dev-pbx*CLI> sip debug SIP Debugging enabled dev-pbx*CLI> <-- SIP read from 10.10.15.177:5060: INVITE sip:101@10.10.15.192 SIP/2.0 Via: SIP/2.0/UDP 10.10.15.177:5060;branch=z9hG4bK-95a923eb From: ;tag=89eddad3704fd6ado0 To: "Test 7960" Call-ID: 413c524b-ed6b0595@10.10.15.177 CSeq: 101 INVITE Max-Forwards: 70 Contact: Expires: 240 User-Agent: Sipura/SPA942-4.1.10(e) Content-Length: 207 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Content-Type: application/sdp dev-pbx*CLI> v=0 o=- 30949 30949 IN IP4 10.10.15.177 s=- c=IN IP4 10.10.15.177 t=0 0 m=audio 16476 RTP/AVP 2 101 a=rtpmap:2 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv --- (13 headers 11 lines)--- Sending to 10.10.15.177 : 5060 (no NAT) Using INVITE request as basis request - 413c524b-ed6b0595@10.10.15.177 Reliably Transmitting (no NAT) to 10.10.15.177:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.10.15.177:5060;branch=z9hG4bK-95a923eb;received=10.10.15.177 From: ;tag=89eddad3704fd6ado0 To: "Test 7960" ;tag=as50599579 Call-ID: 413c524b-ed6b0595@10.10.15.177 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6947db44" Content-Length: 0 --- Scheduling destruction of SIP dialog '413c524b-ed6b0595@10.10.15.177' in 32000 ms (Method: INVITE) Found user '300' dev-pbx*CLI> <-- SIP read from 10.10.15.177:5060: ACK sip:101@10.10.15.192 SIP/2.0 Via: SIP/2.0/UDP 10.10.15.177:5060;branch=z9hG4bK-95a923eb From: ;tag=89eddad3704fd6ado0 To: "Test 7960" ;tag=as50599579 Call-ID: 413c524b-ed6b0595@10.10.15.177 CSeq: 101 ACK Max-Forwards: 70 Contact: User-Agent: Sipura/SPA942-4.1.10(e) Content-Length: 0 --- (10 headers 0 lines)--- dev-pbx*CLI> <-- SIP read from 10.10.15.177:5060: INVITE sip:101@10.10.15.192 SIP/2.0 Via: SIP/2.0/UDP 10.10.15.177:5060;branch=z9hG4bK-b8456d5b From: ;tag=89eddad3704fd6ado0 To: "Test 7960" Call-ID: 413c524b-ed6b0595@10.10.15.177 CSeq: 102 INVITE Max-Forwards: 70 Proxy-Authorization: Digest username="300",realm="asterisk",nonce="6947db44",uri="sip:101@10.10.15.192",algorithm=MD5,response="295464a8a4dee5190e729003eafbfbda" Contact: Expires: 240 User-Agent: Sipura/SPA942-4.1.10(e) Content-Length: 207 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Content-Type: application/sdp v=0 o=- 30949 30949 IN IP4 10.10.15.177 s=- c=IN IP4 10.10.15.177 t=0 0 m=audio 16476 RTP/AVP 2 101 a=rtpmap:2 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv --- (14 headers 11 lines)--- Sending to 10.10.15.177 : 5060 (no NAT) Using INVITE request as basis request - 413c524b-ed6b0595@10.10.15.177 Found user '300' Found RTP audio format 2 Found RTP audio format 101 Peer audio RTP is at port 10.10.15.177:16476 Found description format G726-32 for ID 2 Found description format telephone-event for ID 101 Capabilities: us - 0x810 (g726|g726aal2), peer - audio=0x800 (g726)/video=0x0 (nothing), combined - 0x800 (g726) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.10.15.177:16476 Looking for 101 in international (domain 10.10.15.192) list_route: hop: Transmitting (no NAT) to 10.10.15.177:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.15.177:5060;branch=z9hG4bK-b8456d5b;received=10.10.15.177 From: ;tag=89eddad3704fd6ado0 To: "Test 7960" Call-ID: 413c524b-ed6b0595@10.10.15.177 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 dev-pbx*CLI> --- -- Executing Macro("SIP/300-00701c10", "stdexten|101|SIP/101") -- Executing Dial("SIP/300-00701c10", "SIP/101|20") Audio is at 10.10.15.192 port 13756 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.10.15.180:5060: INVITE