Script started on Thu Nov 9 19:47:02 2006 localhost bin # exitasterisk -r Asterisk SVN-trunk-r47321, 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. ========================================================================= Connected to Asterisk SVN-trunk-r47321 currently running on localhost (pid = 20515) localhost*CLI> Verbosity is at least 4 localhost*CLI> -- Remote UNIX connection localhost*CLI> Really destroying SIP dialog '864effff6b7effff@192.168.254.140' Method: REGISTER localhost*CLI> <--- SIP read from 192.168.254.140:56613 ---> INVITE sip:201@192.168.254.96;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.254.140:56613;branch=z9hG4bK9b4600000768ffff From: ;tag=91afffff57860000 To: Contact: Supported: replaces Call-ID: e2b10000bdc50000@192.168.254.140 CSeq: 19576 INVITE User-Agent: Grandstream HT496 1.0.3.44 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 344 v=0 o=200 8000 8000 IN IP4 192.168.254.140 s=SIP Call c=IN IP4 192.168.254.140 t=0 0 m=audio 22708 RTP/AVP 0 8 4 18 2 15 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 <-------------> --- (13 headers 16 lines) --- Sending to 192.168.254.140 : 56613 (no NAT) Using INVITE request as basis request - e2b10000bdc50000@192.168.254.140 Found user '200' for '200' <--- Reliably Transmitting (no NAT) to 192.168.254.140:56613 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.254.140:56613;branch=z9hG4bK9b4600000768ffff;received=192.168.254.140 From: ;tag=91afffff57860000 To: ;tag=as1f558026 Call-ID: e2b10000bdc50000@192.168.254.140 CSeq: 19576 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="645745c2" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'e2b10000bdc50000@192.168.254.140' in 32000 ms (Method: INVITE) localhost*CLI> <--- SIP read from 192.168.254.140:56613 ---> ACK sip:201@192.168.254.96;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.254.140:56613;branch=z9hG4bK9b4600000768ffff From: ;tag=91afffff57860000 To: ;tag=as1f558026 Contact: Call-ID: e2b10000bdc50000@192.168.254.140 CSeq: 19576 ACK User-Agent: Grandstream HT496 1.0.3.44 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 <-------------> --- (11 headers 0 lines) --- localhost*CLI> <--- SIP read from 192.168.254.140:56613 ---> INVITE sip:201@192.168.254.96;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.254.140:56613;branch=z9hG4bKa2510000a4d70000 From: ;tag=91afffff57860000 To: Contact: Supported: replaces Authorization: Digest username="200", realm="asterisk", algorithm=MD5, uri="sip:201@192.168.254.96;user=phone", nonce="645745c2", response="aa2bdbf067f6a378d53dd86d7413a60a" Call-ID: e2b10000bdc50000@192.168.254.140 CSeq: 19577 INVITE User-Agent: Grandstream HT496 1.0.3.44 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 344 v=0 o=200 8000 8001 IN IP4 192.168.254.140 s=SIP Call c=IN IP4 192.168.254.140 t=0 0 m=audio 22708 RTP/AVP 0 8 4 18 2 15 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 <-------------> --- (14 headers 16 lines) --- Sending to 192.168.254.140 : 56613 (no NAT) Using INVITE request as basis request - e2b10000bdc50000@192.168.254.140 Found user '200' for '200' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 4 Found RTP audio format 18 Found RTP audio format 2 Found RTP audio format 15 Found RTP audio format 101 Peer audio RTP is at port 192.168.254.140:22708 [Nov 9 19:47:20] DEBUG[20548]: chan_sip.c:4889 process_sdp: Peer doesn't provide T.38 UDPTL Found description format PCMU for ID 0 Found description format PCMA for ID 8 Found description format G723 for ID 4 Found description format G729 for ID 18 Found description format G726-32 for ID 2 Found description format PCMU for ID 15 Found description format telephone-event for ID 101 Capabilities: us - 0x8 (alaw), peer - audio=0x90d (g723|ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.254.140:22708 Peer video RTP is at port 192.168.254.140:47799 Looking for 201 in from-sip (domain 192.168.254.96) list_route: hop: <--- Transmitting (no NAT) to 192.168.254.140:56613 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.254.140:56613;branch=z9hG4bKa2510000a4d70000;received=192.168.254.140 From: ;tag=91afffff57860000 To: Call-ID: e2b10000bdc50000@192.168.254.140 CSeq: 19577 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> localhost*CLI> -- Executing [201@from-sip:1] Dial("SIP/200-081c9198", "SIP/201|300") in new stack localhost*CLI> Audio is at 192.168.254.96 port 15584 localhost*CLI> Adding