Asterisk SVN-branch-1.4-r49313M, 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-branch-1.4-r49313M currently running on steven-sokols-computer (pid = 14912) steven-sokols-computer*CLI> nnection Verbosity is at least 8 steven-sokols-computer*CLI> sip debset debug steven-sokols-computer*CLI> SIP Debugging enabled steven-sokols-computer*CLI> <--- SIP read from 192.168.100.163:5060 ---> INVITE sip:6002@192.168.100.249 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.163:5060;branch=z9hG4bK-dd29d47b From: "Paul Barham" ;tag=e1309764c9bf5f93o0 To: "Paul Barham" Call-ID: abd77c78-d7a59d8f@192.168.100.163 CSeq: 101 INVITE Max-Forwards: 70 Contact: "Paul Barham" Expires: 240 User-Agent: Linksys/SPA962-5.1.3 Content-Length: 401 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp v=0 o=- 163112 163112 IN IP4 192.168.100.163 s=- c=IN IP4 192.168.100.163 t=0 0 m=audio 16472 RTP/AVP 0 2 4 8 18 96 97 98 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> steven-sokols-computer*CLI> --- (14 headers 18 lines) --- steven-sokols-computer*CLI> Sending to 192.168.100.163 : 5060 (NAT) steven-sokols-computer*CLI> Using INVITE request as basis request - abd77c78-d7a59d8f@192.168.100.163 steven-sokols-computer*CLI> Found peer '6001' steven-sokols-computer*CLI> <--- Reliably Transmitting (NAT) to 192.168.100.163:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.100.163:5060;branch=z9hG4bK-dd29d47b;received=192.168.100.163 From: "Paul Barham" ;tag=e1309764c9bf5f93o0 To: "Paul Barham" ;tag=as1ba3b097 Call-ID: abd77c78-d7a59d8f@192.168.100.163 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="", nonce="26fba3da" Content-Length: 0 <------------> steven-sokols-computer*CLI> Scheduling destruction of SIP dialog 'abd77c78-d7a59d8f@192.168.100.163' in 32000 ms (Method: INVITE) steven-sokols-computer*CLI> <--- SIP read from 192.168.100.163:5060 ---> ACK sip:6002@192.168.100.249 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.163:5060;branch=z9hG4bK-dd29d47b From: "Paul Barham" ;tag=e1309764c9bf5f93o0 To: "Paul Barham" ;tag=as1ba3b097 Call-ID: abd77c78-d7a59d8f@192.168.100.163 CSeq: 101 ACK Max-Forwards: 70 Contact: "Paul Barham" User-Agent: Linksys/SPA962-5.1.3 Content-Length: 0 <-------------> steven-sokols-computer*CLI> --- (10 headers 0 lines) --- steven-sokols-computer*CLI> <--- SIP read from 192.168.100.163:5060 ---> INVITE sip:6002@192.168.100.249 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.163:5060;branch=z9hG4bK-9d87293b From: "Paul Barham" ;tag=e1309764c9bf5f93o0 To: "Paul Barham" Call-ID: abd77c78-d7a59d8f@192.168.100.163 CSeq: 102 INVITE Max-Forwards: 70 Proxy-Authorization: Digest username="6001",realm="",nonce="26fba3da",uri="sip:6002@192.168.100.249",algorithm=MD5,response="2865a1d790b0f15d90a52776512041e6" Contact: "Paul Barham" Expires: 240 User-Agent: Linksys/SPA962-5.1.3 Content-Length: 401 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp v=0 o=- 163112 163112 IN IP4 192.168.100.163 s=- c=IN IP4 192.168.100.163 t=0 0 m=audio 16472 RTP/AVP 0 2 4 8 18 96 97 98 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> steven-sokols-computer*CLI> --- (15 headers 18 lines) --- steven-sokols-computer*CLI> Sending to 192.168.100.163 : 5060 (NAT) Using INVITE request as basis request - abd77c78-d7a59d8f@192.168.100.163 steven-sokols-computer*CLI> Found peer '6001' steven-sokols-computer*CLI> Found RTP audio format 0 steven-sokols-computer*CLI> Found RTP audio format 2 steven-sokols-computer*CLI> Found RTP audio format 4 steven-sokols-computer*CLI> Found RTP audio format 8 Found RTP audio format 18 steven-sokols-computer*CLI> Found RTP audio format 96 Found RTP audio format 97 steven-sokols-computer*CLI> Found RTP audio format 98 Found RTP audio format 101 steven-sokols-computer*CLI> Peer audio RTP is at port 192.168.100.163:16472 steven-sokols-computer*CLI> Found description format PCMU for ID 0 Found description format G726-32 for ID 2 Found description format G723 for ID 4 Found description format PCMA for ID 8 steven-sokols-computer*CLI> Found description format G729a for ID 18 Found description format G726-40 for ID 96 Found description format G726-24 for ID 97 steven-sokols-computer*CLI> Found description format G726-16 