc4cf;rport=5060 From: ;tag=as77f9b275 To: ivoc ;tag=561074982381cbd6o0 Call-ID: 78a70d40-bcb7cb4e@10.2.0.1 CSeq: 102 INVITE Server: OpenSer (1.1.0-notls (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- a-node2*CLI> <--- SIP read from 10.0.0.1:5060 ---> SIP/2.0 200 OK To: ivoc ;tag=561074982381cbd6o0 From: ;tag=as77f9b275 Call-ID: 78a70d40-bcb7cb4e@10.2.0.1 CSeq: 102 INVITE Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK00efc4cf;rport=5060 Contact: ivoc Server: Linksys/SPA2102-3.3.6 Content-Length: 271 Content-Type: application/sdp v=0 o=- 1067252 1067252 IN IP4 10.2.0.1 s=- c=IN IP4 10.2.0.1 t=0 0 m=image 16474 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:200 a=T38FaxMaxDatagram:200 a=T38FaxUdpEC:t38UDPRedundancy <-------------> --- (10 headers 12 lines) --- Got T.38 offer in SDP in dialog 78a70d40-bcb7cb4e@10.2.0.1 Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. Callid 78a70d40-bcb7cb4e@10.2.0.1 Capabilities: us - 0x50a (gsm|alaw|g729|ilbc), peer - audio=0x0 (nothing)/video=0x0 (nothing), combined - 0x0 (nothing) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) list_route: hop: <--- Reliably Transmitting (NAT) to 10.0.0.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.1;branch=z9hG4bK9a3e.61c2ea42.0;received=10.0.0.1 Via: SIP/2.0/UDP 10.3.0.1:5060;branch=z9hG4bK5CB549DA From: ;tag=18D57F78-7E8 To: "ivoc" ;tag=as60c8dc97 Call-ID: 2e0535e624774ff447e9f60400cbe294@mydomain.tld CSeq: 101 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 353 v=0 o=root 11811 11812 IN IP4 10.0.0.2 s=session c=IN IP4 10.0.0.2 t=0 0 m=image 4400 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:200 a=T38FaxMaxDatagram:200 a=T38FaxUdpEC:t38UDPRedundancy <------------> set_destination: Parsing for address/port to send to set_destination: set destination to 10.2.0.1, port 5242 Transmitting (NAT) to 10.0.0.1:5060: ACK sip:2222@10.2.0.1:5242 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK75dce0c2;rport From: ;tag=as77f9b275 To: ivoc ;tag=561074982381cbd6o0 Contact: Call-ID: 78a70d40-bcb7cb4e@10.2.0.1 CSeq: 102 ACK User-Agent: Asterisk Max-Forwards: 70 Content-Length: 0 --- a-node2*CLI> <--- SIP read from 10.0.0.1:5060 ---> ACK sip:2222@10.0.0.2:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.1;branch=z9hG4bK9a3e.61c2ea42.2 Via: SIP/2.0/UDP 10.3.0.1:5060;branch=z9hG4bK5CB552104 From: ;tag=18D57F78-7E8 To: "ivoc" ;tag=as60c8dc97 Date: Tue, 23 Jan 2007 15:49:27 GMT Call-ID: 2e0535e624774ff447e9f60400cbe294@mydomain.tld Max-Forwards: 69 CSeq: 101 ACK Content-Length: 0 <-------------> --- (10 headers 0 lines) --- a-node2*CLI> <--- SIP read from 10.0.0.1:5060 ---> INVITE sip:2222@10.0.0.2:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.1;branch=z9hG4bK6a3e.47761651.0 Via: SIP/2.0/UDP 10.3.0.1:5060;branch=z9hG4bK5CB5B259F From: ;tag=18D57F78-7E8 To: "ivoc" ;tag=as60c8dc97 Date: Tue, 23 Jan 2007 15:50:02 GMT Call-ID: 2e0535e624774ff447e9f60400cbe294@mydomain.tld Supported: 100rel,timer,replaces Min-SE: 1800 Cisco-Guid: 507747979-2855277019-2520307638-1321745574 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER CSeq: 102 INVITE Max-Forwards: 69 Timestamp: 1169567402 Contact: Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 492 v=0 o=CiscoSystemsSIP-GW-UserAgent 936 9297 IN IP4 10.3.0.1 s=SIP Call c=IN IP4 10.3.0.1 t=0 0 m=image 17240 udptl 101 t38 c=IN IP4 10.3.0.1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=T38FaxVersion:0 a=T38MaxBitRate:7200 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:200 a=T38FaxMaxDatagram:72 a=T38FaxUdpEC:t38UDPRedundancy a=sendonly <-------------> --- (20 headers 20 lines) --- Sending to 10.0.0.1 : 5060 (NAT) Got T.38 offer in SDP in dialog 2e0535e624774ff447e9f60400cbe294@mydomain.tld Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. Callid 2e0535e624774ff447e9f60400cbe294@mydomain.tld Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x50a (gsm|alaw|g729|ilbc), peer - audio=0x0 (nothing)/video=0x0 (nothing), combined - 0x0 (nothing) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) <--- Reliably Transmitting (NAT) to 10.0.0.