codec 0x8 (alaw) to SDP localhost*CLI> Adding non-codec 0x1 (telephone-event) to SDP localhost*CLI> Reliably Transmitting (no NAT) to 192.168.254.153:5060: INVITE sip:201@192.168.254.153 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.96:5060;branch=z9hG4bK40addaf4;rport From: "200" ;tag=as2cf96937 To: Contact: Call-ID: 41be8c9076a3761c784c4cd70685bc98@192.168.254.96 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 09 Nov 2006 19:47:20 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 244 v=0 o=root 6198 6198 IN IP4 192.168.254.140 s=session c=IN IP4 192.168.254.140 t=0 0 m=audio 22708 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- localhost*CLI> -- Called 201 localhost*CLI> <--- SIP read from 192.168.254.153:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.254.96:5060;branch=z9hG4bK40addaf4;rport From: "200" ;tag=as2cf96937 To: Call-ID: 41be8c9076a3761c784c4cd70685bc98@192.168.254.96 CSeq: 102 INVITE User-Agent: Grandstream HT496 1.0.3.44 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- localhost*CLI> <--- SIP read from 192.168.254.153:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.254.96:5060;branch=z9hG4bK40addaf4;rport From: "200" ;tag=as2cf96937 To: ;tag=29710000f1d10000 Call-ID: 41be8c9076a3761c784c4cd70685bc98@192.168.254.96 CSeq: 102 INVITE User-Agent: Grandstream HT496 1.0.3.44 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- localhost*CLI> -- SIP/201-081d1018 is ringing <--- Transmitting (no NAT) to 192.168.254.140:56613 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.254.140:56613;branch=z9hG4bKa2510000a4d70000;received=192.168.254.140 From: ;tag=91afffff57860000 To: ;tag=as7d247b51 Call-ID: e2b10000bdc50000@192.168.254.140 CSeq: 19577 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> localhost*CLI> <--- SIP read from 192.168.254.153:5060 ---> REGISTER sip:192.168.254.96 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.153;branch=z9hG4bKd1adffffc9a40000 From: "201" ;tag=cce60000dfcaffff To: Contact: Authorization: Digest username="201", realm="asterisk", algorithm=MD5, uri="sip:192.168.254.96", nonce="65f624ba", response="b1e65c211c3f5573f544f91f947c4e26" Call-ID: b3a4000003970000@192.168.254.153 CSeq: 174 REGISTER Expires: 3600 localhost*CLI> User-Agent: Grandstream HT496 1.0.3.44 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 <-------------> --- (13 headers 0 lines) --- localhost*CLI> Using latest REGISTER request as basis request Sending to 192.168.254.153 : 5060 (no NAT) <--- Transmitting (no NAT) to 192.168.254.153:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.254.153;branch=z9hG4bKd1adffffc9a40000;received=192.168.254.153 From: "201" ;tag=cce60000dfcaffff To: Call-ID: b3a4000003970000@192.168.254.153 CSeq: 174 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> <--- Transmitting (no NAT) to 192.168.254.153:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.254.153;branch=z9hG4bKd1adffffc9a40000;received=192.168.254.153 From: "201" ;tag=cce60000dfcaffff To: ;tag=as52c486eb Call-ID: b3a4000003970000@192.168.254.153 CSeq: 174 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5b60993e" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'b3a4000003970000@192.168.254.153' in 32000 ms (Method: REGISTER) localhost*CLI> <--- SIP read from 192.168.254.153:5060 ---> REGISTER sip:192.168.254.96 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.153;branch=z9hG4bK9cebffff6ef50000 From: "201" ;tag=cce60000dfcaffff To: Contact: Authorization: Digest username="201", realm="asterisk", algorithm=MD5, uri="sip:192.168.254.96", nonce="5b60993e", response="ec8f080aadf21566e13cdc6843bccf44" Call-ID: b3a4000003970000@192.168.254.153 CSeq: 175 REGISTER Expires: 3600 localhost*CLI> User-Agent: Grandstream HT496 1.0.3.44 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.254.153 : 5060 (no NAT) <--- Transmitting (no NAT) to 192.168.254.153:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.254.153;branch=z9hG4bK9cebffff6ef50000;received=192.168.254.153 From: "201" ;tag=cce60000dfcaffff localhost*CLI> To: Call-ID: b3a4000003970000@192.168.254.153 CSeq: 175 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> localhost*CLI> <--- Transmitting (no NAT) to 192.168.254.153:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.254.153;branch=z9hG4bK9cebffff6ef50000;received=192.168.254.153 From: "201" ;tag=cce60000dfcaffff To: ;tag=as52c486eb Call-ID: b3a4000003970000@192.168.254.153 