Asterisk SVN-trunk-r56126M, 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. ========================================================================= NOTE: This is a development version of Asterisk, and should not be used in production installations. == Parsing '/etc/asterisk/asterisk.conf': Parsing /etc/asterisk/asterisk.conf Found == Parsing '/etc/asterisk/extconfig.conf': Parsing /etc/asterisk/extconfig.conf Found == Binding sippeers to mysql/asterisk/sip_users == Binding sipusers to mysql/asterisk/sip_users == Binding voicemail to mysql/asterisk/voicemail_users Connected to Asterisk SVN-trunk-r56126M currently running on pbx-gr (pid = 19178) pbx-gr*CLI> Verbosity was 4 and is now 5 pbx-gr*CLI> -- Remote UNIX connection pbx-gr*CLI> set debgug 4 pbx-gr*CLI> No such command 'set debug' (type 'help' for help) pbx-gr*CLI> set verbose core set verbose 4 pbx-gr*CLI> Verbosity was 5 and is now 4 pbx-gr*CLI> sip set verbsip set debug 4 pbx-gr*CLI> SIP Debugging enabled pbx-gr*CLI> core set devbbug 4 pbx-gr*CLI> Core debug is at least 4 pbx-gr*CLI> <--- SIP read from 128.139.26.6:53296 ---> INVITE sip:80602@132.64.9.163:5060 SIP/2.0 Via: SIP/2.0/UDP 128.139.26.6:5060;branch=z9hG4bK592174A Remote-Party-ID: "YEHAVI BOURVIN" ;party=calling;screen=no;privacy=off From: "YEHAVI BOURVIN" ;tag=26F00E08-1E38 To: Date: Tue, 27 Feb 2007 05:05:17 GMT Call-ID: F3620912-C55611DB-94AE86A5-87E354FD@128.139.26.6 Supported: 100rel,timer,resource-priority,replaces Min-SE: 1800 Cisco-Guid: 4083128281-3310752219-2223308826-801595720 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER CSeq: 101 INVITE Max-Forwards: 70 Timestamp: 1172552717 Contact: Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 247 v=0 o=CiscoSystemsSIP-GW-UserAgent 6058 5534 IN IP4 128.139.26.6 s=SIP Call cpbx-gr*CLI> =IN IP4 128.139.26.6 t=0 0 m=audio 18082 RTP/AVP 8 101 c=IN IP4 128.139.26.6 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 <-------------> pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 0 [ 42]: INVITE sip:80602@132.64.9.163:5060 SIP/2.0 pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 1 [ 56]: Via: SIP/2.0/UDP 128.139.26.6:5060;branch=z9hG4bK592174A pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 2 [ 94]: Remote-Party-ID: "YEHAVI BOURVIN" ;party=calling;screen=no;privacy=off pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 3 [ 65]: From: "YEHAVI BOURVIN" ;tag=26F00E08-1E38 pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 4 [ 28]: To: pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 5 [ 35]: Date: Tue, 27 Feb 2007 05:05:17 GMT pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 6 [ 57]: Call-ID: F3620912-C55611DB-94AE86A5-87E354FD@128.139.26.6 pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 7 [ 50]: Supported: 100rel,timer,resource-priority,replaces pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 8 [ 13]: Min-SE: 1800 pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 9 [ 54]: Cisco-Guid: 4083128281-3310752219-2223308826-801595720 pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 10 [ 37]: User-Agent: Cisco-SIPGateway/IOS-12.x pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 11 [ 97]: Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 12 [ 16]: CSeq: 101 INVITE pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 13 [ 16]: Max-Forwards: 70 pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 14 [ 21]: Timestamp: 1172552717 pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 15 [ 38]: Contact: pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 16 [ 12]: Expires: 180 pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 17 [ 29]: Allow-Events: telephone-event pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 18 [ 29]: Content-Type: application/sdp pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 19 [ 46]: Content-Disposition: session;handling=required pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 20 [ 19]: Content-Length: 247 pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 21 [ 0]: pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 0 [ 3]: v=0 pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 1 [ 60]: o=CiscoSystemsSIP-GW-UserAgent 6058 5534 IN IP4 128.139.26.6 pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 2 [ 10]: s=SIP Call pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 3 [ 21]: c=IN IP4 128.139.26.6 pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 4 [ 5]: t=0 0 pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 5 [ 27]: m=audio 18082 RTP/AVP 8 101 pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 6 [ 21]: c=IN IP4 128.139.26.6 pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 9 [ 15]: a=fmtp:101 0-16 pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 10 [ 10]: a=ptime:20 pbx-gr*CLI> --- (21 headers 11 lines) --- pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:2842 do_setnat: Setting NAT on RTP to Off pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:2847 do_setnat: Setting NAT on VRTP to Off pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:2852 do_setnat: Setting NAT on UDPTL to Off pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4663 sip_alloc: Allocating new SIP dialog for F3620912-C55611DB-94AE86A5-87E354FD@128.139.26.6 - INVITE (With RTP) pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:15271 handle_request: **** Received INVITE (5) - Command in SIP INVITE pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:1829 parse_sip_options: Begin: parsing SIP "Supported: 100rel,timer,resource-priority,replaces" pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:1837 parse_sip_options: Found SIP option: -100rel- pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:1843 parse_sip_options: Matched SIP option: 100rel pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:1837 parse_sip_options: Found SIP option: -timer- pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:1843 parse_sip_options: Matched SIP option: timer pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:1837 parse_sip_options: Found SIP option: -resource-priority- pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:1843 parse_sip_options: Matched SIP option: resource-priority pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:1837 parse_sip_options: Found SIP option: -replaces- pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:1843 parse_sip_options: Matched SIP option: replaces pbx-gr*CLI> Sending to 128.139.26.6 : 5060 (no NAT) pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:13975 handle_request_invite: Initializing initreq for method INVITE - callid F3620912-C55611DB-94AE86A5-87E354FD@128.139.26.6 pbx-gr*CLI> Using INVITE request as basis request - F3620912-C55611DB-94AE86A5-87E354FD@128.139.26.6 pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: res_config_mysql.c:651 mysql_reconnect: MySQL RealTime: Everything is fine. pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: res_config_mysql.c:139 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_users WHERE name = '89444' pbx-gr*CLI> No user '89444' in SIP users list pbx-gr*CLI> Found peer '128.139.26.6' for '89444' from 128.139.26.6:53296 pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:2842 do_setnat: Setting NAT on RTP to Off pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:2847 do_setnat: Setting NAT on VRTP to Off pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:2852 do_setnat: Setting NAT on UDPTL to Off pbx-gr*CLI> Found RTP audio format 8 pbx-gr*CLI> Found RTP audio format 101 pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:5282 process_sdp: Peer doesn't provide T.38 UDPTL pbx-gr*CLI> Peer audio RTP is at port 128.139.26.6:18082 pbx-gr*CLI> Found description format PCMA for ID 8 pbx-gr*CLI> Found description format telephone-event for ID 101 pbx-gr*CLI> Got unsupported a:fmtp in SDP offer pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:5524 process_sdp: T38 state changed to 0 on channel pbx-gr*CLI> Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) pbx-gr*CLI> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) pbx-gr*CLI> Peer audio RTP is at port 128.139.26.6:18082 pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:5615 process_sdp: We're settling with these formats: 0x8 (alaw) pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:14056 handle_request_invite: Checking SIP call limits for device gr-pbx-link pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:3305 update_call_counter: Updating call counter for incoming call pbx-gr*CLI> Looking for 80602 in huji-remote-gr (domain 132.64.9.163) pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4126 sip_new: *** Our native formats are 0x8 (alaw) pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4127 sip_new: *** Joint capabilities are 0x8 (alaw) pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4128 sip_new: *** Our capabilities are 0x8 (alaw) pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4129 sip_new: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4159 sip_new: This channel will not be able to handle video. pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:8478 build_route: build_route: Contact hop: pbx-gr*CLI> list_route: hop: pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:14131 handle_request_invite: SIP/gr-pbx-link-09607500: New call is still down.... Trying... pbx-gr*CLI> <--- Transmitting (no NAT) to 128.139.26.6:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 128.139.26.6:5060;branch=z9hG4bK592174A;received=128.139.26.6 From: "YEHAVI BOURVIN" ;tag=26F00E08-1E38 To: Call-ID: F3620912-C55611DB-94AE86A5-87E354FD@128.139.26.6 CSeq: 101 INVITE User-Agent: Asterisk PBX SVN-trunk-r56126M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: devicestate.c:444 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/gr-pbx-link-09607500 pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: devicestate.c:302 ast_device_state: No provider found, checking channel drivers for SIP - gr-pbx-link pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:15897 sip_devicestate: Checking device state for peer gr-pbx-link pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: res_config_mysql.c:651 mysql_reconnect: MySQL RealTime: Everything is fine. pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: res_config_mysql.c:139 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_users WHERE name = 'gr-pbx-link' pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: pbx.c:1693 pbx_extension_helper: Launching 'Set' pbx-gr*CLI> -- Executing [80602@huji-remote-gr:1] Set("SIP/gr-pbx-link-09607500", "_To=80602") in new stack pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: pbx.c:1543 pbx_substitute_variables_helper_full: Function result is '89444' pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: pbx.c:1693 pbx_extension_helper: Launching 'Set' pbx-gr*CLI> -- Executing [80602@huji-remote-gr:2] Set("SIP/gr-pbx-link-09607500", "_From=89444") in new stack pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: pbx.c:1693 pbx_extension_helper: Launching 'Set' pbx-gr*CLI> -- Executing [80602@huji-remote-gr:3] Set("SIP/gr-pbx-link-09607500", "DB(80602/LastCaller)=89444") in new stack pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: devicestate.c:428 do_state_change: Changing state for SIP/gr-pbx-link - state 4 (Invalid) pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: app_queue.c:568 changethread: Device 'SIP/gr-pbx-link' changed to state '4' (Invalid) but we don't care because they're not a member of any queue. pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: pbx.c:1693 pbx_extension_helper: Launching 'Set' pbx-gr*CLI> -- Executing [80602@huji-remote-gr:4] Set("SIP/gr-pbx-link-09607500", "DB(89444/LastCalled)=80602") in new stack pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: db.c:198 ast_db_get: Unable to find key 'MoreLinesToRing' in family '80602' pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: func_db.c:73 function_db_read: DB: 80602/MoreLinesToRing not found in database. pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: pbx.c:1543 pbx_substitute_variables_helper_full: Function result is '' pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: pbx.c:1693 pbx_extension_helper: Launching 'Set' pbx-gr*CLI> -- Executing [80602@huji-remote-gr:5] Set("SIP/gr-pbx-link-09607500", "aEXTEN=") in new stack pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: pbx.c:1693 pbx_extension_helper: Launching 'Dial' pbx-gr*CLI> -- Executing [80602@huji-remote-gr:6] Dial("SIP/gr-pbx-link-09607500", "SIP/80602|20|") in new stack pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:15966 sip_request_call: Asked to create a SIP channel with formats: 0x8 (alaw) pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4663 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:2897 create_addr_from_peer: Our T38 capability (3856) pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:2842 do_setnat: Setting NAT on RTP to Off pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:2852 do_setnat: Setting NAT on UDPTL to Off pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:1783 obproxy_get: OBPROXY: Not applying OBproxy to this call pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4126 sip_new: *** Our native formats are 0x8 (alaw) pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4127 sip_new: *** Joint capabilities are 0x8 (alaw) pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4128 sip_new: *** Our capabilities are 0x8 (alaw) pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4129 sip_new: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4131 sip_new: *** Our preferred formats from the incoming channel are 0x8 (alaw) pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4159 sip_new: This channel will not be able to handle video. pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: rtp.c:1677 ast_rtp_make_compatible: Seeded SDP of 'SIP/80602-0960b470' with that of 'SIP/gr-pbx-link-09607500' pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: channel.c:3211 ast_channel_inherit_variables: Not copying variable STACK-huji-remote-gr-80602-6. pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: channel.c:3211 ast_channel_inherit_variables: Not copying variable aEXTEN. pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: channel.c:3211 ast_channel_inherit_variables: Not copying variable STACK-huji-remote-gr-80602-5. pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: channel.c:3211 ast_channel_inherit_variables: Not copying variable STACK-huji-remote-gr-80602-4. pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: channel.c:3211 ast_channel_inherit_variables: Not copying variable STACK-huji-remote-gr-80602-3. pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: channel.c:3198 ast_channel_inherit_variables: Copying soft-transferable variable From. pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: channel.c:3211 ast_channel_inherit_variables: Not copying variable STACK-huji-remote-gr-80602-2. pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: channel.c:3198 ast_channel_inherit_variables: Copying soft-transferable variable To. pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: channel.c:3211 ast_channel_inherit_variables: Not copying variable STACK-huji-remote-gr-80602-1. pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: channel.c:3211 ast_channel_inherit_variables: Not copying variable SIPCALLID. pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: channel.c:3211 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: channel.c:3211 ast_channel_inherit_variables: Not copying variable SIPURI. pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:3122 sip_call: Outgoing Call for 80602 pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:3305 update_call_counter: Updating call counter for outgoing call pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:3140 sip_call: Our T38 capability (3856), joint T38 capability (3856) pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:6674 add_sdp: ** Our capability: 0x8 (alaw) Video flag: False Text flag: False pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:6676 add_sdp: ** Our prefcodec: 0x8 (alaw) pbx-gr*CLI> Audio is at 132.64.9.163 port 18986 pbx-gr*CLI> Adding codec 0x8 (alaw) to SDP pbx-gr*CLI> Adding non-codec 0x1 (telephone-event) to SDP pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:6837 add_sdp: -- Done with adding codecs to SDP pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:6897 add_sdp: Done building SDP. Settling with this capability: 0x8 (alaw) pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:1718 initialize_initreq: Initializing initreq for method INVITE - callid 0203bcd16026f51d4c0e9821089c0e96@132.64.9.163 pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 0 [ 56]: INVITE sip:80602@132.64.4.141:2051;line=kbq2n6ok SIP/2.0 pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 1 [ 63]: Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK65bb7435;rport pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 2 [ 16]: Max-Forwards: 70 pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 3 [ 62]: From: "YEHAVI BOURVIN" ;tag=as4e147432 pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 4 [ 47]: To: pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 5 [ 33]: Contact: pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 6 [ 54]: Call-ID: 0203bcd16026f51d4c0e9821089c0e96@132.64.9.163 pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 7 [ 16]: CSeq: 102 INVITE pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 8 [ 42]: User-Agent: Asterisk PBX SVN-trunk-r56126M pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 9 [ 35]: Date: Tue, 27 Feb 2007 05:01:15 GMT pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 10 [ 66]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 11 [ 19]: Supported: replaces pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 12 [ 29]: Content-Type: application/sdp pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 13 [ 19]: Content-Length: 240 pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 14 [ 0]: pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 0 [ 3]: v=0 pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 1 [ 38]: o=root 19178 19178 IN IP4 132.64.9.163 pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 2 [ 9]: s=session pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 3 [ 21]: c=IN IP4 132.64.9.163 pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 4 [ 5]: t=0 0 pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 5 [ 27]: m=audio 18986 RTP/AVP 8 101 pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 7 [ 33]: a=rtpmap:101 telephone-event/8000 pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 8 [ 15]: a=fmtp:101 0-16 pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 9 [ 25]: a=silenceSupp:off - - - - pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 10 [ 10]: a=ptime:20 pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 11 [ 10]: a=sendrecv pbx-gr*CLI> Reliably Transmitting (no NAT) to 132.64.4.141:2051: INVITE sip:80602@132.64.4.141:2051;line=kbq2n6ok SIP/2.0 Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK65bb7435;rport Max-Forwards: 70 From: "YEHAVI BOURVIN" ;tag=as4e147432 To: Contact: Call-ID: 0203bcd16026f51d4c0e9821089c0e96@132.64.9.163 CSeq: 102 INVITE User-Agent: Asterisk PBX SVN-trunk-r56126M Date: Tue, 27 Feb 2007 05:01:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 240 v=0 o=root 19178 19178 IN IP4 132.64.9.163 s=session c=IN IP4 132.64.9.163 t=0 0 m=audio 18986 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- pbx-gr*CLI> [Feb 27 07:01:15] DEBUG[19189]: chan_sip.c:2115 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #8898 pbx-gr*CLI> -- Called 80602 pbx-gr*CLI> <--- SIP read from 132.64.4.141:2051 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK65bb7435;rport=5060 From: "YEHAVI BOURVIN" ;tag=as4e147432 To: "80602" ;tag=1fnx3usnpb Call-ID: 0203bcd16026f51d4c0e9821089c0e96@132.64.9.163 CSeq: 102 INVITE Contact: ;flow-id=1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0 <-------------> pbx-gr*CLI> [Feb 27 07:01:16] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 0 [ 19]: SIP/2.0 180 Ringing pbx-gr*CLI> [Feb 27 07:01:16] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 1 [ 68]: Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK65bb7435;rport=5060 pbx-gr*CLI> [Feb 27 07:01:16] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 2 [ 62]: From: "YEHAVI BOURVIN" ;tag=as4e147432 pbx-gr*CLI> [Feb 27 07:01:16] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 3 [ 70]: To: "80602" ;tag=1fnx3usnpb pbx-gr*CLI> [Feb 27 07:01:16] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 4 [ 54]: Call-ID: 0203bcd16026f51d4c0e9821089c0e96@132.64.9.163 pbx-gr*CLI> [Feb 27 07:01:16] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 5 [ 16]: CSeq: 102 INVITE pbx-gr*CLI> [Feb 27 07:01:16] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 6 [ 62]: Contact: ;flow-id=1 pbx-gr*CLI> [Feb 27 07:01:16] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 7 [ 88]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO pbx-gr*CLI> [Feb 27 07:01:16] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 8 [ 31]: Allow-Events: talk, hold, refer pbx-gr*CLI> [Feb 27 07:01:16] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 9 [ 17]: Content-Length: 0 pbx-gr*CLI> [Feb 27 07:01:16] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 10 [ 0]: pbx-gr*CLI> --- (10 headers 0 lines) --- pbx-gr*CLI> [Feb 27 07:01:16] DEBUG[19189]: chan_sip.c:2268 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #8898 - INVITE (got response) pbx-gr*CLI> [Feb 27 07:01:16] DEBUG[19189]: chan_sip.c:2277 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '0203bcd16026f51d4c0e9821089c0e96@132.64.9.163' Request 102: Found pbx-gr*CLI> [Feb 27 07:01:16] DEBUG[19189]: chan_sip.c:12287 handle_response_invite: SIP response 180 to standard invite pbx-gr*CLI> [Feb 27 07:01:16] DEBUG[19189]: devicestate.c:444 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/80602-0960b470 pbx-gr*CLI> [Feb 27 07:01:16] DEBUG[19189]: devicestate.c:302 ast_device_state: No provider found, checking channel drivers for SIP - 80602 pbx-gr*CLI> [Feb 27 07:01:16] DEBUG[19189]: chan_sip.c:15897 sip_devicestate: Checking device state for peer 80602 pbx-gr*CLI> [Feb 27 07:01:16] DEBUG[19189]: devicestate.c:428 do_state_change: Changing state for SIP/80602 - state 1 (Not in use) pbx-gr*CLI> [Feb 27 07:01:16] DEBUG[19189]: devicestate.c:302 ast_device_state: No provider found, checking channel drivers for SIP - 80602 pbx-gr*CLI> [Feb 27 07:01:16] DEBUG[19189]: chan_sip.c:15897 sip_devicestate: Checking device state for peer 80602 pbx-gr*CLI> [Feb 27 07:01:16] DEBUG[19189]: app_queue.c:568 changethread: Device 'SIP/80602' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. pbx-gr*CLI> -- SIP/80602-0960b470 is ringing pbx-gr*CLI> [Feb 27 07:01:16] DEBUG[19189]: rtp.c:1607 ast_rtp_early_bridge: Setting early bridge SDP of 'SIP/gr-pbx-link-09607500' with that of 'SIP/80602-0960b470' pbx-gr*CLI> <--- Transmitting (no NAT) to 128.139.26.6:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 128.139.26.6:5060;branch=z9hG4bK592174A;received=128.139.26.6 From: "YEHAVI BOURVIN" ;tag=26F00E08-1E38 To: ;tag=as29e8d94e Call-ID: F3620912-C55611DB-94AE86A5-87E354FD@128.139.26.6 CSeq: 101 INVITE User-Agent: Asterisk PBX SVN-trunk-r56126M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> pbx-gr*CLI> <--- SIP read from 132.64.4.141:2051 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK65bb7435;rport=5060 From: "YEHAVI BOURVIN" ;tag=as4e147432 To: "80602" ;tag=1fnx3usnpb Call-ID: 0203bcd16026f51d4c0e9821089c0e96@132.64.9.163 CSeq: 102 INVITE Contact: ;flow-id=1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0 <-------------> [Feb 27 07:01:16] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 0 [ 19]: SIP/2.0 180 Ringing [Feb 27 07:01:16] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 1 [ 68]: Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK65bb7435;rport=5060 [Feb 27 07:01:16] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 2 [ 62]: From: "YEHAVI BOURVIN" ;tag=as4e147432 [Feb 27 07:01:16] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 3 [ 70]: To: "80602" ;tag=1fnx3usnpb [Feb 27 07:01:16] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 4 [ 54]: Call-ID: 0203bcd16026f51d4c0e9821089c0e96@132.64.9.163 [Feb 27 07:01:16] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 5 [ 16]: CSeq: 102 INVITE [Feb 27 07:01:16] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 6 [ 62]: Contact: ;flow-id=1 [Feb 27 07:01:16] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 7 [ 88]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO [Feb 27 07:01:16] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 8 [ 31]: Allow-Events: talk, hold, refer [Feb 27 07:01:16] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 9 [ 17]: Content-Length: 0 [Feb 27 07:01:16] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 10 [ 0]: --- (10 headers 0 lines) --- [Feb 27 07:01:16] DEBUG[19189]: chan_sip.c:2277 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '0203bcd16026f51d4c0e9821089c0e96@132.64.9.163' Request 102: Found [Feb 27 07:01:16] DEBUG[19189]: chan_sip.c:12287 handle_response_invite: SIP response 180 to standard invite pbx-gr*CLI> -- SIP/80602-0960b470 is ringing [Feb 27 07:01:16] DEBUG[19189]: rtp.c:1607 ast_rtp_early_bridge: Setting early bridge SDP of 'SIP/gr-pbx-link-09607500' with that of 'SIP/80602-0960b470' pbx-gr*CLI> <--- SIP read from 132.64.4.141:2051 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK65bb7435;rport=5060 From: "YEHAVI BOURVIN" ;tag=as4e147432 To: "80602" ;tag=1fnx3usnpb Call-ID: 0203bcd16026f51d4c0e9821089c0e96@132.64.9.163 CSeq: 102 INVITE Contact: ;flow-id=1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0 <-------------> [Feb 27 07:01:17] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 0 [ 19]: SIP/2.0 180 Ringing [Feb 27 07:01:17] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 1 [ 68]: Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK65bb7435;rport=5060 [Feb 27 07:01:17] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 2 [ 62]: From: "YEHAVI BOURVIN" ;tag=as4e147432 [Feb 27 07:01:17] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 3 [ 70]: To: "80602" ;tag=1fnx3usnpb [Feb 27 07:01:17] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 4 [ 54]: Call-ID: 0203bcd16026f51d4c0e9821089c0e96@132.64.9.163 [Feb 27 07:01:17] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 5 [ 16]: CSeq: 102 INVITE [Feb 27 07:01:17] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 6 [ 62]: Contact: ;flow-id=1 [Feb 27 07:01:17] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 7 [ 88]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO [Feb 27 07:01:17] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 8 [ 31]: Allow-Events: talk, hold, refer [Feb 27 07:01:17] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 9 [ 17]: Content-Length: 0 [Feb 27 07:01:17] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 10 [ 0]: --- (10 headers 0 lines) --- [Feb 27 07:01:17] DEBUG[19189]: chan_sip.c:2277 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '0203bcd16026f51d4c0e9821089c0e96@132.64.9.163' Request 102: Found [Feb 27 07:01:17] DEBUG[19189]: chan_sip.c:12287 handle_response_invite: SIP response 180 to standard invite pbx-gr*CLI> -- SIP/80602-0960b470 is ringing [Feb 27 07:01:17] DEBUG[19189]: rtp.c:1607 ast_rtp_early_bridge: Setting early bridge SDP of 'SIP/gr-pbx-link-09607500' with that of 'SIP/80602-0960b470' pbx-gr*CLI> <--- SIP read from 132.64.128.105:43590 ---> <-------------> [Feb 27 07:01:18] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 0 [ 0]: [Feb 27 07:01:18] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 0 [ 0]: --- (0 headers 1 lines) --- pbx-gr*CLI> <--- SIP read from 132.64.4.141:2051 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK65bb7435;rport=5060 From: "YEHAVI BOURVIN" ;tag=as4e147432 To: "80602" ;tag=1fnx3usnpb Call-ID: 0203bcd16026f51d4c0e9821089c0e96@132.64.9.163 CSeq: 102 INVITE Contact: ;flow-id=1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0 <-------------> [Feb 27 07:01:19] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 0 [ 19]: SIP/2.0 180 Ringing [Feb 27 07:01:19] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 1 [ 68]: Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK65bb7435;rport=5060 [Feb 27 07:01:19] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 2 [ 62]: From: "YEHAVI BOURVIN" ;tag=as4e147432 [Feb 27 07:01:19] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 3 [ 70]: To: "80602" ;tag=1fnx3usnpb [Feb 27 07:01:19] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 4 [ 54]: Call-ID: 0203bcd16026f51d4c0e9821089c0e96@132.64.9.163 [Feb 27 07:01:19] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 5 [ 16]: CSeq: 102 INVITE [Feb 27 07:01:19] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 6 [ 62]: Contact: ;flow-id=1 [Feb 27 07:01:19] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 7 [ 88]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO [Feb 27 07:01:19] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 8 [ 31]: Allow-Events: talk, hold, refer [Feb 27 07:01:19] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 9 [ 17]: Content-Length: 0 [Feb 27 07:01:19] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 10 [ 0]: --- (10 headers 0 lines) --- [Feb 27 07:01:19] DEBUG[19189]: chan_sip.c:2277 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '0203bcd16026f51d4c0e9821089c0e96@132.64.9.163' Request 102: Found [Feb 27 07:01:19] DEBUG[19189]: chan_sip.c:12287 handle_response_invite: SIP response 180 to standard invite pbx-gr*CLI> -- SIP/80602-0960b470 is ringing [Feb 27 07:01:19] DEBUG[19189]: rtp.c:1607 ast_rtp_early_bridge: Setting early bridge SDP of 'SIP/gr-pbx-link-09607500' with that of 'SIP/80602-0960b470' pbx-gr*CLI> [Feb 27 07:01:20] DEBUG[19189]: rtp.c:941 ast_rtcp_read: Got RTCP report of 52 bytes pbx-gr*CLI> <--- SIP read from 132.64.4.141:2051 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK65bb7435;rport=5060 From: "YEHAVI BOURVIN" ;tag=as4e147432 To: "80602" ;tag=1fnx3usnpb Call-ID: 0203bcd16026f51d4c0e9821089c0e96@132.64.9.163 CSeq: 102 INVITE Contact: ;flow-id=1 User-Agent: snom320/6.2.2 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Content-Type: application/sdp Content-Length: 327 v=0 o=root 1787986285 1787986286 IN IP4 132.64.4.141 s=call c=IN IP4 132.64.4.141 t=0 0 m=audio 49314 RTP/AVP 8 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:FKUkTJdhQXu+fhL/5a7Z3c6mpSO3tuKR92f2fag3 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=encryption:optional a=sendrecv <-------------> [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 0 [ 14]: SIP/2.0 200 Ok [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 1 [ 68]: Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK65bb7435;rport=5060 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 2 [ 62]: From: "YEHAVI BOURVIN" ;tag=as4e147432 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 3 [ 70]: To: "80602" ;tag=1fnx3usnpb [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 4 [ 54]: Call-ID: 0203bcd16026f51d4c0e9821089c0e96@132.64.9.163 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 5 [ 16]: CSeq: 102 INVITE [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 6 [ 62]: Contact: ;flow-id=1 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 7 [ 25]: User-Agent: snom320/6.2.2 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 8 [ 88]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 9 [ 31]: Allow-Events: talk, hold, refer [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 10 [ 44]: Supported: timer, 100rel, replaces, callerid [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 11 [ 29]: Content-Type: application/sdp [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 12 [ 19]: Content-Length: 327 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 13 [ 0]: [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 0 [ 3]: v=0 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 1 [ 48]: o=root 1787986285 1787986286 IN IP4 132.64.4.141 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 2 [ 6]: s=call [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 3 [ 21]: c=IN IP4 132.64.4.141 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 4 [ 5]: t=0 0 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 5 [ 27]: m=audio 49314 RTP/AVP 8 101 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 6 [ 82]: a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:FKUkTJdhQXu+fhL/5a7Z3c6mpSO3tuKR92f2fag3 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 7 [ 20]: a=rtpmap:8 pcma/8000 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 9 [ 15]: a=fmtp:101 0-16 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 10 [ 10]: a=ptime:20 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 11 [ 21]: a=encryption:optional [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 12 [ 10]: a=sendrecv --- (13 headers 13 lines) --- [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:2218 __sip_ack: Acked pending invite 102 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:2234 __sip_ack: Stopping retransmission on '0203bcd16026f51d4c0e9821089c0e96@132.64.9.163' of Request 102: Match Found [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:12287 handle_response_invite: SIP response 200 to standard invite Found RTP audio format 8 Found RTP audio format 101 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:5282 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 132.64.4.141:49314 Got unsupported a:crypto in SDP offer Found description format pcma for ID 8 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:5524 process_sdp: T38 state changed to 0 on channel SIP/80602-0960b470 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=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 132.64.4.141:49314 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:5615 process_sdp: We're settling with these formats: 0x8 (alaw) [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:5622 process_sdp: We have an owner, now see if we need to change this call [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:3305 update_call_counter: Updating call counter for outgoing call --- set_address_from_contact host '132.64.4.141' [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:8478 build_route: build_route: Contact hop: ;flow-id=1 pbx-gr*CLI> list_route: hop: [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:6050 reqprep: Strict routing enforced for session 0203bcd16026f51d4c0e9821089c0e96@132.64.9.163 set_destination: Parsing for address/port to send to set_destination: set destination to 132.64.4.141, port 2051 Transmitting (no NAT) to 132.64.4.141:2051: ACK sip:80602@132.64.4.141:2051;line=kbq2n6ok SIP/2.0 Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK7ba7061d;rport Max-Forwards: 70 From: "YEHAVI BOURVIN" ;tag=as4e147432 To: ;tag=1fnx3usnpb Contact: Call-ID: 0203bcd16026f51d4c0e9821089c0e96@132.64.9.163 CSeq: 102 ACK User-Agent: Asterisk PBX SVN-trunk-r56126M Content-Length: 0 --- pbx-gr*CLI> -- Call on SIP/80602-0960b470 left from hold [Feb 27 07:01:20] DEBUG[19189]: devicestate.c:444 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/80602-0960b470 -- SIP/80602-0960b470 answered SIP/gr-pbx-link-09607500 [Feb 27 07:01:20] DEBUG[19189]: rtp.c:1607 ast_rtp_early_bridge: Setting early bridge SDP of 'SIP/gr-pbx-link-09607500' with that of 'SIP/80602-0960b470' [Feb 27 07:01:20] DEBUG[19189]: devicestate.c:444 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/gr-pbx-link-09607500 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:3761 sip_answer: SIP answering channel: SIP/gr-pbx-link-09607500 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:6952 transmit_response_with_sdp: Setting framing from config on incoming call [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:6674 add_sdp: ** Our capability: 0x8 (alaw) Video flag: True Text flag: True [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:6676 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 132.64.9.163 port 61938 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:6837 add_sdp: -- Done with adding codecs to SDP [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:6897 add_sdp: Done building SDP. Settling with this capability: 0x8 (alaw) <--- Reliably Transmitting (no NAT) to 128.139.26.