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-a7685f64 From: ;tag=1aeb74d99ae9ae9co0 To: "Test 7960" Call-ID: d7d3db5d-626d9e8@10.10.15.177 CSeq: 101 INVITE Max-Forwards: 70 Contact: Expires: 240 User-Agent: Sipura/SPA942-4.1.10(e) Content-Length: 209 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Content-Type: application/sdp v=0-pbx*CLI> o=- 365346 365346 IN IP4 10.10.15.177 s=- c=IN IP4 10.10.15.177 t=0 0 m=audio 16478 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 - d7d3db5d-626d9e8@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-a7685f64;received=10.10.15.177 From: ;tag=1aeb74d99ae9ae9co0 To: "Test 7960" ;tag=as7f130977 Call-ID: d7d3db5d-626d9e8@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="6e7db633" Content-Length: 0 --- Scheduling destruction of SIP dialog 'd7d3db5d-626d9e8@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-a7685f64 From: ;tag=1aeb74d99ae9ae9co0 To: "Test 7960" ;tag=as7f130977 Call-ID: d7d3db5d-626d9e8@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-c5a4256c From: ;tag=1aeb74d99ae9ae9co0 To: "Test 7960" Call-ID: d7d3db5d-626d9e8@10.10.15.177 CSeq: 102 INVITE Max-Forwards: 70 Proxy-Authorization: Digest username="300",realm="asterisk",nonce="6e7db633",uri="sip:101@10.10.15.192",algorithm=MD5,response="4c78b6b96b6953d8a80770599a22aa8a" Contact: Expires: 240 User-Agent: Sipura/SPA942-4.1.10(e) Content-Length: 209 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Content-Type: application/sdp v=0 o=- 365346 365346 IN IP4 10.10.15.177 s=- c=IN IP4 10.10.15.177 t=0 0 m=audio 16478 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 - d7d3db5d-626d9e8@10.10.15.177 Found user '300' Found RTP audio format 2 Found RTP audio format 101 Peer audio RTP is at port 10.10.15.177:16478 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 10.10.15.177:16478 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-c5a4256c;received=10.10.15.177 From: ;tag=1aeb74d99ae9ae9co0 To: "Test 7960" Call-ID: d7d3db5d-626d9e8@10.10.15.177 Seq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 --- -- Executing Macro("SIP/300-00701530", "stdexten|101|SIP/101") -- Executing Dial("SIP/300-00701530", "SIP/101|20") Audio is at 10.10.15.192 port 15902 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=z9hG4bK444659e2;rport From: "300" ;tag=as1566a6df To: Contact: Call-ID: 5d7b57533eb95bbd7962336152fc4336@10.10.15.192 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 05 Oct 2006 22:03:04 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 15902 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 10.10.15.180:51494: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.15.192:5060;branch=z9hG4bK444659e2;rport From: "300" ;tag=as1566a6df To: Call-ID: 5d7b57533eb95bbd7962336152fc4336@10.10.15.192 Date: Thu, 05 Oct 2006 22:03:04 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 10.10.15.180:51495: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.10.15.192:5060;branch=z9hG4bK444659e2;rport From: "300" ;tag=as1566a6df To: ;tag=001192cd77a901c5368e37ae-3474fc16 Call-ID: 5d7b57533eb95bbd7962336152fc4336@10.10.15.192 Date: Thu, 05 Oct 2006 22:03:04 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/101-00706470 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-c5a4256c;received=10.10.15.177 From: ;tag=1aeb74d99ae9ae9co0 To: "Test 7960" ;tag=as3c1c72b0 Call-ID: d7d3db5d-626d9e8@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 --- Really destroying SIP dialog '0012433d-e04a0002-4851da05-18fec5b6@10.10.15.197' Method: REGISTER Really destroying SIP dialog '0012433d-e04a0003-4eebe70c-04b37b8e@10.10.15.197' Method: REGISTER dev-pbx*CLI> <-- SIP read from 10.10.15.180:51496: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.15.192:5060;branch=z9hG4bK444659e2;rport From: "300" ;tag=as1566a6df To: ;tag=001192cd77a901c5368e37ae-3474fc16 Call-ID: 5d7b57533eb95bbd7962336152fc4336@10.10.15.192 Date: Thu, 05 Oct 2006 22:03:07 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 26245 0 IN IP4 10.10.15.180 s=SIP Call t=0 0 m=audio 31184 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)--- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 10.10.15.180:31184 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 10.10.15.180:31184 list_route: hop: 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=z9hG4bK0adb84c4;rport From: "300" ;tag=as1566a6df To: ;tag=001192cd77a901c5368e37ae-3474fc16 Contact: Call-ID: 5d7b57533eb95bbd7962336152fc4336@10.10.15.192 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/101-00706470 answered SIP/300-00701530 Audio is at 10.10.15.192 port 17478 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-c5a4256c;received=10.10.15.177 From: ;tag=1aeb74d99ae9ae9co0 To: "Test 7960" ;tag=as3c1c72b0 Call-ID: d7d3db5d-626d9e8@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 17478 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-c3910038 From: ;tag=1aeb74d99ae9ae9co0 To: "Test 7960" ;tag=as3c1c72b0 Call-ID: d7d3db5d-626d9e8@10.10.15.177 CSeq: 102 ACK Max-Forwards: 70 Proxy-Authorization: Digest username="300",realm="asterisk",nonce="6e7db633",uri="sip:101@10.10.15.192",algorithm=MD5,response="819a6b2ce4a522c5ce5106c3aef941ac" Contact: User-Agent: Sipura/SPA942-4.1.10(e) Content-Length: 0 --- (11 headers 0 lines)--- dev-pbx*CLI> <-- SIP read from 10.10.15.180:51497: BYE sip:300@10.10.15.192:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.15.180:5060;branch=z9hG4bK35d1acb9 From: ;tag=001192cd77a901c5368e37ae-3474fc16 To: "300" ;tag=as1566a6df Call-ID: 5d7b57533eb95bbd7962336152fc4336@10.10.15.192 Max-Forwards: 70 Date: Thu, 05 Oct 2006 22:03:08 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=z9hG4bK35d1acb9;received=10.10.15.180 From: ;tag=001192cd77a901c5368e37ae-3474fc16 To: "300" ;tag=as1566a6df Call-ID: 5d7b57533eb95bbd7962336152fc4336@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 --- == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'SIP/300-00701530' in macro 'stdexten' == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'SIP/300-00701530' Scheduling destruction of SIP dialog 'd7d3db5d-626d9e8@10.10.15.177' in 32000 ms (Method: ACK) 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=z9hG4bK301ce31c;rport From: "Test 7960" ;tag=as3c1c72b0 To: ;tag=1aeb74d99ae9ae9co0 Contact: Call-ID: d7d3db5d-626d9e8@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=1aeb74d99ae9ae9co0 From: "Test 7960" ;tag=as3c1c72b0 Call-ID: d7d3db5d-626d9e8@10.10.15.177 CSeq: 102 BYE Via: SIP/2.0/UDP 10.10.15.192:5060;branch=z9hG4bK301ce31c 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 '5d7b57533eb95bbd7962336152fc4336@10.10.15.192' Method: BYE Really destroying SIP dialog 'd7d3db5d-626d9e8@10.10.15.177' Method: ACK dev-pbx*CLI> <-- SIP read from 10.10.15.180:51498: REGISTER sip:10.10.15.192 SIP/2.0 Via: SIP/2.0/UDP 10.10.15.180:5060;branch=z9hG4bK77a48a79 From: ;tag=001192cd77a901c6299f52f0-19d63ea4 To: Call-ID: 001192cd-77a90003-4917c06d-702938fc@10.10.15.180 Max-Forwards: 70 Date: Thu, 05 Oct 2006 22:03:10 GMT CSeq: 503 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=z9hG4bK77a48a79;received=10.10.15.180 From: ;tag=001192cd77a901c6299f52f0-19d63ea4 To: Call-ID: 001192cd-77a90003-4917c06d-702938fc@10.10.15.180 CSeq: 503 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=z9hG4bK77a48a79;received=10.10.15.180 From: ;tag=001192cd77a901c6299f52f0-19d63ea4 To: ;tag=as5a743712 Call-ID: 001192cd-77a90003-4917c06d-702938fc@10.10.15.180 CSeq: 503 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4ce8c069" 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 10.10.15.180:51499: REGISTER