sip:101@10.10.15.180:5060;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.10.15.192:5060;branch=z9hG4bK14c917e4;rport From: "300" ;tag=as2c851e78 To: Contact: Call-ID: 72764e9f31ecf4553376dbad7e8e301f@10.10.15.192 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 06 Oct 2006 22:03:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 240 v=0 o=root 28500 28500 IN IP4 10.10.15.192 s=session c=IN IP4 10.10.15.192 t=0 0 m=audio 13756 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=sendrecv --- -- Called 101 dev-pbx*CLI> <-- SIP read from 10.10.15.180:52369: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.15.192:5060;branch=z9hG4bK14c917e4;rport From: "300" ;tag=as2c851e78 To: Call-ID: 72764e9f31ecf4553376dbad7e8e301f@10.10.15.192 Date: Fri, 06 Oct 2006 22:03:02 GMT CSeq: 102 INVITE Server: Cisco-CP7960G/8.0 Contact: Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE Remote-Party-ID: "Line 1" ;party=called;id-type=subscriber;privacy=off;screen=yes Content-Length: 0 --- (12 headers 0 lines)--- dev-pbx*CLI> <-- SIP read from 10.10.15.180:52370: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.10.15.192:5060;branch=z9hG4bK14c917e4;rport From: "300" ;tag=as2c851e78 To: ;tag=001192cd77a90deb3b63cd08-4f5907af Call-ID: 72764e9f31ecf4553376dbad7e8e301f@10.10.15.192 Date: Fri, 06 Oct 2006 22:03:02 GMT CSeq: 102 INVITE Server: Cisco-CP7960G/8.0 Contact: Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE Remote-Party-ID: "Line 1" ;party=called;id-type=subscriber;privacy=off;screen=yes Content-Length: 0 --- (12 headers 0 lines)--- -- SIP/101-00705f00 is ringing Transmitting (no NAT) to 10.10.15.177:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.10.15.177:5060;branch=z9hG4bK-b8456d5b;received=10.10.15.177 From: ;tag=89eddad3704fd6ado0 To: "Test 7960" ;tag=as57c414f4 Call-ID: 413c524b-ed6b0595@10.10.15.177 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 --- dev-pbx*CLI> <-- SIP read from 10.10.15.180:52371: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.15.192:5060;branch=z9hG4bK14c917e4;rport From: "300" ;tag=as2c851e78 To: ;tag=001192cd77a90deb3b63cd08-4f5907af Call-ID: 72764e9f31ecf4553376dbad7e8e301f@10.10.15.192 Date: Fri, 06 Oct 2006 22:03:03 GMT CSeq: 102 INVITE Server: Cisco-CP7960G/8.0 Contact: Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE Remote-Party-ID: "Line 1" ;party=called;id-type=subscriber;privacy=off;screen=yes Supported: replaces,join,norefersub Content-Length: 205 Content-Type: application/sdp Content-Disposition: session;handling=optional v=0 o=Cisco-SIPUA 22456 0 IN IP4 10.10.15.180 s=SIP Call t=0 0 m=audio 22390 RTP/AVP 0 101 c=IN IP4 10.10.15.180 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv --- (15 headers 10 lines)--- [Oct 6 17:03:03] DEBUG[3619]: chan_sip.c:2009 __sip_ack: Acked pending invite 102 Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 10.10.15.180:22390 Found description format PCMU for ID 0 Found description format telephone-event for ID 101 Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.10.15.180:22390 list_route: hop: [Oct 6 17:03:03] DEBUG[3619]: chan_sip.c:5420 reqprep: Strict routing enforced for session 72764e9f31ecf4553376dbad7e8e301f@10.10.15.192 set_destination: Parsing for address/port to send to set_destination: set destination to 10.10.15.180, port 5060 Transmitting (no NAT) to 10.10.15.180:5060: ACK sip:101@10.10.15.180:5060;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.10.15.192:5060;branch=z9hG4bK7a9af289;rport From: "300" ;tag=as2c851e78 To: ;tag=001192cd77a90deb3b63cd08-4f5907af