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.1;branch=z9hG4bK6a3e.47761651.0;received=10.0.0.1 Via: SIP/2.0/UDP 10.3.0.1:5060;branch=z9hG4bK5CB5B259F From: ;tag=18D57F78-7E8 To: "ivoc" ;tag=as60c8dc97 Call-ID: 2e0535e624774ff447e9f60400cbe294@mydomain.tld CSeq: 102 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 351 v=0 o=root 11811 11813 IN IP4 10.0.0.2 s=session c=IN IP4 10.0.0.2 t=0 0 m=image 4400 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:7200 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:72 a=T38FaxMaxDatagram:72 a=T38FaxUdpEC:t38UDPRedundancy <------------> a-node2*CLI> <--- SIP read from 10.0.0.1:5060 ---> ACK sip:2222@10.0.0.2:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.1;branch=z9hG4bK6a3e.47761651.2 Via: SIP/2.0/UDP 10.3.0.1:5060;branch=z9hG4bK5CB5C115 From: ;tag=18D57F78-7E8 To: "ivoc" ;tag=as60c8dc97 Date: Tue, 23 Jan 2007 15:50:02 GMT Call-ID: 2e0535e624774ff447e9f60400cbe294@mydomain.tld Max-Forwards: 69 CSeq: 102 ACK Content-Length: 0 <-------------> --- (10 headers 0 lines) --- a-node2*CLI> <--- SIP read from 10.0.0.1:5060 ---> BYE sip:1234@10.0.0.2:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.1;branch=z9hG4bKe619.c04b9e72.0 Via: SIP/2.0/UDP 10.2.0.1:5242;branch=z9hG4bK-1be9c83d;rport=5242 From: ivoc ;tag=561074982381cbd6o0 To: ;tag=as77f9b275 Call-ID: 78a70d40-bcb7cb4e@10.2.0.1 CSeq: 103 BYE Max-Forwards: 69 Proxy-Authorization: Digest username="2222",realm="mydomain.tld",nonce="45b62fa93a7c18d121a691d82113b5d7c1d0f497",uri="sip:1234@10.0.0.2:5060",algorithm=MD5,response="4c0722e7574ba1aa2b0c38de8f3d0de8" User-Agent: Linksys/SPA2102-3.3.6 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Sending to 10.0.0.1 : 5060 (NAT) <--- Transmitting (NAT) to 10.0.0.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.1;branch=z9hG4bKe619.c04b9e72.0;received=10.0.0.1 Via: SIP/2.0/UDP 10.2.0.1:5242;branch=z9hG4bK-1be9c83d;rport=5242 From: ivoc ;tag=561074982381cbd6o0 To: ;tag=as77f9b275 Call-ID: 78a70d40-bcb7cb4e@10.2.0.1 CSeq: 103 BYE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> Scheduling destruction of SIP dialog '2e0535e624774ff447e9f60400cbe294@mydomain.tld' in 32000 ms (Method: ACK) set_destination: Parsing for address/port to send to set_destination: set destination to 10.0.0.1, port 5060 Reliably Transmitting (NAT) to 10.0.0.1:5060: BYE sip:1234@10.3.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK5d9a14be;rport Route: From: "ivoc" ;tag=as60c8dc97 To: ;tag=18D57F78-7E8 Call-ID: 2e0535e624774ff447e9f60400cbe294@mydomain.tld CSeq: 103 BYE User-Agent: Asterisk Max-Forwards: 70 Remote-Party-ID: "ivoc" ;privacy=off;screen=no Content-Length: 0 --- == Spawn extension (pre-process, 1234, 3) exited non-zero on 'SIP/mydomain.tld-081dc908' a-node2*CLI> <--- SIP read from 10.0.0.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK5d9a14be;rport=5060 From: "ivoc" ;tag=as60c8dc97 To: ;tag=18D57F78-7E8 Date: Tue, 23 Jan 2007 15:50:16 GMT Call-ID: 2e0535e624774ff447e9f60400cbe294@mydomain.tld Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 CSeq: 103 BYE <-------------> --- (9 headers 0 lines) --- SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog '2e0535e624774ff447e9f60400cbe294@mydomain.tld' Method: ACK Really destroying SIP dialog '78a70d40-bcb7cb4e@10.2.0.1' Method: BYE