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 128.139.26.6:5060;branch=z9hG4bK592174A;received=128.139.26.6 From: "YEHAVI BOURVIN" ;tag=26F00E08-1E38 To: ;tag=as29e8d94e Call-ID: F3620912-C55611DB-94AE86A5-87E354FD@128.139.26.6 CSeq: 101 INVITE User-Agent: Asterisk PBX SVN-trunk-r56126M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Spbx-gr*CLI> upported: replaces Contact: Content-Type: application/sdp Content-Length: 240 v=0 o=root 19178 19178 IN IP4 132.64.9.163 s=session c=IN IP4 132.64.9.163 t=0 0 m=audio 61938 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:2115 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #8900 [Feb 27 07:01:20] DEBUG[19189]: devicestate.c:302 ast_device_state: No provider found, checking channel drivers for SIP - 80602 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:15897 sip_devicestate: Checking device state for peer 80602 [Feb 27 07:01:20] DEBUG[19189]: devicestate.c:428 do_state_change: Changing state for SIP/80602 - state 1 (Not in use) [Feb 27 07:01:20] DEBUG[19189]: devicestate.c:302 ast_device_state: No provider found, checking channel drivers for SIP - 80602 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:15897 sip_devicestate: Checking device state for peer 80602 [Feb 27 07:01:20] DEBUG[19189]: devicestate.c:302 ast_device_state: No provider found, checking channel drivers for SIP - gr-pbx-link [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:15897 sip_devicestate: Checking device state for peer gr-pbx-link [Feb 27 07:01:20] DEBUG[19189]: res_config_mysql.c:651 mysql_reconnect: MySQL RealTime: Everything is fine. [Feb 27 07:01:20] DEBUG[19189]: res_config_mysql.c:139 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_users WHERE name = 'gr-pbx-link' pbx-gr*CLI> [Feb 27 07:01:20] DEBUG[19189]: devicestate.c:428 do_state_change: Changing state for SIP/gr-pbx-link - state 4 (Invalid) pbx-gr*CLI> [Feb 27 07:01:20] DEBUG[19189]: app_queue.c:568 changethread: Device 'SIP/80602' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. pbx-gr*CLI> [Feb 27 07:01:20] DEBUG[19189]: app_queue.c:568 changethread: Device 'SIP/gr-pbx-link' changed to state '4' (Invalid) but we don't care because they're not a member of any queue. pbx-gr*CLI> <--- SIP read from 128.139.26.6:53296 ---> ACK sip:80602@132.64.9.163:5060 SIP/2.0 Via: SIP/2.0/UDP 128.139.26.6:5060;branch=z9hG4bK593223 From: "YEHAVI BOURVIN" ;tag=26F00E08-1E38 To: ;tag=as29e8d94e Date: Tue, 27 Feb 2007 05:05:17 GMT Call-ID: F3620912-C55611DB-94AE86A5-87E354FD@128.139.26.6 Max-Forwards: 70 CSeq: 101 ACK Allow-Events: telephone-event Content-Length: 0 <-------------> [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 0 [ 39]: ACK sip:80602@132.64.9.163:5060 SIP/2.0 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 1 [ 55]: Via: SIP/2.0/UDP 128.139.26.6:5060;branch=z9hG4bK593223 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 2 [ 65]: From: "YEHAVI BOURVIN" ;tag=26F00E08-1E38 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 3 [ 43]: To: ;tag=as29e8d94e [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 4 [ 35]: Date: Tue, 27 Feb 2007 05:05:17 GMT [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 5 [ 57]: Call-ID: F3620912-C55611DB-94AE86A5-87E354FD@128.139.26.6 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 6 [ 16]: Max-Forwards: 70 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 7 [ 13]: CSeq: 101 ACK [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 8 [ 29]: Allow-Events: telephone-event [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 9 [ 17]: Content-Length: 0 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 10 [ 0]: --- (10 headers 0 lines) --- [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:15271 handle_request: **** Received ACK (6) - Command in SIP ACK pbx-gr*CLI> [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:2223 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #8900 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:2234 __sip_ack: Stopping retransmission on 'F3620912-C55611DB-94AE86A5-87E354FD@128.139.26.6' of Response 101: Match Found pbx-gr*CLI> -- Native bridging SIP/gr-pbx-link-09607500 and SIP/80602-0960b470 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:17627 sip_set_rtp_peer: Sending reinvite on SIP 'F3620912-C55611DB-94AE86A5-87E354FD@128.139.26.6' - It's audio soon redirected to IP 132.64.4.141 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:6050 reqprep: Strict routing enforced for session F3620912-C55611DB-94AE86A5-87E354FD@128.139.26.6 set_destination: Parsing for address/port to send to set_destination: set destination to 128.139.26.6, port 5060 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:6674 add_sdp: ** Our capability: 0x8 (alaw) Video flag: True Text flag: True [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:6676 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 132.64.9.163 port 61938 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:6837 add_sdp: -- Done with adding codecs to SDP [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:6897 add_sdp: Done building SDP. Settling with this capability: 0x8 (alaw) [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:1716 initialize_initreq: Initializing already initialized SIP dialog F3620912-C55611DB-94AE86A5-87E354FD@128.139.26.6 (presumably reinvite) [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 0 [ 42]: INVITE sip:89444@128.139.26.6:5060 SIP/2.0 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 1 [ 63]: Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK493bb0d9;rport [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 2 [ 16]: Max-Forwards: 70 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 3 [ 45]: From: ;tag=as29e8d94e [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 4 [ 63]: To: "YEHAVI BOURVIN" ;tag=26F00E08-1E38 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 5 [ 33]: Contact: [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 6 [ 57]: Call-ID: F3620912-C55611DB-94AE86A5-87E354FD@128.139.26.6 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 7 [ 16]: CSeq: 102 INVITE [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 8 [ 42]: User-Agent: Asterisk PBX SVN-trunk-r56126M [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 9 [ 66]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 10 [ 19]: Supported: replaces pbx-gr*CLI> [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 12 [ 29]: Content-Type: application/sdp [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 13 [ 19]: Content-Length: 240 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 14 [ 0]: [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 0 [ 3]: v=0 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 1 [ 38]: o=root 19178 19179 IN IP4 132.64.4.141 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 2 [ 9]: s=session [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 3 [ 21]: c=IN IP4 132.64.4.141 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 4 [ 5]: t=0 0 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 5 [ 27]: m=audio 49314 RTP/AVP 8 101 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 7 [ 33]: a=rtpmap:101 telephone-event/8000 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 8 [ 15]: a=fmtp:101 0-16 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 9 [ 25]: a=silenceSupp:off - - - - [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 10 [ 10]: a=ptime:20 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 11 [ 10]: a=sendrecv Reliably Transmitting (no NAT) to 128.139.26.6:5060: INVITE sip:89444@128.139.26.6:5060 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK493bb0d9;rport Max-Forwards: 70 From: ;tag=as29e8d94e To: "YEHAVI BOURVIN" ;tag=26F00E08-1E38 Contact: Call-ID: F3620912-C55611DB-94AE86A5-87E354FD@128.139.26.6 CSeq: 102 INVITE User-Agent: Asterisk PBX SVN-trunk-r56126M 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: 240 v=0 o=root 19178 19179 IN IP4 132.64.4.141 s=session c=IN IP4 132.64.4.141 t=0 0 m=audio 49314 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:2115 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #8902 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:17627 sip_set_rtp_peer: Sending reinvite on SIP '0203bcd16026f51d4c0e9821089c0e96@132.64.9.163' - It's audio soon redirected to IP 128.139.26.6 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:6050 reqprep: Strict routing enforced for session 0203bcd16026f51d4c0e9821089c0e96@132.64.9.163 set_destination: Parsing for address/port to send to set_destination: set destination to 132.64.4.141, port 2051 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:6674 add_sdp: ** Our capability: 0x8 (alaw) Video flag: True Text flag: True [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:6676 add_sdp: ** Our prefcodec: 0x8 (alaw) Audio is at 132.64.9.163 port 18986 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:6837 add_sdp: -- Done with adding codecs to SDP [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:6897 add_sdp: Done building SDP. Settling with this capability: 0x8 (alaw) [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:1716 initialize_initreq: Initializing already initialized SIP dialog 0203bcd16026f51d4c0e9821089c0e96@132.64.9.163 (presumably reinvite) [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 0 [ 56]: INVITE sip:80602@132.64.4.141:2051;line=kbq2n6ok SIP/2.0 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 1 [ 63]: Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK6a1c94ad;rport [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 2 [ 16]: Max-Forwards: 70 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 3 [ 62]: From: "YEHAVI BOURVIN" ;tag=as4e147432 pbx-gr*CLI> [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 4 [ 62]: To: ;tag=1fnx3usnpb [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 5 [ 33]: Contact: [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 6 [ 54]: Call-ID: 0203bcd16026f51d4c0e9821089c0e96@132.64.9.163 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 7 [ 16]: CSeq: 103 INVITE [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 8 [ 42]: User-Agent: Asterisk PBX SVN-trunk-r56126M pbx-gr*CLI> [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 9 [ 66]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 10 [ 19]: Supported: replaces [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 12 [ 29]: Content-Type: application/sdp [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 13 [ 19]: Content-Length: 240 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 14 [ 0]: [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 0 [ 3]: v=0 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 1 [ 38]: o=root 19178 19179 IN IP4 128.139.26.6 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 2 [ 9]: s=session [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 3 [ 21]: c=IN IP4 128.139.26.6 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 4 [ 5]: t=0 0 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 5 [ 27]: m=audio 18082 RTP/AVP 8 101 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 7 [ 33]: a=rtpmap:101 telephone-event/8000 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 8 [ 15]: a=fmtp:101 0-16 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 9 [ 25]: a=silenceSupp:off - - - - [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 10 [ 10]: a=ptime:20 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 11 [ 10]: a=sendrecv Reliably Transmitting (no NAT) to 132.64.4.141:2051: INVITE sip:80602@132.64.4.141:2051;line=kbq2n6ok SIP/2.0 Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK6a1c94ad;rport Max-Forwards: 70 From: "YEHAVI BOURVIN" ;tag=as4e147432 To: ;tag=1fnx3usnpb Contact: Call-ID: 0203bcd16026f51d4c0e9821089c0e96@132.64.9.163 CSeq: 103 INVITE User-Agent: Asterisk PBX SVN-trunk-r56126M 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: 240 v=0 o=root 19178 19179 IN IP4 128.139.26.6 s=session c=IN IP4 128.139.26.6 tpbx-gr*CLI> =0 0 m=audio 18082 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:2115 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #8903 [Feb 27 07:01:20] DEBUG[19189]: rtp.c:2792 ast_rtp_write: Ooh, format changed from unknown to alaw [Feb 27 07:01:20] DEBUG[19189]: rtp.c:2809 ast_rtp_write: Created smoother: format: 8 ms: 20 len: 160 pbx-gr*CLI> [Feb 27 07:01:20] DEBUG[19189]: rtp.c:2792 ast_rtp_write: Ooh, format changed from unknown to alaw [Feb 27 07:01:20] DEBUG[19189]: rtp.c:2809 ast_rtp_write: Created smoother: format: 8 ms: 20 len: 160 pbx-gr*CLI> [Feb 27 07:01:20] DEBUG[19189]: rtp.c:941 ast_rtcp_read: Got RTCP report of 72 bytes pbx-gr*CLI> <--- SIP read from 128.139.26.6:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK493bb0d9;rport From: ;tag=as29e8d94e To: "YEHAVI BOURVIN" ;tag=26F00E08-1E38 Date: Tue, 27 Feb 2007 05:05:22 GMT Call-ID: F3620912-C55611DB-94AE86A5-87E354FD@128.139.26.6 Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow-Events: telephone-event Remote-Party-ID: "YEHAVI BOURVIN" ;party=called;screen=no;privacy=off Content-Length: 0 <-------------> [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 0 [ 18]: SIP/2.0 100 Trying [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 1 [ 63]: Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK493bb0d9;rport [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 2 [ 45]: From: ;tag=as29e8d94e [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 3 [ 63]: To: "YEHAVI BOURVIN" ;tag=26F00E08-1E38 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 4 [ 35]: Date: Tue, 27 Feb 2007 05:05:22 GMT [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 5 [ 57]: Call-ID: F3620912-C55611DB-94AE86A5-87E354FD@128.139.26.6 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 6 [ 33]: Server: Cisco-SIPGateway/IOS-12.x [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 7 [ 16]: CSeq: 102 INVITE [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 8 [ 29]: Allow-Events: telephone-event [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 9 [ 93]: Remote-Party-ID: "YEHAVI BOURVIN" ;party=called;screen=no;privacy=off [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 10 [ 17]: Content-Length: 0 pbx-gr*CLI> [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 11 [ 0]: --- (11 headers 0 lines) --- [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:2268 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #8902 - INVITE (got response) [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:2277 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on 'F3620912-C55611DB-94AE86A5-87E354FD@128.139.26.6' Request 102: Found [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:12285 handle_response_invite: SIP response 100 to RE-invite on outgoing call F3620912-C55611DB-94AE86A5-87E354FD@128.139.26.6 pbx-gr*CLI> <--- SIP read from 128.139.26.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK493bb0d9;rport From: ;tag=as29e8d94e To: "YEHAVI BOURVIN" ;tag=26F00E08-1E38 Date: Tue, 27 Feb 2007 05:05:22 GMT Call-ID: F3620912-C55611DB-94AE86A5-87E354FD@128.139.26.6 Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER Allow-Events: telephone-event Remote-Party-ID: "YEHAVI BOURVIN" ;party=called;screen=no;privacy=off Contact: Supported: replaces Content-Type: application/sdp Content-Length: 218 v=0 o=CiscoSystemsSIP-GW-UserAgent 6058 5535 IN IP4 128.139.26.6 s=SIP Call c=IN IP4 128.139.26.6 t=0 0 m=audio 18082 RTP/AVP 8 c=IN IP4 128.139.26.6 a=rtpmap:8 PCMA/8000 a=ptime:20 a=silenceSupp:off - - - - <-------------> [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 0 [ 14]: SIP/2.0 200 OK [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 1 [ 63]: Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK493bb0d9;rport [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 2 [ 45]: From: ;tag=as29e8d94e [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 3 [ 63]: To: "YEHAVI BOURVIN" ;tag=26F00E08-1E38 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 4 [ 35]: Date: Tue, 27 Feb 2007 05:05:22 GMT [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 5 [ 57]: Call-ID: F3620912-C55611DB-94AE86A5-87E354FD@128.139.26.6 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 6 [ 33]: Server: Cisco-SIPGateway/IOS-12.x [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 7 [ 16]: CSeq: 102 INVITE [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 8 [ 97]: Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 9 [ 29]: Allow-Events: telephone-event [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 10 [ 93]: Remote-Party-ID: "YEHAVI BOURVIN" ;party=called;screen=no;privacy=off [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 11 [ 38]: Contact: [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 12 [ 19]: Supported: replaces [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 13 [ 29]: Content-Type: application/sdp [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 14 [ 19]: Content-Length: 218 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 15 [ 0]: [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 0 [ 3]: v=0 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 1 [ 60]: o=CiscoSystemsSIP-GW-UserAgent 6058 5535 IN IP4 128.139.26.6 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 2 [ 10]: s=SIP Call [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 3 [ 21]: c=IN IP4 128.139.26.6 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 4 [ 5]: t=0 0 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 5 [ 23]: m=audio 18082 RTP/AVP 8 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 6 [ 21]: c=IN IP4 128.139.26.6 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 8 [ 10]: a=ptime:20 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 9 [ 25]: a=silenceSupp:off - - - - --- (15 headers 10 lines) --- [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:2218 __sip_ack: Acked pending invite 102 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:2234 __sip_ack: Stopping retransmission on 'F3620912-C55611DB-94AE86A5-87E354FD@128.139.26.6' of Request 102: Match Found [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:12285 handle_response_invite: SIP response 200 to RE-invite on outgoing call F3620912-C55611DB-94AE86A5-87E354FD@128.139.26.6 Found RTP audio format 8 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:5282 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 128.139.26.6:18082 Found description format PCMA for ID 8 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:5524 process_sdp: T38 state changed to 0 on channel SIP/gr-pbx-link-09607500 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 128.139.26.6:18082 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:5615 process_sdp: We're settling with these formats: 0x8 (alaw) [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:5622 process_sdp: We have an owner, now see if we need to change this call [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:3305 update_call_counter: Updating call counter for incoming call --- set_address_from_contact host '128.139.26.6' [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:8478 build_route: build_route: Contact hop: list_route: hop: [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:6050 reqprep: Strict routing enforced for session F3620912-C55611DB-94AE86A5-87E354FD@128.139.26.6 set_destination: Parsing for address/port to send to set_destination: set destination to 128.139.26.6, port 5060 Transmitting (no NAT) to 