for ID 98 Found description format telephone-event for ID 101 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd0d (g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) steven-sokols-computer*CLI> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.100.163:16472 Looking for 6002 in user-6001 (domain 192.168.100.249) steven-sokols-computer*CLI> list_route: hop: steven-sokols-computer*CLI> <--- Transmitting (NAT) to 192.168.100.163:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.100.163:5060;branch=z9hG4bK-9d87293b;received=192.168.100.163 From: "Paul Barham" ;tag=e1309764c9bf5f93o0 To: "Paul Barham" Call-ID: abd77c78-d7a59d8f@192.168.100.163 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> steven-sokols-computer*CLI> -- Executing [6002@user-6001:1] Macro("SIP/6001-01843c00", "stdexten|6002|SIP/6002&IAX2/6002") in new stack steven-sokols-computer*CLI> -- Executing [s@macro-stdexten:1] Dial("SIP/6001-01843c00", "SIP/6002&IAX2/6002|20") in new stack steven-sokols-computer*CLI> Audio is at 192.168.100.249 port 11674 steven-sokols-computer*CLI> Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP steven-sokols-computer*CLI> Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 192.168.100.171:5060: INVITE sip:6002@192.168.100.171:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.249:5060;branch=z9hG4bK3f2c3a68;rport From: "Steven Sokol" ;tag=as7173673a To: Contact: Call-ID: 4568b4dc3c58472a353399a000b020ef@192.168.100.249 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 08 Jan 2007 16:54:06 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 293 v=0 o=root 14912 14912 IN IP4 192.168.100.249 s=session c=IN IP4 192.168.100.249 t=0 0 m=audio 11674 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=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- steven-sokols-computer*CLI> -- Called 6002 steven-sokols-computer*CLI> [Jan 8 10:54:06] WARNING[14912]: app_dial.c:1081 dial_exec_full: Unable to create channel of type 'IAX2' (cause 3 - No route to destination) steven-sokols-computer*CLI> <--- SIP read from 192.168.100.171:5060 ---> SIP/2.0 100 Trying To: From: "Steven Sokol" ;tag=as7173673a Call-ID: 4568b4dc3c58472a353399a000b020ef@192.168.100.249 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.100.249:5060;branch=z9hG4bK3f2c3a68 Server: Sipura/SPA841-3.1.3(a) Content-Length: 0 <-------------> steven-sokols-computer*CLI> --- (8 headers 0 lines) --- steven-sokols-computer*CLI> <--- SIP read from 192.168.100.171:5060 ---> SIP/2.0 180 Ringing To: ;tag=733bbc4c5c0edf54i0 From: "Steven Sokol" ;tag=as7173673a Call-ID: 4568b4dc3c58472a353399a000b020ef@192.168.100.249 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.100.249:5060;branch=z9hG4bK3f2c3a68 Server: Sipura/SPA841-3.1.3(a) Remote-Party-ID: "Steven Sokol" ;screen=yes;party=called Content-Length: 0 <-------------> steven-sokols-computer*CLI> --- (9 headers 0 lines) --- steven-sokols-computer*CLI> -- SIP/6002-01848c00 is ringing steven-sokols-computer*CLI> <--- Transmitting (NAT) to 192.168.100.163:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.100.163:5060;branch=z9hG4bK-9d87293b;received=192.168.100.163 From: "Paul Barham" ;tag=e1309764c9bf5f93o0 To: "Paul Barham" ;tag=as48574206 Call-ID: abd77c78-d7a59d8f@192.168.100.163 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> steven-sokols-computer*CLI> <--- SIP read from 192.168.100.171:5060 ---> SIP/2.0 200 OK To: ;tag=733bbc4c5c0edf54i0 From: "Steven Sokol" ;tag=as7173673a Call-ID: 4568b4dc3c58472a353399a000b020ef@192.168.100.249 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.100.249:5060;branch=z9hG4bK3f2c3a68 Contact: "Steven Sokol" Server: Sipura/SPA841-3.1.3(a) Remote-Party-ID: "Steven Sokol" ;screen=yes;party=called Content-Length: 210 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Content-Type: application/sdp v=0 o=- 72881 72881 IN IP4 192.168.100.171 s=- c=IN IP4 192.168.100.171 t=0 0 m=audio 16450 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> steven-sokols-computer*CLI> --- (12 headers 11 lines) --- steven-sokols-computer*CLI> Found RTP audio format 0 steven-sokols-computer*CLI> Found RTP audio format 101 Peer audio RTP is at port 192.168.100.171:16450 steven-sokols-computer*CLI> Found