line*CLI> <--- SIP read from 192.168.7.246:62613 ---> INVITE sip:1001@asterisk.hostname.here SIP/2.0 Via: SIP/2.0/UDP 192.168.7.246:62613;branch=z9hG4bK-d87543-2c1c69283a15ed65-1--d87543-;rport Max-Forwards: 70 Contact: To: "1001" From: "1002";tag=6675631d Call-ID: OWNjMmRmOTY0NDc4MTYwOTBlZjAzNTViOGNlNTMxY2U. 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: 516 v=0 o=- 9 2 IN IP4 192.168.7.246 s=CounterPath eyeBeam 1.5 c=IN IP4 192.168.7.246 t=0 0 m=audio 1624 RTP/AVP 107 100 106 6 105 8 18 5 101 a=alt:1 2 : VsaJszjx XcK+RrjH 192.168.6.177 1624 a=alt:2 1 : NRPMEERM OISaAViW 192.168.7.246 1624 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:7406AF7E42AE4E5B9A7D17DBF1F0BD3F <-------------> --- (12 headers 18 lines) --- Sending to 192.168.7.246 : 62613 (NAT) Using INVITE request as basis request - OWNjMmRmOTY0NDc4MTYwOTBlZjAzNTViOGNlNTMxY2U. Found peer '1002' <--- Reliably Transmitting (no NAT) to 192.168.7.246:62613 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.7.246:62613;branch=z9hG4bK-d87543-2c1c69283a15ed65-1--d87543-;received=192.168.7.246;rport=62613 From: "1002";tag=6675631d To: "1001";tag=as24f46674 Call-ID: OWNjMmRmOTY0NDc4MTYwOTBlZjAzNTViOGNlNTMxY2U. 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="406746a2" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'OWNjMmRmOTY0NDc4MTYwOTBlZjAzNTViOGNlNTMxY2U.' in 6784 ms (Method: INVITE) line*CLI> <--- SIP read from 192.168.7.246:62613 ---> ACK sip:1001@asterisk.hostname.here SIP/2.0 Via: SIP/2.0/UDP 192.168.7.246:62613;branch=z9hG4bK-d87543-2c1c69283a15ed65-1--d87543-;rport To: "1001";tag=as24f46674 From: "1002";tag=6675631d Call-ID: OWNjMmRmOTY0NDc4MTYwOTBlZjAzNTViOGNlNTMxY2U. CSeq: 1 ACK Content-Length: 0 <-------------> --- (7 headers 0 lines) --- line*CLI> <--- SIP read from 192.168.7.246:62613 ---> INVITE sip:1001@asterisk.hostname.here SIP/2.0 Via: SIP/2.0/UDP 192.168.7.246:62613;branch=z9hG4bK-d87543-37795963513af960-1--d87543-;rport Max-Forwards: 70 Contact: To: "1001" From: "1002";tag=6675631d Call-ID: OWNjMmRmOTY0NDc4MTYwOTBlZjAzNTViOGNlNTMxY2U. 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="406746a2",uri="sip:1001@asterisk.hostname.here",response="edeffb7ccbbedde38d5e7da494fa89db",algorithm=MD5 User-Agent: eyeBeam release 1009l stamp 37965 Content-Length: 516 v=0 o=- 9 2 IN IP4 192.168.7.246 s=CounterPath eyeBeam 1.5 c=IN IP4 192.168.7.246 t=0 0 m=audio 1624 RTP/AVP 107 100 106 6 105 8 18 5 101 a=alt:1 2 : VsaJszjx XcK+RrjH 192.168.6.177 1624 a=alt:2 1 : NRPMEERM OISaAViW 192.168.7.246 1624 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:7406AF7E42AE4E5B9A7D17DBF1F0BD3F <-------------> --- (13 headers 18 lines) --- Sending to 192.168.7.246 : 62613 (NAT) Using INVITE request as basis request - OWNjMmRmOTY0NDc4MTYwOTBlZjAzNTViOGNlNTMxY2U. 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 192.168.7.246:1624 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 192.168.7.246:1624 Looking for 1001 in ael-default (domain asterisk.hostname.here) list_route: hop: <--- Transmitting (no NAT) to 192.168.7.246:62613 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.7.246:62613;branch=z9hG4bK-d87543-37795963513af960-1--d87543-;received=192.168.7.246;rport=62613 From: "1002";tag=6675631d To: "1001" Call-ID: OWNjMmRmOTY0NDc4MTYwOTBlZjAzNTViOGNlNTMxY2U. 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/1002-08d3baa0", "stdexten|1001|SIP/1001&IAX2/1001") in new stack -- Executing [s@macro-stdexten:1] Set("SIP/1002-08d3baa0", "ext=1001") in new stack -- Executing [s@macro-stdexten:2] Set("SIP/1002-08d3baa0", "dev=SIP/1001&IAX2/1001") in new stack -- Executing [s@macro-stdexten:3] GotoIf("SIP/1002-08d3baa0", "1?4:7") in new stack -- Goto (macro-stdexten,s,4) -- Executing [s@macro-stdexten:4] Set("SIP/1002-08d3baa0", "options=tT") in new stack -- Executing [s@macro-stdexten:5] Set("SIP/1002-08d3baa0", "features=nway-invite-both") in new stack -- Executing [s@macro-stdexten:6] Goto("SIP/1002-08d3baa0", "9") in new stack -- Goto (macro-stdexten,s,9) -- Executing [s@macro-stdexten:9] NoOp("SIP/1002-08d3baa0", "Finish