CSeq: 175 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 180 Contact: ;expires=180 Date: Thu, 09 Nov 2006 19:47:20 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'b3a4000003970000@192.168.254.153' in 32000 ms (Method: REGISTER) localhost*CLI> <--- SIP read from 192.168.254.153:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.254.96:5060;branch=z9hG4bK40addaf4;rport From: "200" ;tag=as2cf96937 To: ;tag=29710000f1d10000 Call-ID: 41be8c9076a3761c784c4cd70685bc98@192.168.254.96 CSeq: 102 INVITE User-Agent: Grandstream HT496 1.0.3.44 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Supported: replaces Content-Length: 160 v=0 o=201 8000 8000 IN IP4 192.168.254.153 s=SIP Call c=IN IP4 192.168.254.153 t=0 0 m=audio 5004 RTP/AVP 8 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 <-------------> --- (12 headers 9 lines) --- Found RTP audio format 8 Peer audio RTP is at port 192.168.254.153:5004 [Nov 9 19:47:22] DEBUG[20548]: chan_sip.c:4889 process_sdp: Peer doesn't provide T.38 UDPTL Found description format PCMA for ID 8 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 192.168.254.153:5004 --- set_address_from_contact host '192.168.254.153' list_route: hop: [Nov 9 19:47:22] DEBUG[20548]: chan_sip.c:5608 reqprep: Strict routing enforced for session 41be8c9076a3761c784c4cd70685bc98@192.168.254.96 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.254.153, port 5060 Transmitting (no NAT) to 192.168.254.153:5060: ACK sip:201@192.168.254.153 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.96:5060;branch=z9hG4bK68987b1a;rport From: "200" ;tag=as2cf96937 To: ;tag=29710000f1d10000 Contact: Call-ID: 41be8c9076a3761c784c4cd70685bc98@192.168.254.96 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- localhost*CLI> -- SIP/201-081d1018 answered SIP/200-081c9198 Audio is at 192.168.254.96 port 11210 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 192.168.254.140:56613 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.254.140:56613;branch=z9hG4bKa2510000a4d70000;received=192.168.254.140 From: ;tag=91afffff57860000 To: ;tag=as7d247b51 Call-ID: e2b10000bdc50000@192.168.254.140 localhost*CLI> CSeq: 19577 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 243 v=0 o=root 6198 6198 IN IP4 192.168.254.153 s=session c=IN IP4 192.168.254.153 t=0 0 m=audio 5004 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> -- Native bridging SIP/200-081c9198 and SIP/201-081d1018 localhost*CLI> <--- SIP read from 192.168.254.140:56613 ---> ACK sip:201@192.168.254.96 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.140:56613;branch=z9hG4bK96fdffffd1670000 From: ;tag=91afffff57860000 To: ;tag=as7d247b51 Contact: Authorization: Digest username="200", realm="asterisk", algorithm=MD5, uri="sip:201@192.168.254.96", nonce="645745c2", response="f67c17b916b8d42f5e17ea5931e5406a" Call-ID: e2b10000bdc50000@192.168.254.140 CSeq: 19577 ACK User-Agent: Grandstream HT496 1.0.3.44 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 <-------------> --- (12 headers 0 lines) --- localhost*CLI> <--- SIP read from 192.168.254.153:5060 ---> INVITE sip:200@192.168.254.96 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.153;branch=z9hG4bK7e5300006dd70000 From: ;tag=29710000f1d10000 To: "200" ;tag=as2cf96937 Contact: Supported: replaces Call-ID: 41be8c9076a3761c784c4cd70685bc98@192.168.254.96 CSeq: 29765 INVITE User-Agent: Grandstream HT496 1.0.3.44 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 353 v=0 o=201 8000 8001 IN IP4 192.168.254.153 s=SIP Call c=IN IP4 192.168.254.153 t=0 0 m=image 5004 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:400 a=T38FaxMaxDatagram:280 a=T38FaxUdpEC:t38UDPRedundancy <-------------> --- (13 headers 15 lines) --- Sending to 192.168.254.153 : 5060 (no NAT) Got T.38 offer in SDP in dialog 41be8c9076a3761c784c4cd70685bc98@192.168.254.96 Peer doesn't provide audio [Nov 9 19:47:27] DEBUG[20548]: chan_sip.c:4885 process_sdp: Peer T.38 UDPTL is at port 192.168.254.153:5004 [Nov 9 19:47:27] DEBUG[20548]: chan_sip.c:5094 process_sdp: Our T38 capability = (3840), peer T38 capability (3872), joint T38 capability (3872) Capabilities: us - 0x8 (alaw), peer - audio=0x0 (nothing)/video=0x0 (nothing), combined - 0x0 (nothing) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) [Nov 9 19:47:27] DEBUG[20548]: chan_sip.c:5608 reqprep: Strict routing enforced for session e2b10000bdc50000@192.168.254.140 