description format PCMU for ID 0 Found description format telephone-event for ID 101 steven-sokols-computer*CLI> Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), 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 192.168.100.171:16450 steven-sokols-computer*CLI> list_route: hop: steven-sokols-computer*CLI> set_destination: Parsing for address/port to send to steven-sokols-computer*CLI> set_destination: set destination to 192.168.100.171, port 5060 Transmitting (NAT) to 192.168.100.171:5060: ACK sip:6002@192.168.100.171:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.249:5060;branch=z9hG4bK7cc3af5f;rport From: "Steven Sokol" ;tag=as7173673a To: ;tag=733bbc4c5c0edf54i0 Contact: Call-ID: 4568b4dc3c58472a353399a000b020ef@192.168.100.249 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- steven-sokols-computer*CLI> -- SIP/6002-01848c00 answered SIP/6001-01843c00 steven-sokols-computer*CLI> Audio is at 192.168.100.249 port 15034 steven-sokols-computer*CLI> Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP steven-sokols-computer*CLI> <--- Reliably Transmitting (NAT) to 192.168.100.163:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.100.163:5060;branch=z9hG4bK-9d87293b;received=192.168.100.163 From: "Paul Barham" ;tag=e1309764c9bf5f93o0 To: "Paul Barham" ;tag=as48574206 Call-ID: abd77c78-d7a59d8f@192.168.100.163 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: 270 v=0 o=root 14912 14912 IN IP4 192.168.100.171 s=session c=IN IP4 192.168.100.171 t=0 0 m=audio 16450 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> steven-sokols-computer*CLI> -- Native bridging SIP/6001-01843c00 and SIP/6002-01848c00 steven-sokols-computer*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.100.171, port 5060 steven-sokols-computer*CLI> Audio is at 192.168.100.249 port 11674 steven-sokols-computer*CLI> Adding codec 0x4 (ulaw) to SDP steven-sokols-computer*CLI> Adding non-codec 0x1 (telephone-event) to SDP steven-sokols-computer*CLI> Reliably Transmitting (NAT) to 192.168.100.171:5060: INVITE sip:6002@192.168.100.171:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.249:5060;branch=z9hG4bK2b7d2522;rport From: "Steven Sokol" ;tag=as7173673a To: ;tag=733bbc4c5c0edf54i0 Contact: Call-ID: 4568b4dc3c58472a353399a000b020ef@192.168.100.249 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: 246 v=0 o=root 14912 14913 IN IP4 192.168.100.163 s=session c=IN IP4 192.168.100.163 t=0 0 m=audio 16472 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- steven-sokols-computer*CLI> <--- SIP read from 192.168.100.163:5060 ---> ACK sip:6002@192.168.100.249 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.163:5060;branch=z9hG4bK-f8ea006 From: "Paul Barham" ;tag=e1309764c9bf5f93o0 To: "Paul Barham" ;tag=as48574206 Call-ID: abd77c78-d7a59d8f@192.168.100.163 CSeq: 102 ACK Max-Forwards: 70 Proxy-Authorization: Digest username="6001",realm="",nonce="26fba3da",uri="sip:6002@192.168.100.249",algorithm=MD5,response="32fd6f8de49033a4649e0e8d68fcb1a8" Contact: "Paul Barham" User-Agent: Linksys/SPA962-5.1.3 Content-Length: 0 <-------------> steven-sokols-computer*CLI> --- (11 headers 0 lines) --- steven-sokols-computer*CLI> <--- SIP read from 192.168.100.171:5060 ---> SIP/2.0 200 OK To: ;tag=733bbc4c5c0edf54i0 From: "Steven Sokol" ;tag=as7173673a Call-ID: 4568b4dc3c58472a353399a000b020ef@192.168.100.249 CSeq: 103 INVITE Via: SIP/2.0/UDP 192.168.100.249:5060;branch=z9hG4bK2b7d2522 Contact: "Steven Sokol" Server: Sipura/SPA841-3.1.3(a) Content-Length: 210 Content-Type: application/sdp v=0 o=- 72969 72969 IN IP4 192.168.100.171 s=- c=IN IP4 192.168.100.171 t=0 0 m=audio 16450 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> steven-sokols-computer*CLI> --- (10 headers 11 lines) --- Found RTP audio format 0 steven-sokols-computer*CLI> Found RTP audio format 101 steven-sokols-computer*CLI> Peer audio RTP is at port 192.168.100.171:16450 steven-sokols-computer*CLI> Found description format PCMU for ID 0 steven-sokols-computer*CLI> Found description format telephone-event for ID 101 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) steven-sokols-computer*CLI> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) steven-sokols-computer*CLI> Peer