Contact: Call-ID: 72764e9f31ecf4553376dbad7e8e301f@10.10.15.192 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/101-00705f00 answered SIP/300-00701c10 Audio is at 10.10.15.192 port 12584 Adding codec 0x800 (g726) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.10.15.177:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.15.177:5060;branch=z9hG4bK-b8456d5b;received=10.10.15.177 From: ;tag=89eddad3704fd6ado0 To: "Test 7960" ;tag=as57c414f4 Call-ID: 413c524b-ed6b0595@10.10.15.177 CSeq: 102 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 28500 28500 IN IP4 10.10.15.192 s=session c=IN IP4 10.10.15.192 t=0 0 m=audio 12584 RTP/AVP 2 101 a=rtpmap:2 G726-32/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=sendrecv --- dev-pbx*CLI> <-- SIP read from 10.10.15.177:5060: ACK sip:101@10.10.15.192 SIP/2.0 Via: SIP/2.0/UDP 10.10.15.177:5060;branch=z9hG4bK-8db6267a From: ;tag=89eddad3704fd6ado0 To: "Test 7960" ;tag=as57c414f4 Call-ID: 413c524b-ed6b0595@10.10.15.177 CSeq: 102 ACK Max-Forwards: 70 Proxy-Authorization: Digest username="300",realm="asterisk",nonce="6947db44",uri="sip:101@10.10.15.192",algorithm=MD5,response="8c51f4be662a19b969b7822e7707dba5" Contact: User-Agent: Sipura/SPA942-4.1.10(e) Content-Length: 0 --- (11 headers 0 lines)--- dev-pbx*CLI> <-- SIP read from 10.10.15.180:52372: BYE sip:300@10.10.15.192:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.15.180:5060;branch=z9hG4bK3659b437 From: ;tag=001192cd77a90deb3b63cd08-4f5907af To: "300" ;tag=as2c851e78 Call-ID: 72764e9f31ecf4553376dbad7e8e301f@10.10.15.192 Max-Forwards: 70 Date: Fri, 06 Oct 2006 22:03:06 GMT CSeq: 101 BYE User-Agent: Cisco-CP7960G/8.0 Content-Length: 0 --- (10 headers 0 lines)--- Sending to 10.10.15.180 : 5060 (no NAT) Transmitting (no NAT) to 10.10.15.180:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.15.180:5060;branch=z9hG4bK3659b437;received=10.10.15.180 From: ;tag=001192cd77a90deb3b63cd08-4f5907af To: "300" ;tag=as2c851e78 Call-ID: 72764e9f31ecf4553376dbad7e8e301f@10.10.15.192 CSeq: 101 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 --- [Oct 6 17:03:06] DEBUG[3646]: channel.c:3610 ast_generic_bridge: Didn't get a frame from channel: SIP/101-00705f00 [Oct 6 17:03:06] DEBUG[3646]: channel.c:3909 ast_channel_bridge: Bridge stops bridging channels SIP/300-00701c10 and SIP/101-00705f00 == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'SIP/300-00701c10' in macro 'stdexten' == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'SIP/300-00701c10' [Oct 6 17:03:06] DEBUG[3646]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '300' [Oct 6 17:03:06] DEBUG[3646]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '300' [Oct 6 17:03:06] DEBUG[3646]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '101' [Oct 6 17:03:06] DEBUG[3646]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is 'international' [Oct 6 17:03:06] DEBUG[3646]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is 'SIP/300-00701c10' [Oct 6 17:03:06] DEBUG[3646]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is 'SIP/101-00705f00' [Oct 6 17:03:06] DEBUG[3646]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is 'Dial' [Oct 6 17:03:06] DEBUG[3646]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is 'SIP/101|20' [Oct 6 17:03:06] DEBUG[3646]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '2006-10-06 17:03:02' [Oct 6 17:03:06] DEBUG[3646]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '2006-10-06 17:03:03' [Oct 6 17:03:06] DEBUG[3646]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '2006-10-06 