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.254.140, port 56613 [Nov 9 19:47:27] DEBUG[20548]: chan_sip.c:6006 add_t38_sdp: T.38 UDPTL is at 192.168.254.96 port 4736 [Nov 9 19:47:27] DEBUG[20548]: chan_sip.c:6012 add_t38_sdp: Our T38 capability (3840), peer T38 capability (3872), joint capability (3872) Reliably Transmitting (no NAT) to 192.168.254.140:56613: INVITE sip:200@192.168.254.140:56613;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.254.96:5060;branch=z9hG4bK652c433e;rport From: ;tag=as7d247b51 To: ;tag=91afffff57860000 Contact: Call-ID: e2b10000bdc50000@192.168.254.140 CSeq: 102 INVITE User-Agent: Asterisk PBX 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: 263 v=0 o=root 6198 6199 IN IP4 192.168.254.153 s=session c=IN IP4 192.168.254.153 t=0 0 m=image 5004 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:280 a=T38FaxMaxDatagram:280 a=T38FaxUdpEC:t38UDPRedundancy --- localhost*CLI> <--- SIP read from 192.168.254.140:56613 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.254.96:5060;branch=z9hG4bK652c433e;rport From: ;tag=as7d247b51 To: ;tag=91afffff57860000 Call-ID: e2b10000bdc50000@192.168.254.140 CSeq: 102 INVITE User-Agent: Grandstream HT496 1.0.3.44 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Supported: replaces Content-Length: 354 v=0 o=200 8000 8002 IN IP4 192.168.254.140 s=SIP Call c=IN IP4 192.168.254.140 t=0 0 m=image 22708 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:280 a=T38FaxMaxDatagram:280 a=T38FaxUdpEC:t38UDPRedundancy <-------------> --- (12 headers 15 lines) --- Got T.38 offer in SDP in dialog e2b10000bdc50000@192.168.254.140 Peer doesn't provide audio [Nov 9 19:47:27] DEBUG[20548]: chan_sip.c:4885 process_sdp: Peer T.38 UDPTL is at port 192.168.254.140:22708 [Nov 9 19:47:27] DEBUG[20548]: chan_sip.c:5094 process_sdp: Our T38 capability = (3840), peer T38 capability (3872), joint T38 capability (3872) Capabilities: us - 0x8 (alaw), peer - audio=0x0 (nothing)/video=0x0 (nothing), combined - 0x0 (nothing) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) --- set_address_from_contact host '192.168.254.140' list_route: hop: [Nov 9 19:47:27] DEBUG[20548]: chan_sip.c:6006 add_t38_sdp: T.38 UDPTL is at 192.168.254.96 port 4924 [Nov 9 19:47:27] DEBUG[20548]: chan_sip.c:6012 add_t38_sdp: Our T38 capability (3840), peer T38 capability (3872), joint capability (3872) <--- Reliably Transmitting (no NAT) to 192.168.254.153:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.254.153;branch=z9hG4bK7e5300006dd70000;received=192.168.254.153 From: ;tag=29710000f1d10000 To: "200" ;tag=as2cf96937 Call-ID: 41be8c9076a3761c784c4cd70685bc98@192.168.254.96 CSeq: 29765 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 264 v=0 o=root 6198 6199 IN IP4 192.168.254.140 s=session c=IN IP4 192.168.254.140 t=0 0 m=image 22708 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:280 a=T38FaxMaxDatagram:280 a=T38FaxUdpEC:t38UDPRedundancy <------------> [Nov 9 19:47:27] DEBUG[20548]: chan_sip.c:5608 reqprep: Strict routing enforced for session e2b10000bdc50000@192.168.254.140 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.254.140, port 56613 Transmitting (no NAT) to 192.168.254.140:56613: ACK sip:200@192.168.254.140:56613;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.254.96:5060;branch=z9hG4bK57dc2f34;rport From: ;tag=as7d247b51 To: ;tag=91afffff57860000 Contact: Call-ID: e2b10000bdc50000@192.168.254.140 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- localhost*CLI> [Nov 9 19:47:27] DEBUG[6198]: chan_sip.c:5608 reqprep: Strict routing enforced for session e2b10000bdc50000@192.168.254.140 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.254.140, port 56613 Audio is at 192.168.254.96 port 11210 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.254.140:56613: INVITE sip:200@192.168.254.140:56613;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.254.96:5060;branch=z9hG4bK48980fc3;rport From: ;tag=as7d247b51 To: ;tag=91afffff57860000 Contact: Call-ID: e2b10000bdc50000@192.168.254.140 CSeq: 103 INVITE User-Agent: Asterisk PBX 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: 242 v=0 o=root 6198 6200 IN IP4 192.168.254.96 s=session c=IN IP4 192.168.254.96 t=0 0 m=audio 11210 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- localhost*CLI> <--- SIP read from 192.168.254.153:5060 ---> ACK sip:200@192.168.254.96 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.153;branch=z9hG4bKdbfdffff85560000 