line*CLI> core set debug 10 Core debug was 0 and is now 10 line*CLI> core set verbose 10 Verbosity was 4 and is now 10 Really destroying SIP dialog '765815973c3b1aac48b4bee057dc9e72@xxx.xxx.xxx.xxx' Method: OPTIONS Really destroying SIP dialog '10eeb0210efaff6d1c92ab8d469e848b@xxx.xxx.xxx.xxx' Method: OPTIONS line*CLI> sip set debug SIP Debugging enabled line*CLI> <--- SIP read from 192.168.7.239:4638 ---> INVITE sip:1001@asterisk.hostname.here SIP/2.0 Via: SIP/2.0/UDP 192.168.7.239:4638;branch=z9hG4bK-d87543-695cd0339542bd6f-1--d87543-;rport Max-Forwards: 70 Contact: To: "1001" From: "1002";tag=a75afd2b Call-ID: Yjc0ZDAxMTgzNGM1YzM4OTFiYjQwOWY4MGQ3M2Q2OWI. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 1009l stamp 37965 Content-Length: 515 v=0 o=- 1 2 IN IP4 192.168.7.239 s=CounterPath eyeBeam 1.5 c=IN IP4 192.168.7.239 t=0 0 m=audio 30308 RTP/AVP 107 100 106 6 105 8 18 5 101 a=alt:1 2 : cqJDC8BW XJxquW+v 10.0.0.42 30308 a=alt:2 1 : QDjQXTgW RWsUBOKb 192.168.7.239 30308 a=fmtp:18 annexb=yes a=fmtp:101 0-15 a=rtpmap:107 BV32/16000 a=rtpmap:100 SPEEX/16000 a=rtpmap:106 SPEEX-FEC/16000 a=rtpmap:105 SPEEX-FEC/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv a=x-rtp-session-id:550E08BEDE3C4876B867C65B725FCB04 <-------------> --- (12 headers 18 lines) --- Sending to 192.168.7.239 : 4638 (NAT) Using INVITE request as basis request - Yjc0ZDAxMTgzNGM1YzM4OTFiYjQwOWY4MGQ3M2Q2OWI. Found peer '1002' <--- Reliably Transmitting (no NAT) to 192.168.7.239:4638 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.7.239:4638;branch=z9hG4bK-d87543-695cd0339542bd6f-1--d87543-;received=192.168.7.239;rport=4638 From: "1002";tag=a75afd2b To: "1001";tag=as7fe66242 Call-ID: Yjc0ZDAxMTgzNGM1YzM4OTFiYjQwOWY4MGQ3M2Q2OWI. CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="46602511" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'Yjc0ZDAxMTgzNGM1YzM4OTFiYjQwOWY4MGQ3M2Q2OWI.' in 11008 ms (Method: INVITE) line*CLI> <--- SIP read from 192.168.7.239:4638 ---> ACK sip:1001@asterisk.hostname.here SIP/2.0 Via: SIP/2.0/UDP 192.168.7.239:4638;branch=z9hG4bK-d87543-695cd0339542bd6f-1--d87543-;rport To: "1001";tag=as7fe66242 From: "1002";tag=a75afd2b Call-ID: Yjc0ZDAxMTgzNGM1YzM4OTFiYjQwOWY4MGQ3M2Q2OWI. CSeq: 1 ACK Content-Length: 0 <-------------> --- (7 headers 0 lines) --- line*CLI> <--- SIP read from 192.168.7.239:4638 ---> INVITE sip:1001@asterisk.hostname.here SIP/2.0 Via: SIP/2.0/UDP 192.168.7.239:4638;branch=z9hG4bK-d87543-d318a91acd57d351-1--d87543-;rport Max-Forwards: 70 Contact: To: "1001" From: "1002";tag=a75afd2b Call-ID: Yjc0ZDAxMTgzNGM1YzM4OTFiYjQwOWY4MGQ3M2Q2OWI. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Proxy-Authorization: Digest username="1002",realm="asterisk",nonce="46602511",uri="sip:1001@asterisk.hostname.here",response="c4e4aaec9c69932f79d0b63e9e69cd85",algorithm=MD5 User-Agent: eyeBeam release 1009l stamp 37965 Content-Length: 515 v=0 o=- 1 2 IN IP4 192.168.7.239 s=CounterPath eyeBeam 1.5 c=IN IP4 192.168.7.239 t=0 0 m=audio 30308 RTP/AVP 107 100 106 6 105 8 18 5 101 a=alt:1 2 : cqJDC8BW XJxquW+v 10.0.0.42 30308 a=alt:2 1 : QDjQXTgW RWsUBOKb 192.168.7.239 30308 a=fmtp:18 annexb=yes a=fmtp:101 0-15 a=rtpmap:107 BV32/16000 a=rtpmap:100 SPEEX/16000 a=rtpmap:106 SPEEX-FEC/16000 a=rtpmap:105 SPEEX-FEC/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv a=x-rtp-session-id:550E08BEDE3C4876B867C65B725FCB04 <-------------> --- (13 headers 18 lines) --- Sending to 192.168.7.239 : 4638 (NAT) Using INVITE request as basis request - Yjc0ZDAxMTgzNGM1YzM4OTFiYjQwOWY4MGQ3M2Q2OWI. Found peer '1002' Found RTP audio format 107 Found RTP audio format 100 Found RTP audio format 106 Found RTP audio format 6 Found RTP audio format 105 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 5 Found RTP audio format 101 Peer audio RTP is at port 192.168.7.239:30308 Got unsupported a:fmtp in SDP offer Got unsupported a:fmtp in SDP offer Found description format BV32 for ID 107 Found description format SPEEX for ID 100 Found description format SPEEX-FEC for ID 106 Found description format SPEEX-FEC for ID 105 Found description format G729 for ID 18 Found description format telephone-event for ID 101 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x328 (alaw|adpcm|g729|speex)/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 192.168.7.239:30308 Looking for 1001 in ael-default (domain asterisk.hostname.here) list_route: hop: <--- Transmitting (no NAT) to 192.168.7.239:4638 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.7.239:4638;branch=z9hG4bK-d87543-d318a91acd57d351-1--d87543-;received=192.168.7.239;rport=4638 From: "1002";tag=a75afd2b To: "1001" Call-ID: Yjc0ZDAxMTgzNGM1YzM4OTFiYjQwOWY4MGQ3M2Q2OWI. CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Executing [1001@ael-default:1] Macro("SIP/1002-08d1dc10", "stdexten|1001|SIP/1001&IAX2/1001") in new stack -- Executing [s@macro-stdexten:1] Set("SIP/1002-08d1dc10", "ext=1001") in new stack -- Executing [s@macro-stdexten:2] Set("SIP/1002-08d1dc10", "dev=SIP/1001&IAX2/1001") in new stack -- Executing [s@macro-stdexten:3] GotoIf("SIP/1002-08d1dc10", "1?4:7") in new stack -- Goto (macro-stdexten,s,4) -- Executing [s@macro-stdexten:4] Set("SIP/1002-08d1dc10", "options=tT") in new stack -- Executing [s@macro-stdexten:5] Set("SIP/1002-08d1dc10", "features=nway-invite-both") in new stack -- Executing [s@macro-stdexten:6] Goto("SIP/1002-08d1dc10", "9") in new stack -- Goto (macro-stdexten,s,9) -- Executing [s@macro-stdexten:9] NoOp("SIP/1002-08d1dc10", "Finish if-stdexten-1") in new stack -- Executing [s@macro-stdexten:10] Set("SIP/1002-08d1dc10", "DYNAMIC_FEATURES=nway-invite-both") in new stack -- Executing [s@macro-stdexten:11] Dial("SIP/1002-08d1dc10", "SIP/1001&IAX2/1001/1001|20|tT") in new stack Audio is at xxx.xxx.xxx.xxx port 16022 Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.7.239:59486: INVITE sip:1001@192.168.7.239:59486;rinstance=15035a15c6cd56e7 SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK7744d618;rport From: "Dmitry Andrianov" ;tag=as099e0b96 To: Contact: Call-ID: 45a31e2351b5e1b9378fd35737c8d2c0@xxx.xxx.xxx.xxx CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 03 May 2007 21:47:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 262 v=0 o=root 31890 31890 IN IP4 xxx.xxx.xxx.xxx s=session c=IN IP4 xxx.xxx.xxx.xxx t=0 0 m=audio 16022 RTP/AVP 8 3 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- -- Called 1001 [May 4 01:47:47] WARNING[1400]: app_dial.c:1099 dial_exec_full: Unable to create channel of type 'IAX2' (cause 3 - No route to destination) line*CLI> <--- SIP read from 192.168.7.239:59486 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK7744d618;rport=5060 Contact: To: ;tag=c3270152 From: "Dmitry Andrianov";tag=as099e0b96 