17:03:06' [Oct 6 17:03:06] DEBUG[3646]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '4' [Oct 6 17:03:06] DEBUG[3646]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '3' [Oct 6 17:03:06] DEBUG[3646]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is 'ANSWERED' [Oct 6 17:03:06] DEBUG[3646]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is 'DOCUMENTATION' [Oct 6 17:03:06] DEBUG[3646]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '' [Oct 6 17:03:06] DEBUG[3646]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '1160172182.0' [Oct 6 17:03:06] DEBUG[3646]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '' Scheduling destruction of SIP dialog '413c524b-ed6b0595@10.10.15.177' in 32000 ms (Method: ACK) [Oct 6 17:03:06] DEBUG[3646]: chan_sip.c:5420 reqprep: Strict routing enforced for session 413c524b-ed6b0595@10.10.15.177 set_destination: Parsing for address/port to send to set_destination: set destination to 10.10.15.177, port 5060 Reliably Transmitting (no NAT) to 10.10.15.177:5060: BYE sip:300@10.10.15.177:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.15.192:5060;branch=z9hG4bK431a0e1d;rport From: "Test 7960" ;tag=as57c414f4 To: ;tag=89eddad3704fd6ado0 Contact: Call-ID: 413c524b-ed6b0595@10.10.15.177 CSeq: 102 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- dev-pbx*CLI> <-- SIP read from 10.10.15.177:5060: SIP/2.0 200 OK To: ;tag=89eddad3704fd6ado0 From: "Test 7960" ;tag=as57c414f4 Call-ID: 413c524b-ed6b0595@10.10.15.177 CSeq: 102 BYE Via: SIP/2.0/UDP 10.10.15.192:5060;branch=z9hG4bK431a0e1d Server: Sipura/SPA942-4.1.10(e) Content-Length: 0 --- (8 headers 0 lines)--- SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog '72764e9f31ecf4553376dbad7e8e301f@10.10.15.192' Method: BYE Really destroying SIP dialog '413c524b-ed6b0595@10.10.15.177' Method: ACK dev-pbx*CLI> <-- SIP read from 10.10.15.180:52373: REGISTER sip:10.10.15.192 SIP/2.0 Via: SIP/2.0/UDP 10.10.15.180:5060;branch=z9hG4bK00726e4b From: ;tag=001192cd77a90dec5c9c017e-39fc5a71 To: Call-ID: 001192cd-77a90003-4917c06d-702938fc@10.10.15.180 Max-Forwards: 70 Date: Fri, 06 Oct 2006 22:03:06 GMT CSeq: 3611 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ;+sip.instance="";+u.sip!model.ccm.cisco.com="7" Content-Length: 0 Expires: 60 --- (12 headers 0 lines)--- Using latest REGISTER request as basis request Sending to 10.10.15.180 : 5060 (no NAT) -- SIP Seeding peer from astdb: '102' at 102@10.10.15.180:5060 for 60 Transmitting (no NAT) to 10.10.15.180:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.15.180:5060;branch=z9hG4bK00726e4b;received=10.10.15.180 From: ;tag=001192cd77a90dec5c9c017e-39fc5a71 To: Call-ID: 001192cd-77a90003-4917c06d-702938fc@10.10.15.180 CSeq: 3611 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 --- Transmitting (no NAT) to 10.10.15.180:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.10.15.180:5060;branch=z9hG4bK00726e4b;received=10.10.15.180 From: ;tag=001192cd77a90dec5c9c017e-39fc5a71 To: ;tag=as1fb36581 Call-ID: 001192cd-77a90003-4917c06d-702938fc@10.10.15.180 CSeq: 3611 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="30eb73f3" Content-Length: 0 --- Scheduling destruction of SIP dialog '001192cd-77a90003-4917c06d-702938fc@10.10.15.180' in 32000 ms (Method: REGISTER) Scheduling destruction of SIP dialog '797ce601755e690f53f8311c6f542b88@10.10.15.192' in 32000 ms (Method: NOTIFY) Reliably Transmitting (no NAT) to 10.10.15.180:5060: NOTIFY sip:102@10.10.15.180:5060;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.10.15.192:5060;branch=z9hG4bK3ba64ed2;rport