128.139.26.6:5060: ACK sip:89444@128.139.26.6:5060 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK1621f23b;rport Max-Forwards: 70 From: ;tag=as29e8d94e To: "YEHAVI BOURVIN" ;tag=26F00E08-1E38 Contact: Call-ID: F3620912-C55611DB-94AE86A5-87E354FD@128.139.26.6 CSeq: 102 ACK User-Agent: Asterisk PBX SVN-trunk-r56126M Content-Length: 0 --- pbx-gr*CLI> <--- SIP read from 132.64.4.141:2051 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK6a1c94ad;rport=5060 From: "YEHAVI BOURVIN" ;tag=as4e147432 To: "80602" ;tag=1fnx3usnpb Call-ID: 0203bcd16026f51d4c0e9821089c0e96@132.64.9.163 CSeq: 103 INVITE Contact: ;flow-id=1 User-Agent: snom320/6.2.2 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Content-Type: application/sdp Content-Length: 327 v=0 o=root 1787986285 1787986287 IN IP4 132.64.4.141 s=call c=IN IP4 132.64.4.141 t=0 0 m=audio 49314 RTP/AVP 8 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:FKUkTJdhQXu+fhL/5a7Z3c6mpSO3tuKR92f2fag3 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=encryption:optional a=sendrecv <-------------> [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 0 [ 14]: SIP/2.0 200 Ok [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 1 [ 68]: Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK6a1c94ad;rport=5060 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 2 [ 62]: From: "YEHAVI BOURVIN" ;tag=as4e147432 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 3 [ 70]: To: "80602" ;tag=1fnx3usnpb [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 4 [ 54]: Call-ID: 0203bcd16026f51d4c0e9821089c0e96@132.64.9.163 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 5 [ 16]: CSeq: 103 INVITE [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 6 [ 62]: Contact: ;flow-id=1 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 7 [ 25]: User-Agent: snom320/6.2.2 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 8 [ 88]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 9 [ 31]: Allow-Events: talk, hold, refer [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 10 [ 44]: Supported: timer, 100rel, replaces, callerid [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 11 [ 29]: Content-Type: application/sdp [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 12 [ 19]: Content-Length: 327 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 13 [ 0]: [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 0 [ 3]: v=0 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 1 [ 48]: o=root 1787986285 1787986287 IN IP4 132.64.4.141 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 2 [ 6]: s=call [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 3 [ 21]: c=IN IP4 132.64.4.141 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 4 [ 5]: t=0 0 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 5 [ 27]: m=audio 49314 RTP/AVP 8 101 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 6 [ 82]: a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:FKUkTJdhQXu+fhL/5a7Z3c6mpSO3tuKR92f2fag3 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 7 [ 20]: a=rtpmap:8 pcma/8000 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 9 [ 15]: a=fmtp:101 0-16 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 10 [ 10]: a=ptime:20 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 11 [ 21]: a=encryption:optional [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 12 [ 10]: a=sendrecv --- (13 headers 13 lines) --- [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:2218 __sip_ack: Acked pending invite 103 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:2223 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #8903 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:2234 __sip_ack: Stopping retransmission on '0203bcd16026f51d4c0e9821089c0e96@132.64.9.163' of Request 103: Match Found [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:12285 handle_response_invite: SIP response 200 to RE-invite on outgoing call 0203bcd16026f51d4c0e9821089c0e96@132.64.9.163 Found RTP audio format 8 Found RTP audio format 101 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:5282 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 132.64.4.141:49314 Got unsupported a:crypto in SDP offer Found description format pcma for ID 8 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:5524 process_sdp: T38 state changed to 0 on channel SIP/80602-0960b470 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=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 132.64.4.141:49314 [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:5615 process_sdp: We're settling with these formats: 0x8 (alaw) [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:5622 process_sdp: We have an owner, now see if we need to change this call [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:3305 update_call_counter: Updating call counter for outgoing call --- set_address_from_contact host '132.64.4.141' [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:8417 build_route: build_route: Retaining previous route: [Feb 27 07:01:20] DEBUG[19189]: chan_sip.c:6050 reqprep: Strict routing enforced for session 0203bcd16026f51d4c0e9821089c0e96@132.64.9.163 set_destination: Parsing for address/port to send to set_destination: set destination to 132.64.4.141, port 2051 Transmitting (no NAT) to 132.64.4.141:2051: ACK sip:80602@132.64.4.141:2051;line=kbq2n6ok SIP/2.0 Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK2bfc3c08;rport Max-Forwards: 70 From: "YEHAVI BOURVIN" ;tag=as4e147432 To: ;tag=1fnx3usnpb Contact: Call-ID: 0203bcd16026f51d4c0e9821089c0e96@132.64.9.163 CSeq: 103 ACK User-Agent: Asterisk PBX SVN-trunk-r56126M Content-Length: 0 --- pbx-gr*CLI> pbx-gr*CLI> ~~~~~~~~~~~~~~~~~~~~~~~~~~ Here comes the "hold key" press pbx-gr*CLI> pbx-gr*CLI> pbx-gr*CLI> pbx-gr*CLI> pbx-gr*CLI> [Feb 27 07:01:41] DEBUG[19189]: db.c:238 ast_db_del: Unable to find key '80682' in family 'SIP/Registry' [Feb 27 07:01:41] DEBUG[19189]: devicestate.c:444 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/80682 [Feb 27 07:01:41] DEBUG[19189]: chan_sip.c:2586 sip_destroy_peer: Destroying SIP peer 80682 pbx-gr*CLI> [Feb 27 07:01:41] DEBUG[19189]: devicestate.c:302 ast_device_state: No provider found, checking channel drivers for SIP - 80682 pbx-gr*CLI> [Feb 27 07:01:41] DEBUG[19189]: chan_sip.c:15897 sip_devicestate: Checking device state for peer 80682 pbx-gr*CLI> [Feb 27 07:01:41] DEBUG[19189]: res_config_mysql.c:651 mysql_reconnect: MySQL RealTime: Everything is fine. pbx-gr*CLI> [Feb 27 07:01:41] DEBUG[19189]: res_config_mysql.c:139 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_users WHERE name = '80682' pbx-gr*CLI> [Feb 27 07:01:41] DEBUG[19189]: db.c:198 ast_db_get: Unable to find key '80682' in family 'SIP/Registry' pbx-gr*CLI> [Feb 27 07:01:41] DEBUG[19189]: chan_sip.c:2714 realtime_peer: -REALTIME- loading peer from database to memory. Name: 80682. Peer objects: 0 pbx-gr*CLI> [Feb 27 07:01:41] DEBUG[19189]: devicestate.c:428 do_state_change: Changing state for SIP/80682 - state 5 (Unavailable) pbx-gr*CLI> [Feb 27 07:01:41] DEBUG[19189]: app_queue.c:568 changethread: Device 'SIP/80682' changed to state '5' (Unavailable) but we don't care because they're not a member of any queue. pbx-gr*CLI> [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4663 sip_alloc: Allocating new SIP dialog for (No Call-ID) - NOTIFY (No RTP) [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:3401 sip_destroy: Destroying SIP dialog 126b2416771ca12815c832dd4ecf2fc0@127.0.0.1 Really destroying SIP dialog '126b2416771ca12815c832dd4ecf2fc0@127.0.0.1' Method: NOTIFY pbx-gr*CLI> <--- SIP read from 132.64.4.141:2051 ---> INVITE sip:89444@132.64.9.163 SIP/2.0 Via: SIP/2.0/UDP 132.64.4.141:2051;branch=z9hG4bK-6bdd8to026ov;rport From: "80602" ;tag=1fnx3usnpb To: "YEHAVI BOURVIN" ;tag=as4e147432 Call-ID: 0203bcd16026f51d4c0e9821089c0e96@132.64.9.163 CSeq: 1 INVITE Max-Forwards: 70 Contact: ;flow-id=1 P-Key-Flags: keys="3" User-Agent: snom320/6.2.2 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Session-Expires: 3600;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: 475 v=0 o=root 1787986285 1787986288 IN IP4 132.64.4.141 s=call c=IN IP4 132.64.4.141 t=0 0 m=audio 49314 RTP/AVP 0 8 9 2 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:FKUkTJdhQXu+fhL/5a7Z3c6mpSO3tuKR92f2fag3 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=encryption:optional a=sendonly <-------------> [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 0 [ 37]: INVITE sip:89444@132.64.9.163 SIP/2.0 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 1 [ 68]: Via: SIP/2.0/UDP 132.64.4.141:2051;branch=z9hG4bK-6bdd8to026ov;rport [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 2 [ 72]: From: "80602" ;tag=1fnx3usnpb [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 3 [ 60]: To: "YEHAVI BOURVIN" ;tag=as4e147432 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 4 [ 54]: Call-ID: 0203bcd16026f51d4c0e9821089c0e96@132.64.9.163 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 5 [ 14]: CSeq: 1 INVITE [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 6 [ 16]: Max-Forwards: 70 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 7 [ 62]: Contact: ;flow-id=1 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 8 [ 21]: P-Key-Flags: keys="3" [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 9 [ 25]: User-Agent: snom320/6.2.2 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 10 [ 23]: Accept: application/sdp [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 11 [ 88]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 12 [ 31]: Allow-Events: talk, hold, refer [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 13 [ 44]: Supported: timer, 100rel, replaces, callerid [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 14 [ 35]: Session-Expires: 3600;refresher=uas [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 15 [ 10]: Min-SE: 90 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 16 [ 29]: Content-Type: application/sdp [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 17 [ 19]: Content-Length: 475 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 18 [ 0]: [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 0 [ 3]: v=0 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 1 [ 48]: o=root 1787986285 1787986288 IN IP4 132.64.4.141 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 2 [ 6]: s=call [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 3 [ 21]: c=IN IP4 132.64.4.141 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 4 [ 5]: t=0 0 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 5 [ 40]: m=audio 49314 RTP/AVP 0 8 9 2 3 18 4 101 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 6 [ 82]: a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:FKUkTJdhQXu+fhL/5a7Z3c6mpSO3tuKR92f2fag3 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 7 [ 20]: a=rtpmap:0 pcmu/8000 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 8 [ 20]: a=rtpmap:8 pcma/8000 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 9 [ 20]: a=rtpmap:9 g722/8000 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 10 [ 23]: a=rtpmap:2 g726-32/8000 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 11 [ 19]: a=rtpmap:3 gsm/8000 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 12 [ 21]: a=rtpmap:18 g729/8000 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 13 [ 20]: a=rtpmap:4 g723/8000 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 14 [ 33]: a=rtpmap:101 telephone-event/8000 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 15 [ 15]: a=fmtp:101 0-16 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 16 [ 10]: a=ptime:20 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 17 [ 21]: a=encryption:optional [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 18 [ 10]: a=sendonly --- (18 headers 19 lines) --- [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:15271 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:1829 parse_sip_options: Begin: parsing SIP "Supported: timer, 100rel, replaces, callerid" [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:1837 parse_sip_options: Found SIP option: -timer- [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:1843 parse_sip_options: Matched SIP option: timer [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:1837 parse_sip_options: Found SIP option: -100rel- [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:1843 parse_sip_options: Matched SIP option: 100rel [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:1837 parse_sip_options: Found SIP option: -replaces- [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:1843 parse_sip_options: Matched SIP option: replaces [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:1837 parse_sip_options: Found SIP option: -callerid- [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:1851 parse_sip_options: Found no match for SIP option: callerid (Please file bug report!) Sending to 132.64.4.141 : 2051 (NAT) [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:13975 handle_request_invite: Initializing initreq for method INVITE - callid 0203bcd16026f51d4c0e9821089c0e96@132.64.9.163 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 9 Found RTP audio format 2 Found RTP audio format 3 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 101 pbx-gr*CLI> [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:5282 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 132.64.4.141:49314 Got unsupported a:crypto in SDP offer Found description format pcmu for ID 0 Found description format pcma for ID 8 Found description format g722 for ID 9 Found description format g726-32 for ID 2 Found description format gsm for ID 3 Found description format g729 for ID 18 Found description format g723 for ID 4 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:5524 process_sdp: T38 state changed to 0 on channel SIP/80602-0960b470 Capabilities: us - 0x8 (alaw), peer - audio=0x190f (g723|gsm|ulaw|alaw|g726|g729|g722)/video=0x0 (nothing)/text=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 132.64.4.141:49314 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:5615 process_sdp: We're settling with these formats: 0x8 (alaw) [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:5622 process_sdp: We have an owner, now see if we need to change this call [Feb 27 07:01:42] DEBUG[19189]: devicestate.c:444 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/80602 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:14109 handle_request_invite: Got a SIP re-invite for call 0203bcd16026f51d4c0e9821089c0e96@132.64.9.163 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:14204 handle_request_invite: SIP/80602-0960b470: This call is UP.... [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:6952 transmit_response_with_sdp: Setting framing from config on incoming call [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:6674 add_sdp: ** Our capability: 0x8 (alaw) Video flag: True Text flag: True [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:6676 add_sdp: ** Our prefcodec: 0x8 (alaw) Audio is at 132.64.9.163 port 18986 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:6837 add_sdp: -- Done with adding codecs to SDP [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:6897 add_sdp: Done building SDP. Settling with this capability: 0x8 (alaw) <--- Reliably Transmitting (NAT) to 132.64.4.141:2051 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 132.64.4.141:2051;branch=z9hG4bK-6bdd8to026ov;received=132.64.4.141;rport=2051 From: "80602" ;tag=1fnx3usnpb To: "YEHAVI BOURVIN" ;tag=as4e147432 Call-ID: 0203bcd16026f51d4c0e9821089c0e96@132.64.9.163 CSeq: 1 INVITE User-Agent: Asterisk PBX SVN-trunk-r56126M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 240 v=0 o=root 19178 19180 IN IP4 128.139.26.6 s=session c=IN IP4 128.139.26.6 t=0 0 m=audio 18082 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=recvonly <------------> [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:2115 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #8906 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:17627 sip_set_rtp_peer: Sending reinvite on SIP 'F3620912-C55611DB-94AE86A5-87E354FD@128.139.26.6' - It's audio soon redirected to IP 132.64.9.163 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:6050 reqprep: Strict routing enforced for session F3620912-C55611DB-94AE86A5-87E354FD@128.139.26.6 set_destination: Parsing for address/port to send to set_destination: set destination to 128.139.26.6, port 5060 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:6674 add_sdp: ** Our capability: 0x8 (alaw) Video flag: True Text flag: True [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:6676 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 132.64.9.163 port 61938 Adding codec 0x8 (alaw) to SDP [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:6837 add_sdp: -- Done with adding codecs to SDP [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:6897 add_sdp: Done building SDP. Settling with this capability: 0x8 (alaw) [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:1716 initialize_initreq: Initializing already initialized SIP dialog F3620912-C55611DB-94AE86A5-87E354FD@128.139.26.6 (presumably reinvite) [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 0 [ 42]: INVITE sip:89444@128.139.26.6:5060 SIP/2.0 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 1 [ 63]: Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK3d5a2b1f;rport [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 2 [ 16]: Max-Forwards: 70 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 3 [ 45]: From: ;tag=as29e8d94e [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 4 [ 63]: To: "YEHAVI BOURVIN" ;tag=26F00E08-1E38 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 5 [ 33]: Contact: [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 6 [ 57]: Call-ID: F3620912-C55611DB-94AE86A5-87E354FD@128.139.26.6 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 7 [ 16]: CSeq: 103 INVITE [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 8 [ 42]: User-Agent: Asterisk PBX SVN-trunk-r56126M [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 9 [ 66]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 10 [ 19]: Supported: replaces [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 12 [ 29]: Content-Type: application/sdp [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 13 [ 19]: Content-Length: 184 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 14 [ 0]: [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 0 [ 3]: v=0 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 1 [ 38]: o=root 19178 19180 IN IP4 132.64.9.163 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 