From: ;tag=29710000f1d10000 To: "200" ;tag=as2cf96937 Contact: Call-ID: 41be8c9076a3761c784c4cd70685bc98@192.168.254.96 CSeq: 29765 ACK User-Agent: Grandstream HT496 1.0.3.44 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 <-------------> --- (11 headers 0 lines) --- [Nov 9 19:47:27] DEBUG[20548]: chan_sip.c:5608 reqprep: Strict routing enforced for session 41be8c9076a3761c784c4cd70685bc98@192.168.254.96 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.254.153, port 5060 Audio is at 192.168.254.96 port 15584 Adding codec 0x8 (alaw) to SDP Reliably Transmitting (no NAT) to 192.168.254.153:5060: INVITE sip:201@192.168.254.153 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.96:5060;branch=z9hG4bK39b363fd;rport From: "200" ;tag=as2cf96937 To: ;tag=29710000f1d10000 Contact: Call-ID: 41be8c9076a3761c784c4cd70685bc98@192.168.254.96 CSeq: 103 INVITE User-Agent: Asterisk PBX 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: 186 v=0 o=root 6198 6200 IN IP4 192.168.254.96 s=session c=IN IP4 192.168.254.96 t=0 0 m=audio 15584 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- localhost*CLI> <--- SIP read from 192.168.254.140:56613 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.254.96:5060;branch=z9hG4bK48980fc3;rport From: ;tag=as7d247b51 To: ;tag=91afffff57860000 Call-ID: e2b10000bdc50000@192.168.254.140 CSeq: 103 INVITE User-Agent: Grandstream HT496 1.0.3.44 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Supported: replaces Content-Length: 217 v=0 o=200 8000 8003 IN IP4 192.168.254.140 s=SIP Call c=IN IP4 192.168.254.140 t=0 0 m=audio 22708 RTP/AVP 8 101 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 <-------------> --- (12 headers 11 lines) --- Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.254.140:22708 [Nov 9 19:47:27] DEBUG[20548]: chan_sip.c:4889 process_sdp: Peer doesn't provide T.38 UDPTL Found description format PCMA for ID 8 Found description format telephone-event for ID 101 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.254.140:22708 Peer video RTP is at port 192.168.254.140:44983 --- set_address_from_contact host '192.168.254.140' [Nov 9 19:47:27] WARNING[20548]: chan_sip.c:11738 handle_response_invite: RTP re-inivte after T38 session not handled yet ! Scheduling destruction of SIP dialog 'e2b10000bdc50000@192.168.254.140' in 32000 ms (Method: ACK) [Nov 9 19:47:27] DEBUG[20548]: chan_sip.c:5608 reqprep: Strict routing enforced for session e2b10000bdc50000@192.168.254.140 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.254.140, port 56613 Transmitting (no NAT) to 192.168.254.140:56613: ACK sip:200@192.168.254.140:56613;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.254.96:5060;branch=z9hG4bK212a65d9;rport From: ;tag=as7d247b51 To: ;tag=91afffff57860000 Contact: localhost*CLI> Call-ID: e2b10000bdc50000@192.168.254.140 CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- localhost*CLI> <--- SIP read from 192.168.254.153:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.254.96:5060;branch=z9hG4bK39b363fd;rport From: "200" ;tag=as2cf96937 To: ;tag=29710000f1d10000 Call-ID: 41be8c9076a3761c784c4cd70685bc98@192.168.254.96 CSeq: 103 INVITE User-Agent: Grandstream HT496 1.0.3.44 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Supported: replaces Content-Length: 160 v=0 o=201 8000 8002 IN IP4 192.168.254.153 s=SIP Call c=IN IP4 192.168.254.153 t=0 0 m=audio 5004 RTP/AVP 8 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 <-------------> localhost*CLI> --- (12 headers 9 lines) --- Found RTP audio format 8 Peer audio RTP is at port 192.168.254.153:5004 [Nov 9 19:47:27] DEBUG[20548]: chan_sip.c:4889 process_sdp: Peer doesn't provide T.38 UDPTL Found description format PCMA for ID 8 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 192.168.254.153:5004 --- set_address_from_contact host '192.168.254.153' [Nov 9 19:47:27] DEBUG[20548]: chan_sip.c:5608 reqprep: Strict routing enforced for session 41be8c9076a3761c784c4cd70685bc98@192.168.254.96 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.254.153, port 5060 Transmitting (no NAT) to 192.168.254.153:5060: ACK sip:201@192.168.254.153 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.96:5060;branch=z9hG4bK3f499581;rport From: "200" ;tag=as2cf96937 To: ;tag=29710000f1d10000 Contact: Call-ID: 41be8c9076a3761c784c4cd70685bc98@192.168.254.96 CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Nov 9 