From: "asterisk" ;tag=as0873484f To: Contact: Call-ID: 797ce601755e690f53f8311c6f542b88@10.10.15.192 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 92 Messages-Waiting: no Message-Account: sip:asterisk@10.10.15.192 Voice-Message: 0/0 (0/0) --- dev-pbx*CLI> <-- SIP read from 10.10.15.180:52374: REGISTER sip:10.10.15.192 SIP/2.0 Via: SIP/2.0/UDP 10.10.15.180:5060;branch=z9hG4bK79a983ac From: ;tag=001192cd77a90dec5c9c017e-39fc5a71 To: Call-ID: 001192cd-77a90003-4917c06d-702938fc@10.10.15.180 Max-Forwards: 70 Date: Fri, 06 Oct 2006 22:03:07 GMT CSeq: 3612 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ;+sip.instance="";+u.sip!model.ccm.cisco.com="7" Authorization: Digest username="102",realm="asterisk",uri="sip:10.10.15.192",response="917b49b9efa6690fdf3665f1e961dcbd",nonce="30eb73f3",algorithm=MD5 Content-Length: 0 Expires: 60 --- (13 headers 0 lines)--- Using latest REGISTER request as basis request Sending to 10.10.15.180 : 5060 (no NAT) Transmitting (no NAT) to 10.10.15.180:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.15.180:5060;branch=z9hG4bK79a983ac;received=10.10.15.180 From: ;tag=001192cd77a90dec5c9c017e-39fc5a71 To: Call-ID: 001192cd-77a90003-4917c06d-702938fc@10.10.15.180 CSeq: 3612 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 --- -- Saved useragent "Cisco-CP7960G/8.0" for peer 102 Transmitting (no NAT) to 10.10.15.180:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.15.180:5060;branch=z9hG4bK79a983ac;received=10.10.15.180 From: ;tag=001192cd77a90dec5c9c017e-39fc5a71 To: ;tag=as1fb36581 Call-ID: 001192cd-77a90003-4917c06d-702938fc@10.10.15.180 CSeq: 3612 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: ;expires=60 Date: Fri, 06 Oct 2006 22:03:07 GMT Content-Length: 0 --- Scheduling destruction of SIP dialog '001192cd-77a90003-4917c06d-702938fc@10.10.15.180' in 32000 ms (Method: REGISTER) dev-pbx*CLI> <-- SIP read from 10.10.15.180:52375: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.15.192:5060;branch=z9hG4bK3ba64ed2;rport From: "asterisk" ;tag=as0873484f To: Call-ID: 797ce601755e690f53f8311c6f542b88@10.10.15.192 Date: Fri, 06 Oct 2006 22:03:07 GMT CSeq: 102 NOTIFY Content-Length: 0 --- (8 headers 0 lines)--- Really destroying SIP dialog '797ce601755e690f53f8311c6f542b88@10.10.15.192' Method: NOTIFY dev-pbx*CLI> <-- SIP read from 10.10.15.180:52376: INVITE sip:300@10.10.15.192 SIP/2.0 Via: SIP/2.0/UDP 10.10.15.180:5060;branch=z9hG4bK769a5b25 From: "Line 1" ;tag=001192cd77a90ded37cc8f2b-79e4b3d8 To: Call-ID: 001192cd-77a9001a-6aca2764-4c25c21f@10.10.15.180 Max-Forwards: 70 Date: Fri, 06 Oct 2006 22:03:13 GMT CSeq: 101 INVITE User-Agent: Cisco-CP7960G/8.0 Contact: Expires: 180 Accept: application/sdp Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE Remote-Party-ID: "Line 1" ;party=calling;id-type=subscriber;privacy=off;screen=yes Supported: replaces,join,norefersub Content-Length: 276 Content-Type: application/sdp Content-Disposition: session;handling=optional v=0 o=Cisco-SIPUA 17485 0 IN IP4 10.10.15.180 s=SIP Call t=0 0 m=audio 26244 RTP/AVP 0 8 18 101 c=IN IP4 10.10.15.180 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv --- (18 headers 13 lines)--- Sending to 10.10.15.180 : 5060 (no NAT) Using INVITE request as basis request - 001192cd-77a9001a-6aca2764-4c25c21f@10.10.15.180 Reliably Transmitting (no NAT) to 10.10.15.180:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.10.15.180:5060;branch=z9hG4bK769a5b25;received=10.10.15.180 From: "Line 