audio RTP is at port 192.168.100.171:16450 steven-sokols-computer*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.100.171, port 5060 steven-sokols-computer*CLI> Transmitting (NAT) to 192.168.100.171:5060: ACK sip:6002@192.168.100.171:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.249:5060;branch=z9hG4bK3d59d580;rport From: "Steven Sokol" ;tag=as7173673a To: ;tag=733bbc4c5c0edf54i0 Contact: Call-ID: 4568b4dc3c58472a353399a000b020ef@192.168.100.249 CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- steven-sokols-computer*CLI> -- Accepting AUTHENTICATED call from 127.0.0.1: > requested format = gsm, > requested prefs = (), > actual format = ulaw, > host prefs = (ulaw|alaw|gsm), > priority = mine steven-sokols-computer*CLI> -- Executing [*846002@user-6000:1] NoOp("IAX2/6000-6", "Call Monitoring Enabled: 6002") in new stack steven-sokols-computer*CLI> -- Executing [*846002@user-6000:2] Answer("IAX2/6000-6", "") in new stack steven-sokols-computer*CLI> -- Executing [*846002@user-6000:3] Playback("IAX2/6000-6", "beep") in new stack steven-sokols-computer*CLI> -- Playing 'beep' (language 'en') steven-sokols-computer*CLI> -- Executing [*846002@user-6000:4] ChanSpy("IAX2/6000-6", "SIP/6002|bq") in new stack steven-sokols-computer*CLI> == Spying on channel SIP/6002-01848c00 steven-sokols-computer*CLI> [Jan 8 10:54:21] NOTICE[14912]: app_chanspy.c:202 start_spying: Attaching IAX2/6000-6 to SIP/6002-01848c00 -- Native bridging SIP/6001-01843c00 and SIP/6002-01848c00 ended steven-sokols-computer*CLI> Scheduling destruction of SIP dialog '4568b4dc3c58472a353399a000b020ef@192.168.100.249' in 32000 ms (Method: INVITE) steven-sokols-computer*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.100.171, port 5060 steven-sokols-computer*CLI> Reliably Transmitting (NAT) to 192.168.100.171:5060: BYE sip:6002@192.168.100.171:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.249:5060;branch=z9hG4bK3efcba27;rport From: "Steven Sokol" ;tag=as7173673a To: ;tag=733bbc4c5c0edf54i0 Call-ID: 4568b4dc3c58472a353399a000b020ef@192.168.100.249 CSeq: 104 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- steven-sokols-computer*CLI> == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'SIP/6001-01843c00' in macro 'stdexten' == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'SIP/6001-01843c00' steven-sokols-computer*CLI> Scheduling destruction of SIP dialog 'abd77c78-d7a59d8f@192.168.100.163' in 32000 ms (Method: ACK) steven-sokols-computer*CLI> set_destination: Parsing for address/port to send to steven-sokols-computer*CLI> set_destination: set destination to 192.168.100.163, port 5060 steven-sokols-computer*CLI> Reliably Transmitting (NAT) to 192.168.100.163:5060: BYE sip:6001@192.168.100.163:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.249:5060;branch=z9hG4bK3d7e86a5;rport From: "Paul Barham" ;tag=as48574206 To: "Paul Barham" ;tag=e1309764c9bf5f93o0 Call-ID: abd77c78-d7a59d8f@192.168.100.163 CSeq: 102 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- steven-sokols-computer*CLI> == Done Spying on channel SIP/6002-01848c00 steven-sokols-computer*CLI> <--- SIP read from 192.168.100.163:5060 ---> SIP/2.0 200 OK To: "Paul Barham" ;tag=e1309764c9bf5f93o0 From: "Paul Barham" ;tag=as48574206 Call-ID: abd77c78-d7a59d8f@192.168.100.163 CSeq: 102 BYE Via: SIP/2.0/UDP 192.168.100.249:5060;branch=z9hG4bK3d7e86a5 Server: Linksys/SPA962-5.1.3 Content-Length: 0 <-------------> steven-sokols-computer*CLI> --- (8 headers 0 lines) --- SIP Response message for INCOMING dialog BYE arrived steven-sokols-computer*CLI> <--- SIP read from 192.168.100.171:5060 ---> SIP/2.0 200 OK To: ;tag=733bbc4c5c0edf54i0 From: "Steven Sokol" ;tag=as7173673a Call-ID: 4568b4dc3c58472a353399a000b020ef@192.168.100.249 CSeq: 104 BYE Via: SIP/2.0/UDP 192.168.100.249:5060;branch=z9hG4bK3efcba27 Server: Sipura/SPA841-3.1.3(a) Content-Length: 0 <-------------> steven-sokols-computer*CLI> --- (8 headers 0 lines) --- steven-sokols-computer*CLI> Really destroying SIP dialog '4568b4dc3c58472a353399a000b020ef@192.168.100.249' Method: INVITE steven-sokols-computer*CLI> Really destroying SIP dialog 'abd77c78-d7a59d8f@192.168.100.163' Method: ACK steven-sokols-computer*CLI> exit