2 [ 9]: s=session [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 3 [ 21]: c=IN IP4 132.64.9.163 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 4 [ 5]: t=0 0 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 5 [ 23]: m=audio 61938 RTP/AVP 8 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 7 [ 25]: a=silenceSupp:off - - - - [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 8 [ 10]: a=ptime:20 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 9 [ 10]: a=sendrecv Reliably Transmitting (no NAT) to 128.139.26.6:5060: INVITE sip:89444@128.139.26.6:5060 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK3d5a2b1f;rport Max-Forwards: 70 From: ;tag=as29e8d94e To: "YEHAVI BOURVIN" ;tag=26F00E08-1E38 Contact: Call-ID: F3620912-C55611DB-94AE86A5-87E354FD@128.139.26.6 CSeq: 103 INVITE User-Agent: Asterisk PBX SVN-trunk-r56126M 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: 184 v=0 o=root 19178 19180 IN IP4 132.64.9.163 s=session c=IN IP4 132.64.9.163 t=0 0 m=audio 61938 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:2115 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #8907 -- Started music on hold, class 'default', on SIP/gr-pbx-link-09607500 [Feb 27 07:01:42] DEBUG[19189]: rtp.c:2937 bridge_native_loop: Oooh, 'SIP/80602-0960b470' changed end address to 0.0.0.0:0 (format 6415) [Feb 27 07:01:42] DEBUG[19189]: rtp.c:2939 bridge_native_loop: Oooh, 'SIP/80602-0960b470' changed end vaddress to 0.0.0.0:0 (format 6415) [Feb 27 07:01:42] DEBUG[19189]: rtp.c:2941 bridge_native_loop: Oooh, 'SIP/80602-0960b470' changed end taddress to 0.0.0.0:0 (format 6415) [Feb 27 07:01:42] DEBUG[19189]: rtp.c:2943 bridge_native_loop: Oooh, 'SIP/80602-0960b470' was 132.64.4.141:49314/(format 8) [Feb 27 07:01:42] DEBUG[19189]: rtp.c:2945 bridge_native_loop: Oooh, 'SIP/80602-0960b470' was 0.0.0.0:0/(format 8) [Feb 27 07:01:42] DEBUG[19189]: rtp.c:2947 bridge_native_loop: Oooh, 'SIP/80602-0960b470' was 0.0.0.0:0/(format 8) [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:17632 sip_set_rtp_peer: Deferring reinvite on SIP 'F3620912-C55611DB-94AE86A5-87E354FD@128.139.26.6' - It's audio will be redirected to IP 132.64.9.163 pbx-gr*CLI> [Feb 27 07:01:42] DEBUG[19189]: devicestate.c:302 ast_device_state: No provider found, checking channel drivers for SIP - 80602 pbx-gr*CLI> [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:15897 sip_devicestate: Checking device state for peer 80602 pbx-gr*CLI> [Feb 27 07:01:42] DEBUG[19189]: devicestate.c:428 do_state_change: Changing state for SIP/80602 - state 8 (On Hold) pbx-gr*CLI> [Feb 27 07:01:42] DEBUG[19189]: devicestate.c:302 ast_device_state: No provider found, checking channel drivers for SIP - 80602 pbx-gr*CLI> [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:15897 sip_devicestate: Checking device state for peer 80602 pbx-gr*CLI> [Feb 27 07:01:42] DEBUG[19189]: devicestate.c:302 ast_device_state: No provider found, checking channel drivers for SIP - 80602 pbx-gr*CLI> [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:15897 sip_devicestate: Checking device state for peer 80602 pbx-gr*CLI> Reliably Transmitting (no NAT) to 132.64.4.144:2051: NOTIFY sip:80601@pbx-gr.cc.huji.ac.il SIP/2.0 Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK6e1b25b9;rport Max-Forwards: 70 From: ;tag=as19934fd9 To: ;tag=1ipswbr4bt Contact: Call-ID: 3c26700d7c83-m8j3bh85uu3m@snom360-0004132384F8 CSeq: 103 NOTIFY User-Agent: Asterisk PBX SVN-trunk-r56126M Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 330 confirmed --- pbx-gr*CLI> [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:2115 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #8908 pbx-gr*CLI> Extension Changed 80602 new state Hold for Notify User 80601 pbx-gr*CLI> [Feb 27 07:01:42] DEBUG[19189]: devicestate.c:302 ast_device_state: No provider found, checking channel drivers for SIP - 80602 pbx-gr*CLI> [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:15897 sip_devicestate: Checking device state for peer 80602 pbx-gr*CLI> Reliably Transmitting (no NAT) to 132.64.4.121:5060: NOTIFY sip:80607@132.64.4.121 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK31e05768;rport Max-Forwards: 70 From: ;tag=as760e1762 To: "80607" ;tag=D7A78ABD-FB1A96BC Contact: Call-ID: 64aeec39-22798afb-f4d0577a@132.64.4.121 CSeq: 137 NOTIFY User-Agent: Asterisk PBX SVN-trunk-r56126M Event: presence Content-Type: application/xpidf+xml Subscription-State: active pbx-gr*CLI> Content-Length: 352
--- pbx-gr*CLI> [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:2115 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #8909 pbx-gr*CLI> Extension Changed 80602 new state Hold for Notify User 80607 pbx-gr*CLI> [Feb 27 07:01:42] DEBUG[19189]: devicestate.c:302 ast_device_state: No provider found, checking channel drivers for SIP - 80602 pbx-gr*CLI> [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:15897 sip_devicestate: Checking device state for peer 80602 pbx-gr*CLI> Reliably Transmitting (no NAT) to 132.64.4.135:5060: NOTIFY sip:80615@132.64.4.135 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK76a4b924;rport Max-Forwards: 70 From: ;tag=as24fb17f2 To: "80615" ;tag=16D7DF64-1DDA1477 Contact: Call-ID: 3e849123-fbaf994a-7e2c465@132.64.4.135 CSeq: 137 NOTIFY User-Agent: Asterisk PBX SVN-trunk-r56126M Event: presence Content-Type: application/xpidf+xml Subscription-State: active Cpbx-gr*CLI> ontent-Length: 352
--- pbx-gr*CLI> [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:2115 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #8910 pbx-gr*CLI> Extension Changed 80602 new state Hold for Notify User 80615 pbx-gr*CLI> [Feb 27 07:01:42] DEBUG[19189]: app_queue.c:568 changethread: Device 'SIP/80602' changed to state '8' (On Hold) but we don't care because they're not a member of any queue. pbx-gr*CLI> <--- SIP read from 128.139.26.6:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK3d5a2b1f;rport From: ;tag=as29e8d94e To: "YEHAVI BOURVIN" ;tag=26F00E08-1E38 Date: Tue, 27 Feb 2007 05:05:44 GMT Call-ID: F3620912-C55611DB-94AE86A5-87E354FD@128.139.26.6 Server: Cisco-SIPGateway/IOS-12.x CSeq: 103 INVITE Allow-Events: telephone-event Remote-Party-ID: "YEHAVI BOURVIN" ;party=called;screen=no;privacy=off Content-Length: 0 <-------------> [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 0 [ 18]: SIP/2.0 100 Trying [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 1 [ 63]: Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK3d5a2b1f;rport [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 2 [ 45]: From: ;tag=as29e8d94e [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 3 [ 63]: To: "YEHAVI BOURVIN" ;tag=26F00E08-1E38 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 4 [ 35]: Date: Tue, 27 Feb 2007 05:05:44 GMT [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 5 [ 57]: Call-ID: F3620912-C55611DB-94AE86A5-87E354FD@128.139.26.6 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 6 [ 33]: Server: Cisco-SIPGateway/IOS-12.x [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 7 [ 16]: CSeq: 103 INVITE [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 8 [ 29]: Allow-Events: telephone-event [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 9 [ 93]: Remote-Party-ID: "YEHAVI BOURVIN" ;party=called;screen=no;privacy=off [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 10 [ 17]: Content-Length: 0 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 11 [ 0]: --- (11 headers 0 lines) --- [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:2268 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #8907 - INVITE (got response) [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:2277 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on 'F3620912-C55611DB-94AE86A5-87E354FD@128.139.26.6' Request 103: Found [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:12285 handle_response_invite: SIP response 100 to RE-invite on outgoing call F3620912-C55611DB-94AE86A5-87E354FD@128.139.26.6 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:12268 check_pendings: Sending pending reinvite on 'F3620912-C55611DB-94AE86A5-87E354FD@128.139.26.6' [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:6050 reqprep: Strict routing enforced for session F3620912-C55611DB-94AE86A5-87E354FD@128.139.26.6 set_destination: Parsing for address/port to send to set_destination: set destination to 128.139.26.6, port 5060 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:6674 add_sdp: ** Our capability: 0x8 (alaw) Video flag: True Text flag: True [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:6676 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 132.64.9.163 port 61938 Adding codec 0x8 (alaw) to SDP [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:6837 add_sdp: -- Done with adding codecs to SDP [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:6897 add_sdp: Done building SDP. Settling with this capability: 0x8 (alaw) [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:1716 initialize_initreq: Initializing already initialized SIP dialog F3620912-C55611DB-94AE86A5-87E354FD@128.139.26.6 (presumably reinvite) [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 0 [ 42]: INVITE sip:89444@128.139.26.6:5060 SIP/2.0 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 1 [ 63]: Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK2bfcf487;rport [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 2 [ 16]: Max-Forwards: 70 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 3 [ 45]: From: ;tag=as29e8d94e [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 4 [ 63]: To: "YEHAVI BOURVIN" ;tag=26F00E08-1E38 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 5 [ 33]: Contact: [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 6 [ 57]: Call-ID: F3620912-C55611DB-94AE86A5-87E354FD@128.139.26.6 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 7 [ 16]: CSeq: 104 INVITE [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 8 [ 42]: User-Agent: Asterisk PBX SVN-trunk-r56126M [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 9 [ 66]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 10 [ 19]: Supported: replaces [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 12 [ 29]: Content-Type: application/sdp [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 13 [ 19]: Content-Length: 184 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 14 [ 0]: [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 0 [ 3]: v=0 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 1 [ 38]: o=root 19178 19181 IN IP4 132.64.9.163 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 2 [ 9]: s=session [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 3 [ 21]: c=IN IP4 132.64.9.163 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 4 [ 5]: t=0 0 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 5 [ 23]: m=audio 61938 RTP/AVP 8 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 7 [ 25]: a=silenceSupp:off - - - - [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 8 [ 10]: a=ptime:20 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 9 [ 10]: a=sendrecv Reliably Transmitting (no NAT) to 128.139.26.6:5060: INVITE sip:89444@128.139.26.6:5060 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK2bfcf487;rport Max-Forwards: 70 From: ;tag=as29e8d94e To: "YEHAVI BOURVIN" ;tag=26F00E08-1E38 Contact: Call-ID: F3620912-C55611DB-94AE86A5-87E354FD@128.139.26.6 CSeq: 104 INVITE User-Agent: Asterisk PBX SVN-trunk-r56126M 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: 184 v=0 o=root 19178 19181 IN IP4 132.64.9.163 s=session c=IN IP4 132.64.9.163 t=0 0 m=audio 61938 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:2115 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #8911 pbx-gr*CLI> <--- SIP read from 128.139.26.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK3d5a2b1f;rport From: ;tag=as29e8d94e To: "YEHAVI BOURVIN" ;tag=26F00E08-1E38 Date: Tue, 27 Feb 2007 05:05:44 GMT Call-ID: F3620912-C55611DB-94AE86A5-87E354FD@128.139.26.6 Server: Cisco-SIPGateway/IOS-12.x CSeq: 103 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER Allow-Events: telephone-event Remote-Party-ID: "YEHAVI BOURVIN" ;party=called;screen=no;privacy=off Contact: Supported: replaces Content-Type: application/sdp Content-Length: 218 v=0 o=CiscoSystemsSIP-GW-UserAgent 6058 5536 IN IP4 128.139.26.6 s=SIP Call c=IN IP4 128.139.26.6 t=0 0 m=audio 18082 RTP/AVP 8 c=IN IP4 128.139.26.6 a=rtpmap:8 PCMA/8000 a=ptime:20 a=silenceSupp:off - - - - <-------------> [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 0 [ 14]: SIP/2.0 200 OK [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 1 [ 63]: Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK3d5a2b1f;rport [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 2 [ 45]: From: ;tag=as29e8d94e [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 3 [ 63]: To: "YEHAVI BOURVIN" ;tag=26F00E08-1E38 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 4 [ 35]: Date: Tue, 27 Feb 2007 05:05:44 GMT [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 5 [ 57]: Call-ID: F3620912-C55611DB-94AE86A5-87E354FD@128.139.26.6 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 6 [ 33]: Server: Cisco-SIPGateway/IOS-12.x [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 7 [ 16]: CSeq: 103 INVITE [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 8 [ 97]: Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 9 [ 29]: Allow-Events: telephone-event [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 10 [ 93]: Remote-Party-ID: "YEHAVI BOURVIN" ;party=called;screen=no;privacy=off [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 11 [ 38]: Contact: [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 12 [ 19]: Supported: replaces [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 13 [ 29]: Content-Type: application/sdp [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 14 [ 19]: Content-Length: 218 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 15 [ 0]: pbx-gr*CLI> [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 0 [ 3]: v=0 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 1 [ 60]: o=CiscoSystemsSIP-GW-UserAgent 6058 5536 IN IP4 128.139.26.6 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 2 [ 10]: s=SIP Call [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 3 [ 21]: c=IN IP4 128.139.26.6 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 4 [ 5]: t=0 0 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 5 [ 23]: m=audio 18082 RTP/AVP 8 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 6 [ 21]: c=IN IP4 128.139.26.6 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 8 [ 10]: a=ptime:20 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 9 [ 25]: a=silenceSupp:off - - - - --- (15 headers 10 lines) --- [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:2234 __sip_ack: Stopping retransmission on 'F3620912-C55611DB-94AE86A5-87E354FD@128.139.26.6' of Request 103: Match Found [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:12285 handle_response_invite: SIP response 200 to RE-invite on outgoing call F3620912-C55611DB-94AE86A5-87E354FD@128.139.26.6 Found RTP audio format 8 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:5282 process_sdp: Peer doesn't provide T.38 UDPTL pbx-gr*CLI> Peer audio RTP is at port 128.139.26.6:18082 Found description format PCMA for ID 8 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:5524 process_sdp: T38 state changed to 0 on channel SIP/gr-pbx-link-09607500 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) pbx-gr*CLI> Peer audio RTP is at port 128.139.26.6:18082 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:5615 process_sdp: We're settling with these formats: 0x8 (alaw) [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:5622 process_sdp: We have an owner, now see if we need to change this call [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:3305 update_call_counter: Updating call counter for incoming call --- set_address_from_contact host '128.139.26.6' [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:8417 build_route: build_route: Retaining previous route: [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:6050 reqprep: Strict routing enforced for session F3620912-C55611DB-94AE86A5-87E354FD@128.139.26.6 set_destination: Parsing for address/port to send to set_destination: set destination to 128.139.26.6, port 5060 Transmitting (no NAT) to 128.139.26.6:5060: ACK sip:89444@128.139.26.6:5060 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK73525bb3;rport Max-Forwards: 70 From: ;tag=as29e8d94e To: "YEHAVI BOURVIN" ;tag=26F00E08-1E38 pbx-gr*CLI> Contact: Call-ID: F3620912-C55611DB-94AE86A5-87E354FD@128.139.26.6 CSeq: 103 ACK User-Agent: Asterisk PBX SVN-trunk-r56126M Content-Length: 0 --- <--- SIP read from 132.64.4.135:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK76a4b924;rport From: ;tag=as24fb17f2 To: "80615" ;tag=16D7DF64-1DDA1477 CSeq: 137 NOTIFY Call-ID: 3e849123-fbaf994a-7e2c465@132.64.4.135 Contact: Event: presence User-Agent: PolycomSoundPointIP-SPIP_430-UA/2.0.3.0127 Content-Length: 0 <-------------> [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 0 [ 14]: SIP/2.0 200 OK [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 1 [ 63]: Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK76a4b924;rport [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 2 [ 45]: From: ;tag=as24fb17f2 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 3 [ 58]: To: "80615" ;tag=16D7DF64-1DDA1477 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 4 [ 16]: CSeq: 137 NOTIFY [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 5 [ 47]: Call-ID: 3e849123-fbaf994a-7e2c465@132.64.4.135 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 6 [ 33]: Contact: [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 7 [ 15]: Event: presence [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 8 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_430-UA/2.0.3.0127 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 9 [ 17]: Content-Length: 0 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 10 [ 0]: --- (10 headers 0 lines) --- [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:2223 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #8910 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:2234 __sip_ack: Stopping retransmission on '3e849123-fbaf994a-7e2c465@132.64.4.135' of Request 137: Match Found SIP Response message for INCOMING dialog NOTIFY arrived pbx-gr*CLI> <--- SIP read from 128.139.26.6:5060 ---> SIP/2.0 500 Internal Server Error Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK2bfcf487;rport From: ;tag=as29e8d94e To: "YEHAVI BOURVIN" ;tag=26F00E08-1E38 Call-ID: F3620912-C55611DB-94AE86A5-87E354FD@128.139.26.6 CSeq: 104 INVITE Reason: Q.850;cause=100 Content-Length: 0 <-------------> [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 0 [ 33]: SIP/2.0 500 Internal Server Error [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 1 [ 63]: Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK2bfcf487;rport [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 2 [ 45]: From: ;tag=as29e8d94e [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 3 [ 63]: To: "YEHAVI BOURVIN" ;tag=26F00E08-1E38 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 4 [ 57]: Call-ID: F3620912-C55611DB-94AE86A5-87E354FD@128.139.26.6 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 5 [ 16]: CSeq: 104 INVITE [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 6 [ 23]: Reason: Q.850;cause=100 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 7 [ 17]: Content-Length: 0 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 8 [ 0]: --- (8 headers 0 lines) --- [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:2218 __sip_ack: Acked pending invite 104 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:2223 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #8911 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:2234 __sip_ack: Stopping retransmission on 'F3620912-C55611DB-94AE86A5-87E354FD@128.139.26.6' of Request 104: Match Found -- Got SIP response 500 "Internal Server Error" back from 128.139.26.6 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:6050 reqprep: Strict routing enforced for session F3620912-C55611DB-94AE86A5-87E354FD@128.139.26.6 set_destination: Parsing for address/port to send to pbx-gr*CLI> set_destination: set destination to 128.139.26.6, port 5060 Transmitting (no NAT) to 128.139.26.6:5060: ACK sip:89444@128.139.26.6:5060 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK73525bb3;rport Max-Forwards: 70 From: ;tag=as29e8d94e To: "YEHAVI BOURVIN" ;tag=26F00E08-1E38 Contact: Call-ID: F3620912-C55611DB-94AE86A5-87E354FD@128.139.26.6 CSeq: 104 ACK User-Agent: Asterisk PBX SVN-trunk-r56126M Content-Length: 0 --- [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:1731 sip_alreadygone: Setting SIP_ALREADYGONE on dialog F3620912-C55611DB-94AE86A5-87E354FD@128.139.26.6 [Feb 27 07:01:42] DEBUG[19189]: rtp.c:3024 bridge_native_loop: Got a FRAME_CONTROL (8) frame on channel SIP/gr-pbx-link-09607500 [Feb 27 07:01:42] DEBUG[19189]: channel.c:3971 ast_channel_bridge: Returning from native bridge, channels: SIP/gr-pbx-link-09607500, SIP/80602-0960b470 [Feb 27 07:01:42] DEBUG[19189]: channel.c:1591 ast_hangup: Hanging up channel 'SIP/80602-0960b470' [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:3603 sip_hangup: Hangup call SIP/80602-0960b470, SIP callid 0203bcd16026f51d4c0e9821089c0e96@132.64.9.163) Scheduling destruction of SIP dialog '0203bcd16026f51d4c0e9821089c0e96@132.64.9.163' in 32000 ms (Method: INVITE) [Feb 27 07:01:42] DEBUG[19189]: devicestate.c:444 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/80602-0960b470 [Feb 27 07:01:42] DEBUG[19189]: rtp.c:1566 ast_rtp_early_bridge: Channel '' has no RTP, not doing anything [Feb 27 07:01:42] DEBUG[19189]: app_dial.c:1710 dial_exec_full: Exiting with DIALSTATUS=ANSWER. [Feb 27 07:01:42] DEBUG[19189]: pbx.c:2289 __ast_pbx_run: Spawn extension (huji-remote-gr,80602,6) exited non-zero on 'SIP/gr-pbx-link-09607500' == Spawn extension (huji-remote-gr, 80602, 6) exited non-zero on 'SIP/gr-pbx-link-09607500' [Feb 27 07:01:42] DEBUG[19189]: pbx.c:1693 pbx_extension_helper: Launching 'NoOp' -- Executing [h@huji-remote-gr:1] NoOp("SIP/gr-pbx-link-09607500", "89444 80602 h") in new stack [Feb 27 07:01:42] DEBUG[19189]: db.c:198 ast_db_get: Unable to find key 'CallBack' in family '89444' [Feb 27 07:01:42] DEBUG[19189]: func_db.c:73 function_db_read: DB: 89444/CallBack not found in database. [Feb 27 07:01:42] DEBUG[19189]: pbx.c:1543 pbx_substitute_variables_helper_full: Function result is '' [Feb 27 07:01:42] DEBUG[19189]: pbx.c:1693 pbx_extension_helper: Launching 'Set' -- Executing [h@huji-remote-gr:2] Set("SIP/gr-pbx-link-09607500", "tmp=") in new stack [Feb 27 07:01:42] DEBUG[19189]: pbx.c:1693 pbx_extension_helper: Launching 'NoOp' -- Executing [h@huji-remote-gr:3] NoOp("SIP/gr-pbx-link-09607500", "89444 ") in new stack [Feb 27 07:01:42] DEBUG[19189]: pbx.c:1693 pbx_extension_helper: Launching 'GotoIf' -- Executing [h@huji-remote-gr:4] GotoIf("SIP/gr-pbx-link-09607500", "?5:103") in new stack -- Goto (huji-remote-gr,h,103) [Feb 27 07:01:42] DEBUG[19189]: pbx.c:1693 pbx_extension_helper: Launching 'NoOp' -- Executing [h@huji-remote-gr:103] NoOp("SIP/gr-pbx-link-09607500", "Nothing to call") in new stack -- Stopped music on hold on SIP/gr-pbx-link-09607500 [Feb 27 07:01:42] DEBUG[19189]: pbx.c:1543 pbx_substitute_variables_helper_full: Function result is '"YEHAVI BOURVIN" <89444>' [Feb 27 07:01:42] DEBUG[19189]: pbx.c:1543 pbx_substitute_variables_helper_full: Function result is '89444' [Feb 27 07:01:42] DEBUG[19189]: pbx.c:1543 pbx_substitute_variables_helper_full: Function result is '80602' [Feb 27 07:01:42] DEBUG[19189]: pbx.c:1543 pbx_substitute_variables_helper_full: Function result is 'huji-remote-gr' [Feb 27 07:01:42] DEBUG[19189]: pbx.c:1543 pbx_substitute_variables_helper_full: Function result is 'SIP/gr-pbx-link-09607500' [Feb 27 07:01:42] DEBUG[19189]: pbx.c:1543 pbx_substitute_variables_helper_full: Function result is 'SIP/80602-0960b470' [Feb 27 07:01:42] DEBUG[19189]: pbx.c:1543 pbx_substitute_variables_helper_full: Function result is 'Dial' [Feb 27 07:01:42] DEBUG[19189]: pbx.c:1543 pbx_substitute_variables_helper_full: Function result is 'SIP/80602|20|' [Feb 27 07:01:42] DEBUG[19189]: pbx.c:1543 pbx_substitute_variables_helper_full: Function result is '2007-02-27 07:01:15' [Feb 27 07:01:42] DEBUG[19189]: pbx.c:1543 pbx_substitute_variables_helper_full: Function result is '2007-02-27 07:01:20' [Feb 27 07:01:42] DEBUG[19189]: pbx.c:1543 pbx_substitute_variables_helper_full: Function result is '2007-02-27 07:01:42' [Feb 27 07:01:42] DEBUG[19189]: pbx.c:1543 pbx_substitute_variables_helper_full: Function result is '27' [Feb 27 07:01:42] DEBUG[19189]: pbx.c:1543 pbx_substitute_variables_helper_full: Function result is '22' [Feb 27 07:01:42] DEBUG[19189]: pbx.c:1543 pbx_substitute_variables_helper_full: Function result is 'ANSWERED' [Feb 27 07:01:42] DEBUG[19189]: pbx.c:1543 pbx_substitute_variables_helper_full: Function result is 'DOCUMENTATION' [Feb 27 07:01:42] DEBUG[19189]: pbx.c:1543 pbx_substitute_variables_helper_full: Function result is '' [Feb 27 07:01:42] DEBUG[19189]: pbx.c:1543 pbx_substitute_variables_helper_full: Function result is '1172552475.159' [Feb 27 07:01:42] DEBUG[19189]: pbx.c:1543 pbx_substitute_variables_helper_full: Function result is '' [Feb 27 07:01:42] DEBUG[19189]: channel.c:1591 ast_hangup: Hanging up channel 'SIP/gr-pbx-link-09607500' [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:3603 sip_hangup: Hangup call SIP/gr-pbx-link-09607500, SIP callid F3620912-C55611DB-94AE86A5-87E354FD@128.139.26.6) [Feb 27 07:01:42] DEBUG[19189]: devicestate.c:444 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/gr-pbx-link-09607500 pbx-gr*CLI> [Feb 27 07:01:42] DEBUG[19189]: devicestate.c:302 ast_device_state: No provider found, checking channel drivers for SIP - 80602 pbx-gr*CLI> [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:15897 sip_devicestate: Checking device state for peer 80602 pbx-gr*CLI> [Feb 27 07:01:42] DEBUG[19189]: devicestate.c:428 do_state_change: Changing state for SIP/80602 - state 8 (On Hold) pbx-gr*CLI> [Feb 27 07:01:42] DEBUG[19189]: devicestate.c:302 ast_device_state: No provider found, checking channel drivers for SIP - 80602 pbx-gr*CLI> [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:15897 sip_devicestate: Checking device state for peer 80602 pbx-gr*CLI> [Feb 27 07:01:42] DEBUG[19189]: devicestate.c:302 ast_device_state: No provider found, checking channel drivers for SIP - gr-pbx-link pbx-gr*CLI> [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:15897 sip_devicestate: Checking device state for peer gr-pbx-link pbx-gr*CLI> [Feb 27 07:01:42] DEBUG[19189]: res_config_mysql.c:651 mysql_reconnect: MySQL RealTime: Everything is fine. pbx-gr*CLI> [Feb 27 07:01:42] DEBUG[19189]: res_config_mysql.c:139 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_users WHERE name = 'gr-pbx-link' pbx-gr*CLI> [Feb 27 07:01:42] DEBUG[19189]: app_queue.c:568 changethread: Device 'SIP/80602' changed to state '8' (On Hold) but we don't care because they're not a member of any queue. pbx-gr*CLI> [Feb 27 07:01:42] DEBUG[19189]: devicestate.c:428 do_state_change: Changing state for SIP/gr-pbx-link - state 4 (Invalid) pbx-gr*CLI> [Feb 27 07:01:42] DEBUG[19189]: app_queue.c:568 changethread: Device 'SIP/gr-pbx-link' changed to state '4' (Invalid) but we don't care because they're not a member of any queue. pbx-gr*CLI> <--- SIP read from 132.64.4.144:2051 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK6e1b25b9;rport=5060 From: ;tag=as19934fd9 To: ;tag=1ipswbr4bt Call-ID: 3c26700d7c83-m8j3bh85uu3m@snom360-0004132384F8 CSeq: 103 NOTIFY Content-Length: 0 <-------------> [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 0 [ 14]: SIP/2.0 200 Ok [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 1 [ 68]: Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK6e1b25b9;rport=5060 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 2 [ 53]: From: ;tag=as19934fd9 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 3 [ 51]: To: ;tag=1ipswbr4bt [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 4 [ 55]: Call-ID: 3c26700d7c83-m8j3bh85uu3m@snom360-0004132384F8 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 5 [ 16]: CSeq: 103 NOTIFY [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 6 [ 17]: Content-Length: 0 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 7 [ 0]: --- (7 headers 0 lines) --- [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:2223 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #8908 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:2234 __sip_ack: Stopping retransmission on '3c26700d7c83-m8j3bh85uu3m@snom360-0004132384F8' of Request 103: Match Found SIP Response message for INCOMING dialog NOTIFY arrived Really destroying SIP dialog 'F3620912-C55611DB-94AE86A5-87E354FD@128.139.26.6' Method: ACK pbx-gr*CLI> <--- SIP read from 132.64.4.121:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK31e05768;rport From: ;tag=as760e1762 To: "80607" ;tag=D7A78ABD-FB1A96BC CSeq: 137 NOTIFY Call-ID: 64aeec39-22798afb-f4d0577a@132.64.4.121 Contact: Event: presence User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.0.3.0127 Content-Length: 0 <-------------> [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 0 [ 14]: SIP/2.0 200 OK [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 1 [ 63]: Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK31e05768;rport [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 2 [ 45]: From: ;tag=as760e1762 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 3 [ 58]: To: "80607" ;tag=D7A78ABD-FB1A96BC [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 4 [ 16]: CSeq: 137 NOTIFY [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 5 [ 48]: Call-ID: 64aeec39-22798afb-f4d0577a@132.64.4.121 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 6 [ 33]: Contact: [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 7 [ 15]: Event: presence [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 8 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.0.3.0127 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 9 [ 17]: Content-Length: 0 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 10 [ 0]: --- (10 headers 0 lines) --- [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:2223 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #8909 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:2234 __sip_ack: Stopping retransmission on '64aeec39-22798afb-f4d0577a@132.64.4.121' of Request 137: Match Found SIP Response message for INCOMING dialog NOTIFY arrived pbx-gr*CLI> <--- SIP read from 132.64.4.141:2051 ---> ACK sip:89444@132.64.9.163 SIP/2.0 Via: SIP/2.0/UDP 132.64.4.141:2051;branch=z9hG4bK-sn73fg85mfbq;rport From: "80602" ;tag=1fnx3usnpb To: "YEHAVI BOURVIN" ;tag=as4e147432 Call-ID: 0203bcd16026f51d4c0e9821089c0e96@132.64.9.163 CSeq: 1 ACK Max-Forwards: 70 Contact: ;flow-id=1 Content-Length: 0 <-------------> [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 0 [ 34]: ACK sip:89444@132.64.9.163 SIP/2.0 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 1 [ 68]: Via: SIP/2.0/UDP 132.64.4.141:2051;branch=z9hG4bK-sn73fg85mfbq;rport [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 2 [ 72]: From: "80602" ;tag=1fnx3usnpb [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 3 [ 60]: To: "YEHAVI BOURVIN" ;tag=as4e147432 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 4 [ 54]: Call-ID: 0203bcd16026f51d4c0e9821089c0e96@132.64.9.163 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 5 [ 11]: CSeq: 1 ACK [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 6 [ 16]: Max-Forwards: 70 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 7 [ 62]: Contact: ;flow-id=1 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 8 [ 17]: Content-Length: 0 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 9 [ 0]: --- (9 headers 0 lines) --- [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:15271 handle_request: **** Received ACK (6) - Command in SIP ACK [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:2223 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #8906 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:2234 __sip_ack: Stopping retransmission on '0203bcd16026f51d4c0e9821089c0e96@132.64.9.163' of Response 1: Match Found [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:6050 reqprep: Strict routing enforced for session 0203bcd16026f51d4c0e9821089c0e96@132.64.9.163 set_destination: Parsing for address/port to send to set_destination: set destination to 132.64.4.141, port 2051 Reliably Transmitting (NAT) to 132.64.4.141:2051: BYE sip:80602@132.64.4.141:2051;line=kbq2n6ok SIP/2.0 Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK198d349a;rport Max-Forwards: 70 From: "YEHAVI BOURVIN" ;tag=as4e147432 To: "80602" ;tag=1fnx3usnpb Call-ID: 0203bcd16026f51d4c0e9821089c0e96@132.64.9.163 CSeq: 104 BYE User-Agent: Asterisk PBX SVN-trunk-r56126M Content-Length: 0 --- [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:2115 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #8913 Scheduling destruction of SIP dialog '0203bcd16026f51d4c0e9821089c0e96@132.64.9.163' in 32000 ms (Method: ACK) pbx-gr*CLI> <--- SIP read from 132.64.4.141:2051 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK198d349a;rport=5060 From: "YEHAVI BOURVIN" ;tag=as4e147432 To: "80602" ;tag=1fnx3usnpb Call-ID: 0203bcd16026f51d4c0e9821089c0e96@132.64.9.163 CSeq: 104 BYE Contact: ;flow-id=1 User-Agent: snom320/6.2.2 Content-Length: 0 <-------------> [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 0 [ 14]: SIP/2.0 200 OK [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 1 [ 68]: Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK198d349a;rport=5060 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 2 [ 62]: From: "YEHAVI BOURVIN" ;tag=as4e147432 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 3 [ 70]: To: "80602" ;tag=1fnx3usnpb [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 4 [ 54]: Call-ID: 0203bcd16026f51d4c0e9821089c0e96@132.64.9.163 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 5 [ 13]: CSeq: 104 BYE [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 6 [ 62]: Contact: ;flow-id=1 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 7 [ 25]: User-Agent: snom320/6.2.2 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 8 [ 17]: Content-Length: 0 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 9 [ 0]: --- (9 headers 0 lines) --- [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:2223 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #8913 [Feb 27 07:01:42] DEBUG[19189]: chan_sip.c:2234 __sip_ack: Stopping retransmission on '0203bcd16026f51d4c0e9821089c0e96@132.64.9.163' of Request 104: Match Found SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog '0203bcd16026f51d4c0e9821089c0e96@132.64.9.163' Method: ACK pbx-gr*CLI> <--- SIP read from 132.64.128.105:43590 ---> <-------------> [Feb 27 07:01:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 0 [ 0]: [Feb 27 07:01:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 0 [ 0]: --- (0 headers 1 lines) --- pbx-gr*CLI> <--- SIP read from 128.139.26.6:53296 ---> BYE sip:80602@132.64.9.163:5060 SIP/2.0 Via: SIP/2.0/UDP 128.139.26.6:5060;branch=z9hG4bK594894 From: "YEHAVI BOURVIN" ;tag=26F00E08-1E38 To: ;tag=as29e8d94e Date: Tue, 27 Feb 2007 05:05:44 GMT Call-ID: F3620912-C55611DB-94AE86A5-87E354FD@128.139.26.6 User-Agent: Cisco-SIPGateway/IOS-12.x Max-Forwards: 70 Timestamp: 1172552755 CSeq: 102 BYE Content-Length: 0 <-------------> [Feb 27 07:01:53] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 0 [ 39]: BYE sip:80602@132.64.9.163:5060 SIP/2.0 [Feb 27 07:01:53] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 1 [ 55]: Via: SIP/2.0/UDP 128.139.26.6:5060;branch=z9hG4bK594894 [Feb 27 07:01:53] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 2 [ 65]: From: "YEHAVI BOURVIN" ;tag=26F00E08-1E38 [Feb 27 07:01:53] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 3 [ 43]: To: ;tag=as29e8d94e [Feb 27 07:01:53] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 4 [ 35]: Date: Tue, 27 Feb 2007 05:05:44 GMT [Feb 27 07:01:53] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 5 [ 57]: Call-ID: F3620912-C55611DB-94AE86A5-87E354FD@128.139.26.6 [Feb 27 07:01:53] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 6 [ 37]: User-Agent: Cisco-SIPGateway/IOS-12.x [Feb 27 07:01:53] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 7 [ 16]: Max-Forwards: 70 [Feb 27 07:01:53] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 8 [ 21]: Timestamp: 1172552755 [Feb 27 07:01:53] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 9 [ 13]: CSeq: 102 BYE [Feb 27 07:01:53] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 10 [ 17]: Content-Length: 0 [Feb 27 07:01:53] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 11 [ 0]: --- (11 headers 0 lines) --- <--- Transmitting (no NAT) to 128.139.26.6:53296 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 128.139.26.6:5060;branch=z9hG4bK594894;received=128.139.26.6 From: "YEHAVI BOURVIN" ;tag=26F00E08-1E38 To: ;tag=as29e8d94e Call-ID: F3620912-C55611DB-94AE86A5-87E354FD@128.139.26.6 CSeq: 102 BYE User-Agent: Asterisk PBX