1" ;tag=001192cd77a90ded37cc8f2b-79e4b3d8 To: ;tag=as402444ba Call-ID: 001192cd-77a9001a-6aca2764-4c25c21f@10.10.15.180 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="169229c0" Content-Length: 0 --- Scheduling destruction of SIP dialog '001192cd-77a9001a-6aca2764-4c25c21f@10.10.15.180' in 32000 ms (Method: INVITE) Found user '101' dev-pbx*CLI> <-- SIP read from 10.10.15.180:52377: ACK sip:300@10.10.15.192 SIP/2.0 Via: SIP/2.0/UDP 10.10.15.180:5060;branch=z9hG4bK769a5b25 From: "Line 1" ;tag=001192cd77a90ded37cc8f2b-79e4b3d8 To: ;tag=as402444ba Call-ID: 001192cd-77a9001a-6aca2764-4c25c21f@10.10.15.180 Date: Fri, 06 Oct 2006 22:03:13 GMT CSeq: 101 ACK Content-Length: 0 --- (8 headers 0 lines)--- dev-pbx*CLI> <-- SIP read from 10.10.15.180:52378: INVITE sip:300@10.10.15.192 SIP/2.0 Via: SIP/2.0/UDP 10.10.15.180:5060;branch=z9hG4bK4f09e051 From: "Line 1" ;tag=001192cd77a90ded37cc8f2b-79e4b3d8 To: Call-ID: 001192cd-77a9001a-6aca2764-4c25c21f@10.10.15.180 Max-Forwards: 70 Date: Fri, 06 Oct 2006 22:03:13 GMT CSeq: 102 INVITE User-Agent: Cisco-CP7960G/8.0 Contact: Proxy-Authorization: Digest username="101",realm="asterisk",uri="sip:300@10.10.15.192",response="2ed8bcf39f6015998ea91db087f1d20f",nonce="169229c0",algorithm=MD5 Expires: 180 Accept: application/sdp Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE Remote-Party-ID: "Line 1" ;party=calling;id-type=subscriber;privacy=off;screen=yes Supported: replaces,join,norefersub Content-Length: 276 Content-Type: application/sdp Content-Disposition: session;handling=optional v=0 o=Cisco-SIPUA 17485 0 IN IP4 10.10.15.180 s=SIP Call t=0 0 m=audio 26244 RTP/AVP 0 8 18 101 c=IN IP4 10.10.15.180 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv --- (19 headers 13 lines)--- Sending to 10.10.15.180 : 5060 (no NAT) Using INVITE request as basis request - 001192cd-77a9001a-6aca2764-4c25c21f@10.10.15.180 Found user '101' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 10.10.15.180:26244 Found description format PCMU for ID 0 Found description format PCMA for ID 8 Found description format G729 for ID 18 Found description format telephone-event for ID 101 Capabilities: us - 0x4 (ulaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.10.15.180:26244 Looking for 300 in international (domain 10.10.15.192) list_route: hop: Transmitting (no NAT) to 10.10.15.180:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.15.180:5060;branch=z9hG4bK4f09e051;received=10.10.15.180 From: "Line 1" ;tag=001192cd77a90ded37cc8f2b-79e4b3d8 To: Call-ID: 001192cd-77a9001a-6aca2764-4c25c21f@10.10.15.180 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 --- -- Executing [300@international:1] Macro("SIP/101-007034c0", "stdexten|300|SIP/300") in new stack -- Executing Dial("SIP/101-007034c0", "SIP/300|20") Audio is at 10.10.15.192 port 14254 Adding codec 0x800 (g726) to SDP Adding codec 0x10 (g726aal2) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.10.15.177:5060: INVITE sip:300@10.10.15.177:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.15.192:5060;branch=z9hG4bK5f129353;rport From: "Test 7960" ;tag=as5c7773dd To: Contact: Call-ID: 60ce8a3931fe6195046a56d562f8586b@10.10.15.192 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 06 Oct 2006 22:03:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 295 v=0 o=root 28500 28500 IN IP4 10.10.15.192 s=session c=IN IP4 10.10.15.192 t=0 0 m=audio 14254 RTP/AVP 111 112 101 a=rtpmap:111 G726-32/8000 a=ptime:20 a=rtpmap:112 AAL2-G726-32/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=sendrecv --- -- Called 300 dev-pbx*CLI> <-- SIP read from 10.10.15.177:5060: SIP/2.0 100 Trying To: From: "Test 7960" ;tag=as5c7773dd Call-ID: 60ce8a3931fe6195046a56d562f8586b@10.10.15.192 CSeq: 102 INVITE Via: SIP/2.0/UDP 10.10.15.192:5060;branch=z9hG4bK5f129353 Server: Sipura/SPA942-4.1.10(e) Content-Length: 0 --- (8 headers 0 lines)--- dev-pbx*CLI> <-- SIP read from 10.10.15.177:5060: SIP/2.0 488 Not Acceptable Here To: ;tag=8252ac03fa754ee5i0 From: "Test 7960" ;tag=as5c7773dd Call-ID: 60ce8a3931fe6195046a56d562f8586b@10.10.15.192 CSeq: 102 INVITE Via: SIP/2.0/UDP 10.10.15.192:5060;branch=z9hG4bK5f129353 Warning: 304 spa "Media type not available" Server: Sipura/SPA942-4.1.10(e) Content-Length: 0 --- (9 headers 0 lines)--- [Oct 6 17:03:13] DEBUG[3619]: chan_sip.c:2009 __sip_ack: Acked pending invite 102 -- Got SIP response 488 "Not Acceptable Here" back from 10.10.15.177 Transmitting (no NAT) to 10.10.15.177:5060: ACK sip:300@10.10.15.177:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.15.192:5060;branch=z9hG4bK5f129353;rport From: "Test 7960" ;tag=as5c7773dd To: ;tag=8252ac03fa754ee5i0 Contact: Call-ID: 60ce8a3931fe6195046a56d562f8586b@10.10.15.192 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/300-007081a0 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing Goto("SIP/101-007034c0", "s-CONGESTION|1") -- Goto (macro-stdexten,s-CONGESTION,1) == Auto fallthrough, channel 'SIP/101-007034c0' status is 'CONGESTION' Transmitting (no NAT) to 10.10.15.180:5060: SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 10.10.15.180:5060;branch=z9hG4bK4f09e051;received=10.10.15.180 From: "Line 1" ;tag=001192cd77a90ded37cc8f2b-79e4b3d8 To: ;tag=as6123a77f Call-ID: 001192cd-77a9001a-6aca2764-4c25c21f@10.10.15.180 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 X-Asterisk-HangupCause: Bearer capability not available X-Asterisk-HangupCauseCode: 58 --- [Oct 6 17:03:13] DEBUG[3654]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '"Test 7960" <101>' [Oct 6 17:03:13] DEBUG[3654]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '101' [Oct 6 17:03:13] DEBUG[3654]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '300' [Oct 6 17:03:13] DEBUG[3654]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is 'international' [Oct 6 17:03:13] DEBUG[3654]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is 'SIP/101-007034c0' [Oct 6 17:03:13] DEBUG[3654]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is 'SIP/300-007081a0' [Oct 6 17:03:13] DEBUG[3654]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is 'Goto' [Oct 6 17:03:13] DEBUG[3654]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is 's-CONGESTION|1' [Oct 6 17:03:13] DEBUG[3654]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '2006-10-06 17:03:13' [Oct 6 17:03:13] DEBUG[3654]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '' [Oct 6 17:03:13] DEBUG[3654]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '2006-10-06 17:03:13' [Oct 6 17:03:13] DEBUG[3654]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '0' [Oct 6 17:03:13] DEBUG[3654]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '0' [Oct 6 17:03:13] DEBUG[3654]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is 'FAILED' [Oct 6 17:03:13] DEBUG[3654]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is 'DOCUMENTATION' [Oct 6 17:03:13] DEBUG[3654]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '' [Oct 6 17:03:13] DEBUG[3654]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '1160172193.2' [Oct 6 