19:47:27] DEBUG[20548]: chan_sip.c:5608 reqprep: Strict routing enforced for session 41be8c9076a3761c784c4cd70685bc98@192.168.254.96 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.254.153, port 5060 Audio is at 192.168.254.96 port 15584 Adding codec 0x8 (alaw) to SDP Reliably Transmitting (no NAT) to 192.168.254.153:5060: INVITE sip:201@192.168.254.153 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.96:5060;branch=z9hG4bK7f88e626;rport From: "200" ;tag=as2cf96937 To: ;tag=29710000f1d10000 Contact: Call-ID: 41be8c9076a3761c784c4cd70685bc98@192.168.254.96 CSeq: 104 INVITE User-Agent: Asterisk PBX 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: 188 v=0 o=root 6198 6201 IN IP4 192.168.254.140 s=session c=IN IP4 192.168.254.140 t=0 0 m=audio 22708 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- localhost*CLI> [Nov 9 19:47:27] DEBUG[6198]: chan_sip.c:5608 reqprep: Strict routing enforced for session e2b10000bdc50000@192.168.254.140 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.254.140, port 56613 Audio is at 192.168.254.96 port 11210 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.254.140:56613: INVITE sip:200@192.168.254.140:56613;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.254.96:5060;branch=z9hG4bK3a3760bf;rport From: ;tag=as7d247b51 To: ;tag=91afffff57860000 Contact: Call-ID: e2b10000bdc50000@192.168.254.140 CSeq: 104 INVITE User-Agent: Asterisk PBX 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: 243 v=0 o=root 6198 6201 IN IP4 192.168.254.153 s=session c=IN IP4 192.168.254.153 t=0 0 m=audio 5004 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- localhost*CLI> <--- SIP read from 192.168.254.153:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.254.96:5060;branch=z9hG4bK7f88e626;rport From: "200" ;tag=as2cf96937 To: ;tag=29710000f1d10000 Call-ID: 41be8c9076a3761c784c4cd70685bc98@192.168.254.96 CSeq: 104 INVITE User-Agent: Grandstream HT496 1.0.3.44 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Supported: replaces Content-Length: 160 v=0 o=201 8000 8003 IN IP4 192.168.254.153 s=SIP Call c=IN IP4 192.168.254.153 t=0 0 m=audio 5004 RTP/AVP 8 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 <-------------> --- (12 headers 9 lines) --- Found RTP audio format 8 Peer audio RTP is at port 192.168.254.153:5004 [Nov 9 19:47:28] DEBUG[20548]: chan_sip.c:4889 process_sdp: Peer doesn't provide T.38 UDPTL Found description format PCMA for ID 8 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 192.168.254.153:5004 --- set_address_from_contact host '192.168.254.153' [Nov 9 19:47:28] DEBUG[20548]: chan_sip.c:5608 reqprep: Strict routing enforced for session 41be8c9076a3761c784c4cd70685bc98@192.168.254.96 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.254.153, port 5060 Transmitting (no NAT) to 192.168.254.153:5060: ACK sip:201@192.168.254.153 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.96:5060;branch=z9hG4bK3ea2e1fe;rport From: "200" ;tag=as2cf96937 To: ;tag=29710000f1d10000 Contact: Call-ID: 41be8c9076a3761c784c4cd70685bc98@192.168.254.96 CSeq: 104 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- localhost*CLI> <--- SIP read from 192.168.254.140:56613 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.254.96:5060;branch=z9hG4bK3a3760bf;rport From: ;tag=as7d247b51 To: ;tag=91afffff57860000 Call-ID: e2b10000bdc50000@192.168.254.140 CSeq: 104 INVITE User-Agent: Grandstream HT496 1.0.3.44 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Supported: replaces Content-Length: 217 v=0 o=200 8000 8004 IN IP4 192.168.254.140 s=SIP Call c=IN IP4 192.168.254.140 t=0 0 m=audio 22708 RTP/AVP 8 101 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 <-------------> --- (12 headers 11 lines) --- Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.254.140:22708 [Nov 9 19:47:28] DEBUG[20548]: chan_sip.c:4889 process_sdp: Peer doesn't provide T.38 UDPTL Found description format PCMA for ID 8 Found description format telephone-event for ID 101 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.254.140:22708 --- set_address_from_contact host '192.168.254.140' [Nov 9 19:47:28] DEBUG[20548]: chan_sip.c:5608 reqprep: Strict routing enforced for session e2b10000bdc50000@192.168.254.140 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.254.140, port 56613 Transmitting (no NAT) to 192.168.254.140:56613: ACK sip:200@192.168.254.140:56613;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.254.96:5060;branch=z9hG4bK492cf0ea;rport