Call-ID: 45a31e2351b5e1b9378fd35737c8d2c0@xxx.xxx.xxx.xxx CSeq: 102 INVITE User-Agent: X-Lite release 1009l stamp 38210 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- -- SIP/1001-08d1c508 is ringing <--- Transmitting (no NAT) to 192.168.7.239:4638 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.7.239:4638;branch=z9hG4bK-d87543-d318a91acd57d351-1--d87543-;received=192.168.7.239;rport=4638 From: "1002";tag=a75afd2b To: "1001";tag=as0687a791 Call-ID: Yjc0ZDAxMTgzNGM1YzM4OTFiYjQwOWY4MGQ3M2Q2OWI. CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> Audio is at xxx.xxx.xxx.xxx port 16380 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Transmitting (no NAT) to 192.168.7.239:4638 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.7.239:4638;branch=z9hG4bK-d87543-d318a91acd57d351-1--d87543-;received=192.168.7.239;rport=4638 From: "1002";tag=a75afd2b To: "1001";tag=as0687a791 Call-ID: Yjc0ZDAxMTgzNGM1YzM4OTFiYjQwOWY4MGQ3M2Q2OWI. CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 215 v=0 o=root 31890 31890 IN IP4 xxx.xxx.xxx.xxx s=session c=IN IP4 xxx.xxx.xxx.xxx t=0 0 m=audio 16380 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> line*CLI> <--- SIP read from 192.168.7.239:59486 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK7744d618;rport=5060 Contact: To: ;tag=c3270152 From: "Dmitry Andrianov";tag=as099e0b96 Call-ID: 45a31e2351b5e1b9378fd35737c8d2c0@xxx.xxx.xxx.xxx CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1009l stamp 38210 Content-Length: 189 v=0 o=- 3 2 IN IP4 192.168.7.239 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.7.239 t=0 0 m=audio 60874 RTP/AVP 8 3 0 101 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv <-------------> --- (11 headers 9 lines) --- Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.7.239:60874 Got unsupported a:fmtp in SDP offer Found description format telephone-event for ID 101 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.7.239:60874 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.7.239, port 59486 Transmitting (no NAT) to 192.168.7.239:59486: ACK sip:1001@192.168.7.239:59486;rinstance=15035a15c6cd56e7 SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK275cfbcb;rport From: "Dmitry Andrianov" ;tag=as099e0b96 To: ;tag=c3270152 Contact: Call-ID: 45a31e2351b5e1b9378fd35737c8d2c0@xxx.xxx.xxx.xxx CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- Call on SIP/1001-08d1c508 left from hold -- SIP/1001-08d1c508 answered SIP/1002-08d1dc10 Audio is at xxx.xxx.xxx.xxx port 16380 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 192.168.7.239:4638 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.7.239:4638;branch=z9hG4bK-d87543-d318a91acd57d351-1--d87543-;received=192.168.7.239;rport=4638 From: "1002";tag=a75afd2b To: "1001";tag=as0687a791 Call-ID: Yjc0ZDAxMTgzNGM1YzM4OTFiYjQwOWY4MGQ3M2Q2OWI. CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 215 v=0 o=root 31890 31891 IN IP4 xxx.xxx.xxx.xxx s=session c=IN IP4 xxx.xxx.xxx.xxx t=0 0 m=audio 16380 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> line*CLI> <--- SIP read from 192.168.7.239:4638 ---> ACK sip:1001@xxx.xxx.xxx.xxx SIP/2.0 Via: SIP/2.0/UDP 192.168.7.239:4638;branch=z9hG4bK-d87543-9267db02a0307678-1--d87543-;rport Max-Forwards: 70 Contact: To: "1001";tag=as0687a791 From: "1002";tag=a75afd2b Call-ID: Yjc0ZDAxMTgzNGM1YzM4OTFiYjQwOWY4MGQ3M2Q2OWI. CSeq: 