if-stdexten-9") in new stack -- Executing [s@macro-stdexten:10] Set("SIP/1002-08d3baa0", "DYNAMIC_FEATURES=nway-invite-both") in new stack -- Executing [s@macro-stdexten:11] NoOp("SIP/1002-08d3baa0", "User Name") in new stack -- Executing [s@macro-stdexten:12] NoOp("SIP/1002-08d3baa0", "1002") in new stack -- Executing [s@macro-stdexten:13] NoOp("SIP/1002-08d3baa0", ""User Name" <1002>") in new stack -- Executing [s@macro-stdexten:14] Dial("SIP/1002-08d3baa0", "SIP/1001&IAX2/1001/1001|20|tT") in new stack Audio is at xx.xx.xx.xx port 17356 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 192.168.7.246:41923: INVITE sip:1001@192.168.7.246:41923;rinstance=719ef5653e4dd1f6 SIP/2.0 Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK74fa97ea;rport From: "User Name" ;tag=as47bc680c To: Contact: Call-ID: 255adb12661bf30b665a7cf325a08447@xx.xx.xx.xx CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 04 May 2007 13:53:04 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 xx.xx.xx.xx s=session c=IN IP4 xx.xx.xx.xx t=0 0 m=audio 17356 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 17:53:04] WARNING[4505]: 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 192.168.7.246:41923 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK74fa97ea;rport=5060 Contact: To: ;tag=4361de5d From: "User Name";tag=as47bc680c Call-ID: 255adb12661bf30b665a7cf325a08447@xx.xx.xx.xx CSeq: 102 INVITE User-Agent: X-Lite release 1009l stamp 38210 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- -- SIP/1001-08d2a2c0 is ringing <--- Transmitting (no NAT) to 192.168.7.246:62613 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.7.246:62613;branch=z9hG4bK-d87543-37795963513af960-1--d87543-;received=192.168.7.246;rport=62613 From: "1002";tag=6675631d To: "1001";tag=as72ac1dc2 Call-ID: OWNjMmRmOTY0NDc4MTYwOTBlZjAzNTViOGNlNTMxY2U. 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 xx.xx.xx.xx port 15270 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Transmitting (no NAT) to 192.168.7.246:62613 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.7.246:62613;branch=z9hG4bK-d87543-37795963513af960-1--d87543-;received=192.168.7.246;rport=62613 From: "1002";tag=6675631d To: "1001";tag=as72ac1dc2 Call-ID: OWNjMmRmOTY0NDc4MTYwOTBlZjAzNTViOGNlNTMxY2U. 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 xx.xx.xx.xx s=session c=IN IP4 xx.xx.xx.xx t=0 0 m=audio 15270 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 192.168.7.246:41923 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK74fa97ea;rport=5060 Contact: To: ;tag=4361de5d From: "User Name";tag=as47bc680c Call-ID: 255adb12661bf30b665a7cf325a08447@xx.xx.xx.xx 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=- 5 2 IN IP4 192.168.7.246 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.7.246 t=0 0 m=audio 14034 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 192.168.7.246:14034 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 192.168.7.246:14034 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.7.246, port 41923 Transmitting (no NAT) to 192.168.7.246:41923: ACK sip:1001@192.168.7.246:41923;rinstance=719ef5653e4dd1f6 SIP/2.0 Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK3d5070b0;rport From: "User Name" ;tag=as47bc680c To: ;tag=4361de5d Contact: Call-ID: 255adb12661bf30b665a7cf325a08447@xx.xx.xx.xx CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- Call on SIP/1001-08d2a2c0 left from hold -- SIP/1001-08d2a2c0 answered SIP/1002-08d3baa0 Audio is at xx.xx.xx.xx port 15270 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 192.168.7.246:62613 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.7.246:62613;branch=z9hG4bK-d87543-37795963513af960-1--d87543-;received=192.168.7.246;rport=62613 From: "1002";tag=6675631d To: "1001";tag=as72ac1dc2 Call-ID: OWNjMmRmOTY0NDc4MTYwOTBlZjAzNTViOGNlNTMxY2U. 