From: ;tag=as7d247b51 To: ;tag=91afffff57860000 Contact: Call-ID: e2b10000bdc50000@192.168.254.140 CSeq: 104 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- localhost*CLI> show c Really destroying SIP dialog '3c90fa9d2f54c8796f77923a73ec0d7a@127.0.0.1' Method: REGISTER localhost*CLI> ssip show ch channels channel localhost*CLI> sip show channels localhost*CLI> Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message 192.168.254.153 (None) b3a40000039 00101/00175 unkn No Rx: REGISTER 192.168.254.153 201 41be8c9076a 00104/29765 alaw No Tx: ACK 192.168.254.140 200 e2b10000bdc 00104/19577 alaw No Tx: ACK 3 active SIP channels localhost*CLI> <--- SIP read from 192.168.254.153:5060 ---> <-------------> --- (0 headers 0 lines) Nat keepalive --- localhost*CLI> sip show channels localhost*CLI> Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message localhost*CLI> 192.168.254.153 (None) b3a40000039 00101/00175 unkn No Rx: REGISTER 192.168.254.153 201 41be8c9076a 00104/29765 alaw No Tx: ACK 192.168.254.140 200 e2b10000bdc 00104/19577 alaw No Tx: ACK 3 active SIP channels localhost*CLI> sip show channels localhost*CLI> Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message localhost*CLI> 192.168.254.153 (None) b3a40000039 00101/00175 unkn No Rx: REGISTER localhost*CLI> 192.168.254.153 201 41be8c9076a 00104/29765 alaw No Tx: ACK localhost*CLI> 192.168.254.140 200 e2b10000bdc 00104/19577 alaw No Tx: ACK 3 active SIP channels localhost*CLI> sip show channels localhost*CLI> Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message localhost*CLI> 192.168.254.153 (None) b3a40000039 00101/00175 unkn No Rx: REGISTER localhost*CLI> 192.168.254.153 201 41be8c9076a 00104/29765 alaw No Tx: ACK localhost*CLI> 192.168.254.140 200 e2b10000bdc 00104/19577 alaw No Tx: ACK localhost*CLI> 3 active SIP channels localhost*CLI> sip show channels localhost*CLI> Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message localhost*CLI> 192.168.254.153 (None) b3a40000039 00101/00175 unkn No Rx: REGISTER localhost*CLI> 192.168.254.153 201 41be8c9076a 00104/29765 alaw No Tx: ACK localhost*CLI> 192.168.254.140 200 e2b10000bdc 00104/19577 alaw No Tx: ACK localhost*CLI> 3 active SIP channels localhost*CLI> Really destroying SIP dialog 'b3a4000003970000@192.168.254.153' Method: REGISTER localhost*CLI> sip show channels localhost*CLI> Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message localhost*CLI> 192.168.254.153 201 41be8c9076a 00104/29765 alaw No Tx: ACK localhost*CLI> 192.168.254.140 200 e2b10000bdc 00104/19577 alaw No Tx: ACK localhost*CLI> 2 active SIP channels localhost*CLI> sip show channels localhost*CLI> Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message localhost*CLI> 192.168.254.153 201 41be8c9076a 00104/29765 alaw No Tx: ACK localhost*CLI> 192.168.254.140 200 e2b10000bdc 00104/19577 alaw No Tx: ACK localhost*CLI> 2 active SIP channels localhost*CLI> <--- SIP read from 192.168.254.153:5060 ---> <-------------> --- (0 headers 0 lines) Nat keepalive --- localhost*CLI> sip show channels localhost*CLI> Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message localhost*CLI> 192.168.254.153 201 41be8c9076a 00104/29765 alaw No Tx: ACK 192.168.254.140 200 e2b10000bdc 00104/19577 alaw No Tx: ACK 2 active SIP channels localhost*CLI> sip show channels localhost*CLI> Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message localhost*CLI> 192.168.254.153 201 41be8c9076a 00104/29765 alaw No Tx: ACK localhost*CLI> 192.168.254.140 200 e2b10000bdc 00104/19577 alaw No Tx: ACK 2 active SIP channels localhost*CLI> sip show channels localhost*CLI> Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message 192.168.254.153 201 41be8c9076a 00104/29765 alaw No Tx: ACK 192.168.254.140 200 e2b10000bdc 00104/19577 alaw No Tx: ACK 2 active SIP channels localhost*CLI> sip show channels localhost*CLI> Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message localhost*CLI> 192.168.254.153 201 41be8c9076a 00104/29765 alaw No Tx: ACK localhost*CLI> 192.168.254.140 200 e2b10000bdc 00104/19577 alaw No Tx: ACK localhost*CLI> 2 active SIP channels localhost*CLI> <--- SIP read from 192.168.254.153:5060 ---> BYE sip:200@192.168.254.96 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.153;branch=z9hG4bKbef0000034420000 From: ;tag=29710000f1d10000 To: "200" ;tag=as2cf96937 Call-ID: 41be8c9076a3761c784c4cd70685bc98@192.168.254.96 CSeq: 29766 