sip:10.10.15.192 SIP/2.0 Via: SIP/2.0/UDP 10.10.15.180:5060;branch=z9hG4bK74158d55 From: ;tag=001192cd77a901c6299f52f0-19d63ea4 To: Call-ID: 001192cd-77a90003-4917c06d-702938fc@10.10.15.180 Max-Forwards: 70 Date: Thu, 05 Oct 2006 22:03:10 GMT CSeq: 504 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="ef8116efcc0dacc214c7f07cd1b63bc5",nonce="4ce8c069",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=z9hG4bK74158d55;received=10.10.15.180 From: ;tag=001192cd77a901c6299f52f0-19d63ea4 To: Call-ID: 001192cd-77a90003-4917c06d-702938fc@10.10.15.180 CSeq: 504 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=z9hG4bK74158d55;received=10.10.15.180 From: ;tag=001192cd77a901c6299f52f0-19d63ea4 To: ;tag=as5a743712 Call-ID: 001192cd-77a90003-4917c06d-702938fc@10.10.15.180 CSeq: 504 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: ;expires=60 Date: Thu, 05 Oct 2006 22:03:10 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 10.10.15.180:51500: REGISTER sip:10.10.15.192 SIP/2.0 Via: SIP/2.0/UDP 10.10.15.180:5060;branch=z9hG4bK74189a47 From: ;tag=001192cd77a901c77b4ab756-71ae3281 To: Call-ID: 001192cd-77a90002-312f52e8-351b961e@10.10.15.180 Max-Forwards: 70 Date: Thu, 05 Oct 2006 22:03:10 GMT CSeq: 501 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=z9hG4bK74189a47;received=10.10.15.180 From: ;tag=001192cd77a901c77b4ab756-71ae3281 To: Call-ID: 001192cd-77a90002-312f52e8-351b961e@10.10.15.180 CSeq: 501 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=z9hG4bK74189a47;received=10.10.15.180 From: ;tag=001192cd77a901c77b4ab756-71ae3281 To: ;tag=as651676d9 Call-ID: 001192cd-77a90002-312f52e8-351b961e@10.10.15.180 CSeq: 501 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="54c19250" 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 10.10.15.180:51501: REGISTER sip:10.10.15.192 SIP/2.0 Via: SIP/2.0/UDP 10.10.15.180:5060;branch=z9hG4bK4ea398f0 From: ;tag=001192cd77a901c77b4ab756-71ae3281 To: Call-ID: 001192cd-77a90002-312f52e8-351b961e@10.10.15.180 Max-Forwards: 70 Date: Thu, 05 Oct 2006 22:03:11 GMT CSeq: 502 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="18e57d3c7a264ad2af092b8f822f77e6",nonce="54c19250",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=z9hG4bK4ea398f0;received=10.10.15.180 From: ;tag=001192cd77a901c77b4ab756-71ae3281 To: Call-ID: 001192cd-77a90002-312f52e8-351b961e@10.10.15.180 CSeq: 502 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=z9hG4bK4ea398f0;received=10.10.15.180 From: ;tag=001192cd77a901c77b4ab756-71ae3281 To: ;tag=as651676d9 Call-ID: 001192cd-77a90002-312f52e8-351b961e@10.10.15.180 CSeq: 502 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: ;expires=60 Date: Thu, 05 Oct 2006 22:03:11 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> <-- SIP read from 10.10.15.180:51502: INVITE sip:300@10.10.15.192 SIP/2.0 Via: SIP/2.0/UDP 10.10.15.180:5060;branch=z9hG4bK439322ac From: "Line 1" ;tag=001192cd77a901c850b19851-4994f5bd To: Call-ID: 001192cd-77a90019-2d5a5554-73a76fef@10.10.15.180 Max-Forwards: 70 Date: Thu, 05 Oct 2006 22:03:11 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: 275 Content-Type: application/sdp Content-Disposition: session;handling=optional v=0 o=Cisco-SIPUA 8298 0 IN IP4 10.10.15.180 s=SIP Call t=0 0 m=audio 22336 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-77a90019-2d5a5554-73a76fef@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=z9hG4bK439322ac;received=10.10.15.180 From: "Line 1" ;tag=001192cd77a901c850b19851-4994f5bd To: ;tag=as1fd380d7 Call-ID: 001192cd-77a90019-2d5a5554-73a76fef@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="0ceba6c8" Content-Length: 0 --- Scheduling destruction of SIP dialog '001192cd-77a90019-2d5a5554-73a76fef@10.10.15.180' in 32000 ms (Method: INVITE) Found user '101' dev-pbx*CLI> <-- SIP read from 10.10.15.180:51503: ACK sip:300@10.10.15.192 SIP/2.0 Via: SIP/2.0/UDP 10.10.15.180:5060;branch=z9hG4bK439322ac From: "Line 1" ;tag=001192cd77a901c850b19851-4994f5bd To: ;tag=as1fd380d7 Call-ID: 001192cd-77a90019-2d5a5554-73a76fef@10.10.15.180 Date: Thu, 05 Oct 2006 22:03:12 GMT CSeq: 101 ACK Content-Length: 0 --- (8 headers 0 lines)--- dev-pbx*CLI> <-- SIP read from 10.10.15.180:51504: INVITE sip:300@10.10.15.192 SIP/2.0 Via: SIP/2.0/UDP 10.10.15.180:5060;branch=z9hG4bK764ec71a From: "Line 1" ;tag=001192cd77a901c850b19851-4994f5bd To: Call-ID: 001192cd-77a90019-2d5a5554-73a76fef@10.10.15.180 Max-Forwards: 70 Date: Thu, 05 Oct 2006 22:03:12 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="3ed64927596c86e84a8d447421a75a75",nonce="0ceba6c8",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: 275 Content-Type: application/sdp Content-Disposition: session;handling=optional v=0 o=Cisco-SIPUA 8298 0 IN IP4 10.10.15.180 s=SIP Call t=0 0 m=audio 22336 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-77a90019-2d5a5554-73a76fef@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 10.10.15.180:22336 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 10.10.15.180:22336 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=z9hG4bK764ec71a;received=10.10.15.180 From: "Line 1" ;tag=001192cd77a901c850b19851-4994f5bd To: Call-ID: 001192cd-77a90019-2d5a5554-73a76fef@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/101-007040b0", "stdexten|300|SIP/300") in new stack -- Executing Dial("SIP/101-007040b0", "SIP/300|20") Audio is at 10.10.15.192 port 15052 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=z9hG4bK57c028ff;rport From: "Test 7960" ;tag=as7130b2e5 To: Contact: Call-ID: 7c008c8f75adbdef054b984f2b805d6a@10.10.15.192 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 05 Oct 2006 22:03:12 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 295 v=0 =root 28500 28500 IN IP4 10.10.15.192 s=session c=IN IP4 10.10.15.192 t=0 0 m=audio 15052 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=as7130b2e5 Call-ID: 7c008c8f75adbdef054b984f2b805d6a@10.10.15.192 CSeq: 102 INVITE Via: SIP/2.0/UDP 10.10.15.192:5060;branch=z9hG4bK57c028ff 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=c13466c9e4266206i0 From: "Test 7960" ;tag=as7130b2e5 Call-ID: 7c008c8f75adbdef054b984f2b805d6a@10.10.15.192 CSeq: 102 INVITE Via: SIP/2.0/UDP 10.10.15.192:5060;branch=z9hG4bK57c028ff Warning: 304 spa "Media type not available" Server: Sipura/SPA942-4.1.10(e) Content-Length: 0 --- (9 headers 0 lines)--- -- 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=z9hG4bK57c028ff;rport From: "Test 7960" ;tag=as7130b2e5 To: ;tag=c13466c9e4266206i0 Contact: Call-ID: 7c008c8f75adbdef054b984f2b805d6a@10.10.15.192 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/300-0070be40 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing Goto("SIP/101-007040b0", "s-CONGESTION|1") -- Goto (macro-stdexten,s-CONGESTION,1) == Auto fallthrough, channel 'SIP/101-007040b0' 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=z9hG4bK764ec71a;received=10.10.15.180 From: "Line 1" ;tag=001192cd77a901c850b19851-4994f5bd To: ;tag=as0ca6d9c4 Call-ID: 001192cd-77a90019-2d5a5554-73a76fef@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 --- dev-pbx*CLI> <-- SIP read from 10.10.15.180:51505: ACK sip:300@10.10.15.192 SIP/2.0 Via: SIP/2.0/UDP 10.10.15.180:5060;branch=z9hG4bK764ec71a From: "Line 1" ;tag=001192cd77a901c850b19851-4994f5bd To: ;tag=as0ca6d9c4 Call-ID: 001192cd-77a90019-2d5a5554-73a76fef@10.10.15.180 Date: Thu, 05 Oct 2006 22:03:12 GMT CSeq: 102 ACK Content-Length: 0 --- (8 headers 0 lines)--- Really destroying SIP dialog '7c008c8f75adbdef054b984f2b805d6a@10.10.15.192' Method: INVITE Really destroying SIP dialog '001192cd-77a90019-2d5a5554-73a76fef@10.10.15.180' Method: ACK dev-pbx*CLI>