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 xx.xx.xx.xx s=session c=IN IP4 xx.xx.xx.xx t=0 0 m=audio 15270 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 192.168.7.246:62613 ---> ACK sip:1001@xx.xx.xx.xx SIP/2.0 Via: SIP/2.0/UDP 192.168.7.246:62613;branch=z9hG4bK-d87543-4c0fd04aef726377-1--d87543-;rport Max-Forwards: 70 Contact: To: "1001";tag=as72ac1dc2 From: "1002";tag=6675631d Call-ID: OWNjMmRmOTY0NDc4MTYwOTBlZjAzNTViOGNlNTMxY2U. CSeq: 2 ACK Proxy-Authorization: Digest username="1002",realm="asterisk",nonce="406746a2",uri="sip:1001@asterisk.hostname.here",response="edeffb7ccbbedde38d5e7da494fa89db",algorithm=MD5 User-Agent: eyeBeam release 1009l stamp 37965 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- -- Started music on hold, class 'default', on SIP/1002-08d3baa0 -- Playing 'pbx-transfer' (language 'en') Reliably Transmitting (no NAT) to 192.168.10.131:5060: OPTIONS sip:1010@192.168.10.131:5060;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK59924371;rport From: "asterisk" ;tag=as02ae67c1 To: Contact: Call-ID: 058799531a79d2e9707cc2201ac46345@xx.xx.xx.xx CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 04 May 2007 13:53:09 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- line*CLI> <--- SIP read from 192.168.10.131:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK59924371;rport From: "asterisk" ;tag=as02ae67c1 To: ;tag=156948524 Call-ID: 058799531a79d2e9707cc2201ac46345@xx.xx.xx.xx CSeq: 102 OPTIONS Server: Cisco-CP7912/8.0.1-060412A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces Content-Length: 285 Content-Type: application/sdp v=0 o=1010 257294 257294 IN IP4 192.168.10.131 s=Cisco 7912 SIP Call c=IN IP4 192.168.10.131 t=0 0 m=audio 16384 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:18 G729/8000/1 a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (11 headers 12 lines) --- Really destroying SIP dialog '058799531a79d2e9707cc2201ac46345@xx.xx.xx.xx' Method: OPTIONS line*CLI> <--- SIP read from 192.168.7.246:5060 ---> <-------------> --- (0 headers 1 lines) --- -- Executing [1011@ael-default:1] Macro("Local/1011@ael-default-f3db,2", "stdexten|1011|SIP/1011&IAX2/1011") in new stack -- Executing [s@macro-stdexten:1] Set("Local/1011@ael-default-f3db,2", "ext=1011") in new stack -- Executing [s@macro-stdexten:2] Set("Local/1011@ael-default-f3db,2", "dev=SIP/1011&IAX2/1011") in new stack -- Executing [s@macro-stdexten:3] GotoIf("Local/1011@ael-default-f3db,2", "1?4:7") in new stack -- Goto (macro-stdexten,s,4) -- Executing [s@macro-stdexten:4] Set("Local/1011@ael-default-f3db,2", "options=tT") in new stack -- Executing [s@macro-stdexten:5] Set("Local/1011@ael-default-f3db,2", "features=nway-invite-both") in new stack -- Executing [s@macro-stdexten:6] Goto("Local/1011@ael-default-f3db,2", "9") in new stack -- Goto (macro-stdexten,s,9) -- Executing [s@macro-stdexten:9] NoOp("Local/1011@ael-default-f3db,2", "Finish if-stdexten-9") in new stack -- Executing [s@macro-stdexten:10] Set("Local/1011@ael-default-f3db,2", "DYNAMIC_FEATURES=nway-invite-both") in new stack -- Executing [s@macro-stdexten:11] NoOp("Local/1011@ael-default-f3db,2", "Ï Hÿÿÿÿ€€å¾ HÀ¤(H") in new stack -- Executing [s@macro-stdexten:12] NoOp("Local/1011@ael-default-f3db,2", "1001") in new stack -- Executing [s@macro-stdexten:13] NoOp("Local/1011@ael-default-f3db,2", ""Ï Hÿÿÿÿ€€å¾ HÀ¤(H" <1001>") in new stack -- Executing [s@macro-stdexten:14] Dial("Local/1011@ael-default-f3db,2", "SIP/1011&IAX2/1011/1011|20|tT") in new stack Audio is at xx.xx.xx.xx port 12332 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 xx.xx.xx.xx:5060;branch=z9hG4bK194f4e45;rport From: "Ï Hÿÿÿÿ€€å¾ HÀ¤(H" ;tag=as2c5849e0 To: Contact: Call-ID: 75a0d7ae4f4b84472c72b7ee4de4c376@xx.xx.xx.xx CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 04 May 2007 13:53:13 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 xx.xx.xx.xx s=session c=IN IP4 xx.xx.xx.xx t=0 0 m=audio 12332 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 17:53:13] WARNING[4510]: 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 xx.xx.xx.xx:5060;branch=z9hG4bK194f4e45;rport=5060 Contact: To: ;tag=a6364e23 From: "Ï Hÿÿÿÿ€€å¾ HÀ¤(H";tag=as2c5849e0 Call-ID: 75a0d7ae4f4b84472c72b7ee4de4c376@xx.xx.xx.xx CSeq: 102 INVITE User-Agent: X-Lite release 1009l stamp 38210 Content-Length: 0