BYE User-Agent: Grandstream HT496 1.0.3.44 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 192.168.254.153 : 5060 (no NAT) <--- Transmitting (no NAT) to 192.168.254.153:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.254.153;branch=z9hG4bKbef0000034420000;received=192.168.254.153 From: ;tag=29710000f1d10000 To: "200" ;tag=as2cf96937 Call-ID: 41be8c9076a3761c784c4cd70685bc98@192.168.254.96 CSeq: 29766 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces localhost*CLI> Contact: Content-Length: 0 <------------> localhost*CLI> [Nov 9 19:48:12] DEBUG[6198]: chan_sip.c:5608 reqprep: Strict routing enforced for session e2b10000bdc50000@192.168.254.140 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.254.140, port 56613 Audio is at 192.168.254.96 port 11210 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.254.140:56613: INVITE sip:200@192.168.254.140:56613;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.254.96:5060;branch=z9hG4bK69f3d376;rport From: ;tag=as7d247b51 To: ;tag=91afffff57860000 Contact: Call-ID: e2b10000bdc50000@192.168.254.140 CSeq: 105 INVITE User-Agent: Asterisk PBX 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: 242 v=0 o=root 6198 6202 IN IP4 192.168.254.96 s=session c=IN IP4 192.168.254.96 t=0 0 m=audio 11210 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- == Spawn extension (from-sip, 201, 1) exited non-zero on 'SIP/200-081c9198' Scheduling destruction of SIP dialog 'e2b10000bdc50000@192.168.254.140' in 32000 ms (Method: ACK) localhost*CLI> <--- SIP read from 192.168.254.140:56613 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.254.96:5060;branch=z9hG4bK69f3d376;rport From: ;tag=as7d247b51 To: ;tag=91afffff57860000 Call-ID: e2b10000bdc50000@192.168.254.140 CSeq: 105 INVITE User-Agent: Grandstream HT496 1.0.3.44 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Supported: replaces Content-Length: 217 v=0 o=200 8000 8005 IN IP4 192.168.254.140 s=SIP Call c=IN IP4 192.168.254.140 t=0 0 m=audio 22708 RTP/AVP 8 101 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 <-------------> --- (12 headers 11 lines) --- Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.254.140:22708 [Nov 9 19:48:12] DEBUG[20548]: chan_sip.c:4889 process_sdp: Peer doesn't provide T.38 UDPTL Found description format PCMA for ID 8 Found description format telephone-event for ID 101 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.254.140:22708 --- set_address_from_contact host '192.168.254.140' [Nov 9 19:48:12] DEBUG[20548]: chan_sip.c:5608 reqprep: Strict routing enforced for session e2b10000bdc50000@192.168.254.140 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.254.140, port 56613 Transmitting (no NAT) to 192.168.254.140:56613: ACK sip:200@192.168.254.140:56613;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.254.96:5060;branch=z9hG4bK33103bad;rport From: ;tag=as7d247b51 To: ;tag=91afffff57860000 Contact: Call-ID: e2b10000bdc50000@192.168.254.140 CSeq: 105 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Nov 9 19:48:12] DEBUG[20548]: chan_sip.c:5608 reqprep: Strict routing enforced for session e2b10000bdc50000@192.168.254.140 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.254.140, port 56613 Reliably Transmitting (no NAT) to 192.168.254.140:56613: BYE sip:200@192.168.254.140:56613;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.254.96:5060;branch=z9hG4bK2132f051;rport From: ;tag=as7d247b51 To: ;tag=91afffff57860000 Call-ID: e2b10000bdc50000@192.168.254.140 CSeq: 106 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of SIP dialog 'e2b10000bdc50000@192.168.254.140' in 32000 ms (Method: ACK) Really destroying SIP dialog '41be8c9076a3761c784c4cd70685bc98@192.168.254.96' Method: BYE localhost*CLI> <--- SIP read from 192.168.254.140:56613 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.254.96:5060;branch=z9hG4bK2132f051;rport From: ;tag=as7d247b51 To: ;tag=91afffff57860000 Call-ID: e2b10000bdc50000@192.168.254.140 CSeq: 106 BYE User-Agent: Grandstream HT496 1.0.3.44 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Supported: replaces Content-Length: 0 < localhost*CLI> -------------> --- (11 headers 0 lines) --- Really destroying SIP dialog 'e2b10000bdc50000@192.168.254.140' Method: ACK localhost*CLI> <--- SIP read from 192.168.254.153:5060 ---> <-------------> --- (0 headers 0 lines) Nat keepalive --- localhost*CLI> quit localhost bin # exit Script done on Thu Nov 9 19:48:38 2006