17:03:13] DEBUG[3654]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '' dev-pbx*CLI> <-- SIP read from 10.10.15.180:52379: ACK sip:300@10.10.15.192 SIP/2.0 Via: SIP/2.0/UDP 10.10.15.180:5060;branch=z9hG4bK4f09e051 From: "Line 1" ;tag=001192cd77a90ded37cc8f2b-79e4b3d8 To: ;tag=as6123a77f Call-ID: 001192cd-77a9001a-6aca2764-4c25c21f@10.10.15.180 Date: Fri, 06 Oct 2006 22:03:14 GMT CSeq: 102 ACK Content-Length: 0 --- (8 headers 0 lines)--- Really destroying SIP dialog '60ce8a3931fe6195046a56d562f8586b@10.10.15.192' Method: INVITE Really destroying SIP dialog '001192cd-77a9001a-6aca2764-4c25c21f@10.10.15.180' Method: ACK dev-pbx*CLI> <-- SIP read from 10.10.15.180:52380: REGISTER sip:10.10.15.192 SIP/2.0 Via: SIP/2.0/UDP 10.10.15.180:5060;branch=z9hG4bK1d93ffca From: ;tag=001192cd77a90dee3e4114de-42294eb1 To: Call-ID: 001192cd-77a90002-312f52e8-351b961e@10.10.15.180 Max-Forwards: 70 Date: Fri, 06 Oct 2006 22:03:14 GMT CSeq: 3609 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ;+sip.instance="";+u.sip!model.ccm.cisco.com="7" Content-Length: 0 Expires: 60 --- (12 headers 0 lines)--- Using latest REGISTER request as basis request Sending to 10.10.15.180 : 5060 (no NAT) Transmitting (no NAT) to 10.10.15.180:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.15.180:5060;branch=z9hG4bK1d93ffca;received=10.10.15.180 From: ;tag=001192cd77a90dee3e4114de-42294eb1 To: Call-ID: 001192cd-77a90002-312f52e8-351b961e@10.10.15.180 CSeq: 3609 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 --- Transmitting (no NAT) to 10.10.15.180:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.10.15.180:5060;branch=z9hG4bK1d93ffca;received=10.10.15.180 From: ;tag=001192cd77a90dee3e4114de-42294eb1 To: ;tag=as2cd113f5 Call-ID: 001192cd-77a90002-312f52e8-351b961e@10.10.15.180 CSeq: 3609 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3a5b473e" Content-Length: 0 --- Scheduling destruction of SIP dialog '001192cd-77a90002-312f52e8-351b961e@10.10.15.180' in 32000 ms (Method: REGISTER) dev-pbx*CLI> <-- SIP read from 10.10.15.180:52381: REGISTER sip:10.10.15.192 SIP/2.0 Via: SIP/2.0/UDP 10.10.15.180:5060;branch=z9hG4bK173aa5a3 From: ;tag=001192cd77a90dee3e4114de-42294eb1 To: Call-ID: 001192cd-77a90002-312f52e8-351b961e@10.10.15.180 Max-Forwards: 70 Date: Fri, 06 Oct 2006 22:03:14 GMT CSeq: 3610 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ;+sip.instance="";+u.sip!model.ccm.cisco.com="7" Authorization: Digest username="101",realm="asterisk",uri="sip:10.10.15.192",response="29773480b6a47d2a8c1d91eaa2587dc3",nonce="3a5b473e",algorithm=MD5 Content-Length: 0 Expires: 60 --- (13 headers 0 lines)--- Using latest REGISTER request as basis request Sending to 10.10.15.180 : 5060 (no NAT) Transmitting (no NAT) to 10.10.15.180:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.15.180:5060;branch=z9hG4bK173aa5a3;received=10.10.15.180 From: ;tag=001192cd77a90dee3e4114de-42294eb1 To: Call-ID: 001192cd-77a90002-312f52e8-351b961e@10.10.15.180 CSeq: 3610 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 --- Transmitting (no NAT) to 10.10.15.180:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.15.180:5060;branch=z9hG4bK173aa5a3;received=10.10.15.180 From: ;tag=001192cd77a90dee3e4114de-42294eb1 To: ;tag=as2cd113f5 Call-ID: 001192cd-77a90002-312f52e8-351b961e@10.10.15.180 CSeq: 3610 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: ;expires=60 Date: Fri, 06 Oct 2006 22:03:14 GMT Content-Length: 0 --- Scheduling destruction of SIP dialog '001192cd-77a90002-312f52e8-351b961e@10.10.15.180' in 32000 ms (Method: REGISTER) dev-pbx*CLI> exit