2 ACK Proxy-Authorization: Digest username="1002",realm="asterisk",nonce="46602511",uri="sip:1001@asterisk.hostname.here",response="c4e4aaec9c69932f79d0b63e9e69cd85",algorithm=MD5 User-Agent: eyeBeam release 1009l stamp 37965 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- line*CLI> <--- SIP read from 192.168.7.239:59486 ---> <-------------> --- (0 headers 1 lines) --- line*CLI> <--- SIP read from 192.168.10.202:40918 ---> <-------------> --- (0 headers 1 lines) --- -- Started music on hold, class 'default', on SIP/1002-08d1dc10 -- Playing 'pbx-transfer' (language 'en') line*CLI> <--- SIP read from 192.168.7.239:5060 ---> <-------------> --- (0 headers 1 lines) --- line*CLI> <--- SIP read from 192.168.7.239:4638 ---> <-------------> --- (0 headers 1 lines) --- -- Executing [1011@ael-default:1] Macro("Local/1011@ael-default-e51c,2", "stdexten|1011|SIP/1011&IAX2/1011") in new stack -- Executing [s@macro-stdexten:1] Set("Local/1011@ael-default-e51c,2", "ext=1011") in new stack -- Executing [s@macro-stdexten:2] Set("Local/1011@ael-default-e51c,2", "dev=SIP/1011&IAX2/1011") in new stack -- Executing [s@macro-stdexten:3] GotoIf("Local/1011@ael-default-e51c,2", "1?4:7") in new stack -- Goto (macro-stdexten,s,4) -- Executing [s@macro-stdexten:4] Set("Local/1011@ael-default-e51c,2", "options=tT") in new stack -- Executing [s@macro-stdexten:5] Set("Local/1011@ael-default-e51c,2", "features=nway-invite-both") in new stack -- Executing [s@macro-stdexten:6] Goto("Local/1011@ael-default-e51c,2", "9") in new stack -- Goto (macro-stdexten,s,9) -- Executing [s@macro-stdexten:9] NoOp("Local/1011@ael-default-e51c,2", "Finish if-stdexten-1") in new stack -- Executing [s@macro-stdexten:10] Set("Local/1011@ael-default-e51c,2", "DYNAMIC_FEATURES=nway-invite-both") in new stack -- Executing [s@macro-stdexten:11] Dial("Local/1011@ael-default-e51c,2", "SIP/1011&IAX2/1011/1011|20|tT") in new stack Audio is at xxx.xxx.xxx.xxx port 10772 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.10.202:40918: INVITE sip:1011@192.168.10.202:40918;rinstance=2d9231dc606dfcd4 SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK5daf2254;rport From: "Ï Hÿÿÿÿ€3¾ HÀ¤(H" ;tag=as245bf1fc To: Contact: Call-ID: 688babf20919afd11e77196d5a5d12c8@xxx.xxx.xxx.xxx CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 03 May 2007 21:47:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 262 v=0 o=root 31890 31890 IN IP4 xxx.xxx.xxx.xxx s=session c=IN IP4 xxx.xxx.xxx.xxx t=0 0 m=audio 10772 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- -- Called 1011 [May 4 01:47:57] WARNING[1406]: app_dial.c:1099 dial_exec_full: Unable to create channel of type 'IAX2' (cause 3 - No route to destination) line*CLI> <--- SIP read from 192.168.10.202:40918 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK5daf2254;rport=5060 Contact: To: ;tag=03564d52 From: "Ï Hÿÿÿÿ€3¾ HÀ¤(H";tag=as245bf1fc Call-ID: 688babf20919afd11e77196d5a5d12c8@xxx.xxx.xxx.xxx CSeq: 102 INVITE User-Agent: X-Lite release 1009l stamp 38210 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- -- SIP/1011-08d39570 is ringing -- Local/1011@ael-default-e51c,1 is ringing [May 4 01:47:59] NOTICE[1405]: rtp.c:1256 ast_rtp_read: Unknown RTP codec 126 received from '192.168.7.239' line*CLI> <--- SIP read from 192.168.11.1:61994 ---> <-------------> --- (0 headers 1 lines) --- line*CLI>