SVN-trunk-r56126M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> [Feb 27 07:01:53] DEBUG[19189]: chan_sip.c:4791 find_call: That's odd... Got a request in unknown dialog. Callid F3620912-C55611DB-94AE86A5-87E354FD@128.139.26.6 [Feb 27 07:01:53] DEBUG[19189]: chan_sip.c:15450 sipsock_read: Invalid SIP message - rejected , no callid, len 422 pbx-gr*CLI> quit Executing last minute cleanups Asterisk ending (0). ======================================================================= Asterisk SVN-trunk-r56126M, 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. ========================================================================= NOTE: This is a development version of Asterisk, and should not be used in production installations. == Parsing '/etc/asterisk/asterisk.conf': Parsing /etc/asterisk/asterisk.conf Found == Parsing '/etc/asterisk/extconfig.conf': Parsing /etc/asterisk/extconfig.conf Found == Binding sippeers to mysql/asterisk/sip_users == Binding sipusers to mysql/asterisk/sip_users == Binding voicemail to mysql/asterisk/voicemail_users Connected to Asterisk SVN-trunk-r56126M currently running on pbx-gr (pid = 19178) pbx-gr*CLI> Verbosity is at least 5 Core debug is at least 5 pbx-gr*CLI> <--- SIP read from 128.139.26.6:50245 ---> INVITE sip:80604@132.64.9.163:5060 SIP/2.0 Via: SIP/2.0/UDP 128.139.26.6:5060;branch=z9hG4bK5AE1582 Remote-Party-ID: "YEHAVI BOURVIN" ;party=calling;screen=no;privacy=off From: "YEHAVI BOURVIN" ;tag=270AED48-1804 To: Date: Tue, 27 Feb 2007 05:34:38 GMT Call-ID: D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6 Supported: 100rel,timer,resource-priority,replaces Min-SE: 1800 Cisco-Guid: 219210638-3311079899-2224029722-801595720 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER CSeq: 101 INVITE Max-Forwards: 70 Timestamp: 1172554478 Contact: Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 247 v=0 o=CiscoSystemsSIP-GW-UserAgent 8474 8949 IN IP4 128.139.26.6 s=SIP Call c=IN IP4 128.139.26.6 t=0 0 m=audio 19256 RTP/AVP 8 101 c=IN IP4 128.139.26.6 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 <-------------> [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 0 [ 42]: INVITE sip:80604@132.64.9.163:5060 SIP/2.0 [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 1 [ 56]: Via: SIP/2.0/UDP 128.139.26.6:5060;branch=z9hG4bK5AE1582 [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 2 [ 94]: Remote-Party-ID: "YEHAVI BOURVIN" ;party=calling;screen=no;privacy=off [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 3 [ 65]: From: "YEHAVI BOURVIN" ;tag=270AED48-1804 [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 4 [ 28]: To: [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 5 [ 35]: Date: Tue, 27 Feb 2007 05:34:38 GMT pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 6 [ 56]: Call-ID: D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6 [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 7 [ 50]: Supported: 100rel,timer,resource-priority,replaces [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 8 [ 13]: Min-SE: 1800 [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 9 [ 53]: Cisco-Guid: 219210638-3311079899-2224029722-801595720 [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 10 [ 37]: User-Agent: Cisco-SIPGateway/IOS-12.x [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 11 [ 97]: Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 12 [ 16]: CSeq: 101 INVITE [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 13 [ 16]: Max-Forwards: 70 [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 14 [ 21]: Timestamp: 1172554478 [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 15 [ 38]: Contact: [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 16 [ 12]: Expires: 180 [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 17 [ 29]: Allow-Events: telephone-event [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 18 [ 29]: Content-Type: application/sdp [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 19 [ 46]: Content-Disposition: session;handling=required [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 20 [ 19]: Content-Length: 247 [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 21 [ 0]: [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 0 [ 3]: v=0 [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 1 [ 60]: o=CiscoSystemsSIP-GW-UserAgent 8474 8949 IN IP4 128.139.26.6 [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 2 [ 10]: s=SIP Call [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 3 [ 21]: c=IN IP4 128.139.26.6 [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 4 [ 5]: t=0 0 [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 5 [ 27]: m=audio 19256 RTP/AVP 8 101 [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 6 [ 21]: c=IN IP4 128.139.26.6 [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 9 [ 15]: a=fmtp:101 0-16 [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 10 [ 10]: a=ptime:20 --- (21 headers 11 lines) --- [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4727 find_call: = No match Their Call ID: 3c26700d7c83-qr0f9ckwnsm1@snom360-0004132384F8 Their Tag mi5g3aesmx Our tag: as27f232fc [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4727 find_call: = No match Their Call ID: 3c26700d7c83-m8j3bh85uu3m@snom360-0004132384F8 Their Tag 1ipswbr4bt Our tag: as4af57878 [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4727 find_call: = No match Their Call ID: 3c26700d7c83-60hzuh819b4l@snom360-0004132384F8 Their Tag 27tc0487d3 Our tag: as3a18e567 [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4727 find_call: = No match Their Call ID: 3c26700c5091-h3po9dsz8ull@snom320-0004132480B4 Their Tag 0xqyctmw0w Our tag: as385f4cf3 [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4727 find_call: = No match Their Call ID: 3c26700c5091-z9why8kfv9np@snom320-0004132480B4 Their Tag st7fjxq5un Our tag: as0c33aea5 [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4727 find_call: = No match Their Call ID: 3c26700c5091-6fozttksacr3@snom320-0004132480B4 Their Tag xriv93rx4h Our tag: as0d02ce20 [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4727 find_call: = No match Their Call ID: 3dacddc4-e99e4206-6bcb53c5@132.64.4.121 Their Tag 3755C748-BA41B887 Our tag: as01510987 [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4727 find_call: = No match Their Call ID: 64aeec39-22798afb-f4d0577a@132.64.4.121 Their Tag D7A78ABD-FB1A96BC Our tag: as760e1762 [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4727 find_call: = No match Their Call ID: d8620a30-ea21a472-fe5c3b31@132.64.4.121 Their Tag 20C25FB4-AE7E55F3 Our tag: as7672f197 [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4727 find_call: = No match Their Call ID: 8004e467-8179a29-4fa4eb28@132.64.4.121 Their Tag 519930EB-5285016A Our tag: as5302e30a [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4727 find_call: = No match Their Call ID: 8d5eade-28d1dc20-204d75f@132.64.4.121 Their Tag B1DA6E62-DC10921 Our tag: as7b7a15e6 [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4727 find_call: = No match Their Call ID: 6e178d95-48cada57-ddda6fd6@132.64.4.121 Their Tag DB388819-37C8DD18 Our tag: as3b303d11 [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4727 find_call: = No match Their Call ID: d0103c8c-b8104ce-f188248d@132.64.4.121 Their Tag 9069EE10-7036ED4F Our tag: as069fd33e [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4727 find_call: = No match Their Call ID: 3b280b12-7a0a228d-8e47726c@132.64.4.135 Their Tag 49667ADF-A4719856 Our tag: as5557dee1 [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4727 find_call: = No match Their Call ID: 8b0a67c3-5276cb85-68746584@132.64.4.121 Their Tag D4291047-F5A9A9C6 Our tag: as45c7a8f4 [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4727 find_call: = No match Their Call ID: 3e849123-fbaf994a-7e2c465@132.64.4.135 Their Tag 16D7DF64-1DDA1477 Our tag: as24fb17f2 [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4727 find_call: = No match Their Call ID: 9b324fb8-b84e063b-fc90d882@132.64.4.135 Their Tag 8EBD433D-EA6E55C Our tag: as29972d1c [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4727 find_call: = No match Their Call ID: ec263f3f-92e9c536-331b9e1@132.64.4.135 Their Tag A4477F30-85603053 Our tag: as7669c5c0 [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4727 find_call: = No match Their Call ID: a5ea4d44-f2eeacd7-89f392ee@132.64.4.135 Their Tag BE7AA539-DE8177A8 Our tag: as2b2c3ef9 [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4727 find_call: = No match Their Call ID: 1441609d-20c6773c-c0e53f6f@132.64.4.135 Their Tag E3C2A1A6-6469BD91 Our tag: as790f216d [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4727 find_call: = No match Their Call ID: f959139a-faf1d075-45de9a34@132.64.4.135 Their Tag 88E25707-5607D15E Our tag: as3fb45c9c [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4727 find_call: = No match Their Call ID: 9948bdcb-e36cd8d2-acc85d4d@132.64.4.135 Their Tag 5C04962C-6412539F Our tag: as5ad1b515 [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:2842 do_setnat: Setting NAT on RTP to Off [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:2847 do_setnat: Setting NAT on VRTP to Off [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:2852 do_setnat: Setting NAT on UDPTL to Off [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4663 sip_alloc: Allocating new SIP dialog for D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6 - INVITE (With RTP) [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:15271 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:1829 parse_sip_options: Begin: parsing SIP "Supported: 100rel,timer,resource-priority,replaces" [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:1837 parse_sip_options: Found SIP option: -100rel- [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:1843 parse_sip_options: Matched SIP option: 100rel [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:1837 parse_sip_options: Found SIP option: -timer- [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:1843 parse_sip_options: Matched SIP option: timer [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:1837 parse_sip_options: Found SIP option: -resource-priority- [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:1843 parse_sip_options: Matched SIP option: resource-priority [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:1837 parse_sip_options: Found SIP option: -replaces- [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:1843 parse_sip_options: Matched SIP option: replaces Sending to 128.139.26.6 : 5060 (no NAT) [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:13975 handle_request_invite: Initializing initreq for method INVITE - callid D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6 Using INVITE request as basis request - D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6 pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: res_config_mysql.c:651 mysql_reconnect: MySQL RealTime: Everything is fine. [Feb 27 07:30:37] DEBUG[19189]: res_config_mysql.c:139 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_users WHERE name = '89444' pbx-gr*CLI> No user '89444' in SIP users list Found peer '128.139.26.6' for '89444' from 128.139.26.6:50245 [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:2842 do_setnat: Setting NAT on RTP to Off [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:2847 do_setnat: Setting NAT on VRTP to Off [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:2852 do_setnat: Setting NAT on UDPTL to Off Found RTP audio format 8 Found RTP audio format 101 [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:5282 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 128.139.26.6:19256 Found description format PCMA for ID 8 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:5524 process_sdp: T38 state changed to 0 on channel Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=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 128.139.26.6:19256 [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:5615 process_sdp: We're settling with these formats: 0x8 (alaw) [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:14056 handle_request_invite: Checking SIP call limits for device gr-pbx-link [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:3305 update_call_counter: Updating call counter for incoming call Looking for 80604 in huji-remote-gr (domain 132.64.9.163) [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4126 sip_new: *** Our native formats are 0x8 (alaw) [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4127 sip_new: *** Joint capabilities are 0x8 (alaw) [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4128 sip_new: *** Our capabilities are 0x8 (alaw) [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4129 sip_new: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4159 sip_new: This channel will not be able to handle video. [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:8478 build_route: build_route: Contact hop: list_route: hop: [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:14131 handle_request_invite: SIP/gr-pbx-link-0960d288: New call is still down.... Trying... <--- Transmitting (no NAT) to 128.139.26.6:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 128.139.26.6:5060;branch=z9hG4bK5AE1582;received=128.139.26.6 From: "YEHAVI BOURVIN" ;tag=270AED48-1804 To: Call-ID: D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6 CSeq: 101 INVITE User-Agent: Asterisk PBX SVN-trunk-r56126M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 27 07:30:37] DEBUG[19189]: devicestate.c:444 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/gr-pbx-link-0960d288 pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: devicestate.c:302 ast_device_state: No provider found, checking channel drivers for SIP - gr-pbx-link [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:15897 sip_devicestate: Checking device state for peer gr-pbx-link [Feb 27 07:30:37] DEBUG[19189]: res_config_mysql.c:651 mysql_reconnect: MySQL RealTime: Everything is fine. [Feb 27 07:30:37] DEBUG[19189]: res_config_mysql.c:139 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_users WHERE name = 'gr-pbx-link' [Feb 27 07:30:37] DEBUG[19189]: pbx.c:1693 pbx_extension_helper: Launching 'Set' -- Executing [80604@huji-remote-gr:1] Set("SIP/gr-pbx-link-0960d288", "_To=80604") in new stack [Feb 27 07:30:37] DEBUG[19189]: pbx.c:1543 pbx_substitute_variables_helper_full: Function result is '89444' [Feb 27 07:30:37] DEBUG[19189]: pbx.c:1693 pbx_extension_helper: Launching 'Set' -- Executing [80604@huji-remote-gr:2] Set("SIP/gr-pbx-link-0960d288", "_From=89444") in new stack [Feb 27 07:30:37] DEBUG[19189]: pbx.c:1693 pbx_extension_helper: Launching 'Set' -- Executing [80604@huji-remote-gr:3] Set("SIP/gr-pbx-link-0960d288", "DB(80604/LastCaller)=89444") in new stack pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: devicestate.c:428 do_state_change: Changing state for SIP/gr-pbx-link - state 4 (Invalid) pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: app_queue.c:568 changethread: Device 'SIP/gr-pbx-link' changed to state '4' (Invalid) but we don't care because they're not a member of any queue. pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: pbx.c:1693 pbx_extension_helper: Launching 'Set' pbx-gr*CLI> -- Executing [80604@huji-remote-gr:4] Set("SIP/gr-pbx-link-0960d288", "DB(89444/LastCalled)=80604") in new stack pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: db.c:198 ast_db_get: Unable to find key 'MoreLinesToRing' in family '80604' pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: func_db.c:73 function_db_read: DB: 80604/MoreLinesToRing not found in database. pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: pbx.c:1543 pbx_substitute_variables_helper_full: Function result is '' pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: pbx.c:1693 pbx_extension_helper: Launching 'Set' pbx-gr*CLI> -- Executing [80604@huji-remote-gr:5] Set("SIP/gr-pbx-link-0960d288", "aEXTEN=") in new stack pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: pbx.c:1693 pbx_extension_helper: Launching 'Dial' pbx-gr*CLI> -- Executing [80604@huji-remote-gr:6] Dial("SIP/gr-pbx-link-0960d288", "SIP/80604|20|") in new stack pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:15966 sip_request_call: Asked to create a SIP channel with formats: 0x8 (alaw) pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4663 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:2897 create_addr_from_peer: Our T38 capability (3856) pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:2842 do_setnat: Setting NAT on RTP to Off pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:2852 do_setnat: Setting NAT on UDPTL to Off pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:1783 obproxy_get: OBPROXY: Not applying OBproxy to this call pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4126 sip_new: *** Our native formats are 0x8 (alaw) pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4127 sip_new: *** Joint capabilities are 0x8 (alaw) pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4128 sip_new: *** Our capabilities are 0x8 (alaw) pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4129 sip_new: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4131 sip_new: *** Our preferred formats from the incoming channel are 0x8 (alaw) pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4159 sip_new: This channel will not be able to handle video. pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: rtp.c:1677 ast_rtp_make_compatible: Seeded SDP of 'SIP/80604-096128e8' with that of 'SIP/gr-pbx-link-0960d288' pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: channel.c:3211 ast_channel_inherit_variables: Not copying variable STACK-huji-remote-gr-80604-6. pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: channel.c:3211 ast_channel_inherit_variables: Not copying variable aEXTEN. pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: channel.c:3211 ast_channel_inherit_variables: Not copying variable STACK-huji-remote-gr-80604-5. pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: channel.c:3211 ast_channel_inherit_variables: Not copying variable STACK-huji-remote-gr-80604-4. pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: channel.c:3211 ast_channel_inherit_variables: Not copying variable STACK-huji-remote-gr-80604-3. pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: channel.c:3198 ast_channel_inherit_variables: Copying soft-transferable variable From. pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: channel.c:3211 ast_channel_inherit_variables: Not copying variable STACK-huji-remote-gr-80604-2. pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: channel.c:3198 ast_channel_inherit_variables: Copying soft-transferable variable To. pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: channel.c:3211 ast_channel_inherit_variables: Not copying variable STACK-huji-remote-gr-80604-1. pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: channel.c:3211 ast_channel_inherit_variables: Not copying variable SIPCALLID. pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: channel.c:3211 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: channel.c:3211 ast_channel_inherit_variables: Not copying variable SIPURI. pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:3122 sip_call: Outgoing Call for 80604 pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:3305 update_call_counter: Updating call counter for outgoing call pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:3140 sip_call: Our T38 capability (3856), joint T38 capability (3856) pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:6674 add_sdp: ** Our capability: 0x8 (alaw) Video flag: False Text flag: False pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:6676 add_sdp: ** Our prefcodec: 0x8 (alaw) pbx-gr*CLI> Audio is at 132.64.9.163 port 64524 pbx-gr*CLI> Adding codec 0x8 (alaw) to SDP pbx-gr*CLI> Adding non-codec 0x1 (telephone-event) to SDP pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:6837 add_sdp: -- Done with adding codecs to SDP pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:6897 add_sdp: Done building SDP. Settling with this capability: 0x8 (alaw) pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:1718 initialize_initreq: Initializing initreq for method INVITE - callid 7921261c58176f0d7c530e722904adff@132.64.9.163 pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 0 [ 67]: INVITE sip:80604@132.64.4.128:5060;user=phone;transport=udp SIP/2.0 pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 1 [ 63]: Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK62e00341;rport pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 2 [ 16]: Max-Forwards: 70 pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 3 [ 62]: From: "YEHAVI BOURVIN" ;tag=as668cf442 pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 4 [ 58]: To: pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 5 [ 33]: Contact: pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 6 [ 54]: Call-ID: 7921261c58176f0d7c530e722904adff@132.64.9.163 pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 7 [ 16]: CSeq: 102 INVITE pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 8 [ 42]: User-Agent: Asterisk PBX SVN-trunk-r56126M pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 9 [ 35]: Date: Tue, 27 Feb 2007 05:30:37 GMT pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 10 [ 66]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 11 [ 19]: Supported: replaces pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 12 [ 29]: Content-Type: application/sdp pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 13 [ 19]: Content-Length: 240 pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 14 [ 0]: pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 0 [ 3]: v=0 pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 1 [ 38]: o=root 19178 19178 IN IP4 132.64.9.163 pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 2 [ 9]: s=session pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 3 [ 21]: c=IN IP4 132.64.9.163 pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 4 [ 5]: t=0 0 pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 5 [ 27]: m=audio 64524 RTP/AVP 8 101 pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 7 [ 33]: a=rtpmap:101 telephone-event/8000 pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 8 [ 15]: a=fmtp:101 0-16 pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 9 [ 25]: a=silenceSupp:off - - - - pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 10 [ 10]: a=ptime:20 pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 11 [ 10]: a=sendrecv pbx-gr*CLI> Reliably Transmitting (no NAT) to 132.64.4.128:5060: INVITE sip:80604@132.64.4.128:5060;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK62e00341;rport Max-Forwards: 70 From: "YEHAVI BOURVIN" ;tag=as668cf442 To: Contact: Call-ID: 7921261c58176f0d7c530e722904adff@132.64.9.163 CSeq: 102 INVITE User-Agent: Asterisk PBX SVN-trunk-r56126M Date: Tue, 27 Feb 2007 05:30:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 240 v=0 o=root 19178 19178 IN IP4 132.64.9.163 s=session c=IN IP4 132.64.9.163 t=0 0 m=audio 64524 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:2115 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #9282 pbx-gr*CLI> -- Called 80604 pbx-gr*CLI> <--- SIP read from 132.64.4.128:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK62e00341;rport From: "YEHAVI BOURVIN" ;tag=as668cf442 To: ;tag=1268759603 Call-ID: 7921261c58176f0d7c530e722904adff@132.64.9.163 CSeq: 102 INVITE Server: Cisco-CP7912/8.0.1-060412A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Content-Length: 0 <-------------> [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 0 [ 18]: SIP/2.0 100 Trying [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 1 [ 63]: Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK62e00341;rport [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 2 [ 62]: From: "YEHAVI BOURVIN" ;tag=as668cf442 [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 3 [ 73]: To: ;tag=1268759603 [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 4 [ 54]: Call-ID: 7921261c58176f0d7c530e722904adff@132.64.9.163 [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 5 [ 16]: CSeq: 102 INVITE [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 6 [ 34]: Server: Cisco-CP7912/8.0.1-060412A [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 7 [ 80]: Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 8 [ 17]: Content-Length: 0 [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 9 [ 0]: --- (9 headers 0 lines) --- [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4727 find_call: = Found Their Call ID: 7921261c58176f0d7c530e722904adff@132.64.9.163 Their Tag Our tag: as668cf442 [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:2268 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #9282 - INVITE (got response) [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:2277 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '7921261c58176f0d7c530e722904adff@132.64.9.163' Request 102: Found [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:12287 handle_response_invite: SIP response 100 to standard invite pbx-gr*CLI> <--- SIP read from 132.64.4.128:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK62e00341;rport From: "YEHAVI BOURVIN" ;tag=as668cf442 To: ;tag=1268759603 Call-ID: 7921261c58176f0d7c530e722904adff@132.64.9.163 CSeq: 102 INVITE Contact: Yehavi 7912 Server: Cisco-CP7912/8.0.1-060412A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Content-Length: 0 <-------------> [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 0 [ 19]: SIP/2.0 180 Ringing [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 1 [ 63]: Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK62e00341;rport [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 2 [ 62]: From: "YEHAVI BOURVIN" ;tag=as668cf442 [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 3 [ 73]: To: ;tag=1268759603 [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 4 [ 54]: Call-ID: 7921261c58176f0d7c530e722904adff@132.64.9.163 [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 5 [ 16]: CSeq: 102 INVITE [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 6 [ 75]: Contact: Yehavi 7912 [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 7 [ 34]: Server: Cisco-CP7912/8.0.1-060412A [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 8 [ 80]: Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 9 [ 17]: Content-Length: 0 [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 10 [ 0]: --- (10 headers 0 lines) --- [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:4727 find_call: = Found Their Call ID: 7921261c58176f0d7c530e722904adff@132.64.9.163 Their Tag 1268759603 Our tag: as668cf442 [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:2277 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '7921261c58176f0d7c530e722904adff@132.64.9.163' Request 102: Found [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:12287 handle_response_invite: SIP response 180 to standard invite [Feb 27 07:30:37] DEBUG[19189]: devicestate.c:444 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/80604-096128e8 pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: devicestate.c:302 ast_device_state: No provider found, checking channel drivers for SIP - 80604 [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:15897 sip_devicestate: Checking device state for peer 80604 [Feb 27 07:30:37] DEBUG[19189]: devicestate.c:428 do_state_change: Changing state for SIP/80604 - state 8 (On Hold) [Feb 27 07:30:37] DEBUG[19189]: devicestate.c:302 ast_device_state: No provider found, checking channel drivers for SIP - 80604 [Feb 27 07:30:37] DEBUG[19189]: chan_sip.c:15897 sip_devicestate: Checking device state for peer 80604 -- SIP/80604-096128e8 is ringing [Feb 27 07:30:37] DEBUG[19189]: rtp.c:1607 ast_rtp_early_bridge: Setting early bridge SDP of 'SIP/gr-pbx-link-0960d288' with that of 'SIP/80604-096128e8' <--- Transmitting (no NAT) to 128.139.26.6:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 128.139.26.6:5060;branch=z9hG4bK5AE1582;received=128.139.26.6 From: "YEHAVI BOURVIN" ;tag=270AED48-1804 To: ;tag=as268edf01 Call-ID: D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6 CSeq: 101 INVITE User-Agent: Asterisk PBX SVN-trunk-r56126M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> pbx-gr*CLI> [Feb 27 07:30:37] DEBUG[19189]: app_queue.c:568 changethread: Device 'SIP/80604' changed to state '8' (On Hold) but we don't care because they're not a member of any queue. pbx-gr*CLI> <--- SIP read from 132.64.4.128:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK62e00341;rport From: "YEHAVI BOURVIN" ;tag=as668cf442 To: ;tag=1268759603 Call-ID: 7921261c58176f0d7c530e722904adff@132.64.9.163 CSeq: 102 INVITE Contact: Yehavi 7912 Server: Cisco-CP7912/8.0.1-060412A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces Content-Length: 204 Content-Type: application/sdp v=0 o=80604 52550 52550 IN IP4 132.64.4.128 s=Cisco 7912 SIP Call c=IN IP4 132.64.4.128 t=0 0 m=audio 16384 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 0 [ 14]: SIP/2.0 200 OK [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 1 [ 63]: Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK62e00341;rport [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 2 [ 62]: From: "YEHAVI BOURVIN" ;tag=as668cf442 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 3 [ 73]: To: ;tag=1268759603 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 4 [ 54]: Call-ID: 7921261c58176f0d7c530e722904adff@132.64.9.163 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 5 [ 16]: CSeq: 102 INVITE [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 6 [ 75]: Contact: Yehavi 7912 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 7 [ 34]: Server: Cisco-CP7912/8.0.1-060412A [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 8 [ 80]: Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 9 [ 19]: Supported: replaces [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 10 [ 19]: Content-Length: 204 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 11 [ 29]: Content-Type: application/sdp [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 12 [ 0]: [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 0 [ 3]: v=0 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 1 [ 39]: o=80604 52550 52550 IN IP4 132.64.4.128 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 2 [ 21]: s=Cisco 7912 SIP Call [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 3 [ 21]: c=IN IP4 132.64.4.128 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 4 [ 5]: t=0 0 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 5 [ 27]: m=audio 16384 RTP/AVP 8 101 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 6 [ 22]: a=rtpmap:8 PCMA/8000/1 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 7 [ 33]: a=rtpmap:101 telephone-event/8000 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 8 [ 15]: a=fmtp:101 0-15 --- (12 headers 9 lines) --- [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4727 find_call: = Found Their Call ID: 7921261c58176f0d7c530e722904adff@132.64.9.163 Their Tag 1268759603 Our tag: as668cf442 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:2218 __sip_ack: Acked pending invite 102 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:2234 __sip_ack: Stopping retransmission on '7921261c58176f0d7c530e722904adff@132.64.9.163' of Request 102: Match Found [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:12287 handle_response_invite: SIP response 200 to standard invite Found RTP audio format 8 Found RTP audio format 101 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:5282 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 132.64.4.128:16384 Found description format PCMA for ID 8 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:5524 process_sdp: T38 state changed to 0 on channel SIP/80604-096128e8 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=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 132.64.4.128:16384 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:5615 process_sdp: We're settling with these formats: 0x8 (alaw) [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:5622 process_sdp: We have an owner, now see if we need to change this call [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:3305 update_call_counter: Updating call counter for outgoing call --- set_address_from_contact host '132.64.4.128' [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:8478 build_route: build_route: Contact hop: Yehavi 7912 list_route: hop: [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:6050 reqprep: Strict routing enforced for session 7921261c58176f0d7c530e722904adff@132.64.9.163 set_destination: Parsing for address/port to send to set_destination: set destination to 132.64.4.128, port 5060 Transmitting (no NAT) to 132.64.4.128:5060: ACK sip:80604@132.64.4.128:5060;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK4806c41b;rport Max-Forwards: 70 From: "YEHAVI BOURVIN" ;tag=as668cf442 To: ;tag=1268759603 Contact: Call-ID: 7921261c58176f0d7c530e722904adff@132.64.9.163 CSeq: 102 ACK User-Agent: Asterisk PBX SVN-trunk-r56126M Content-Length: 0 --- pbx-gr*CLI> -- Call on SIP/80604-096128e8 left from hold [Feb 27 07:30:40] DEBUG[19189]: devicestate.c:444 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/80604-096128e8 -- SIP/80604-096128e8 answered SIP/gr-pbx-link-0960d288 [Feb 27 07:30:40] DEBUG[19189]: rtp.c:1607 ast_rtp_early_bridge: Setting early bridge SDP of 'SIP/gr-pbx-link-0960d288' with that of 'SIP/80604-096128e8' [Feb 27 07:30:40] DEBUG[19189]: devicestate.c:444 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/gr-pbx-link-0960d288 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:3761 sip_answer: SIP answering channel: SIP/gr-pbx-link-0960d288 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:6952 transmit_response_with_sdp: Setting framing from config on incoming call [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:6674 add_sdp: ** Our capability: 0x8 (alaw) Video flag: True Text flag: True [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:6676 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 132.64.9.163 port 27044 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:6837 add_sdp: -- Done with adding codecs to SDP [Feb 27 07:30:40] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:6897 add_sdp: Done building SDP. Settling with this capability: 0x8 (alaw) <--- Reliably Transmitting (no NAT) to 128.139.26.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 128.139.26.6:5060;branch=z9hG4bK5AE1582;received=128.139.26.6 From: "YEHAVI BOURVIN" ;tag=270AED48-1804 To: ;tag=as268edf01 Call-ID: D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6 CSeq: 101 INVITE User-Agent: Asterisk PBX SVN-trunk-r56126M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 240 v=0 o=root 19178 19178 IN IP4 132.64.9.163 s=session c=IN IP4 132.64.9.163 t=0 0 m=audio 27044 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 apbx-gr*CLI> =fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:2115 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #9284 [Feb 27 07:30:40] DEBUG[19189]: devicestate.c:302 ast_device_state: No provider found, checking channel drivers for SIP - 80604 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:15897 sip_devicestate: Checking device state for peer 80604 [Feb 27 07:30:40] DEBUG[19189]: devicestate.c:428 do_state_change: Changing state for SIP/80604 - state 8 (On Hold) [Feb 27 07:30:40] DEBUG[19189]: devicestate.c:302 ast_device_state: No provider found, checking channel drivers for SIP - 80604 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:15897 sip_devicestate: Checking device state for peer 80604 [Feb 27 07:30:40] DEBUG[19189]: devicestate.c:302 ast_device_state: No provider found, checking channel drivers for SIP - gr-pbx-link [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:15897 sip_devicestate: Checking device state for peer gr-pbx-link [Feb 27 07:30:40] DEBUG[19189]: res_config_mysql.c:651 mysql_reconnect: MySQL RealTime: Everything is fine. [Feb 27 07:30:40] DEBUG[19189]: res_config_mysql.c:139 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_users WHERE name = 'gr-pbx-link' pbx-gr*CLI> [Feb 27 07:30:40] DEBUG[19189]: devicestate.c:428 do_state_change: Changing state for SIP/gr-pbx-link - state 4 (Invalid) pbx-gr*CLI> [Feb 27 07:30:40] DEBUG[19189]: app_queue.c:568 changethread: Device 'SIP/80604' changed to state '8' (On Hold) but we don't care because they're not a member of any queue. pbx-gr*CLI> [Feb 27 07:30:40] DEBUG[19189]: app_queue.c:568 changethread: Device 'SIP/gr-pbx-link' changed to state '4' (Invalid) but we don't care because they're not a member of any queue. pbx-gr*CLI> <--- SIP read from 128.139.26.6:50245 ---> ACK sip:80604@132.64.9.163:5060 SIP/2.0 Via: SIP/2.0/UDP 128.139.26.6:5060;branch=z9hG4bK5AF1FD From: "YEHAVI BOURVIN" ;tag=270AED48-1804 To: ;tag=as268edf01 Date: Tue, 27 Feb 2007 05:34:38 GMT Call-ID: D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6 Max-Forwards: 70 CSeq: 101 ACK Allow-Events: telephone-event Content-Length: 0 <-------------> [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 0 [ 39]: ACK sip:80604@132.64.9.163:5060 SIP/2.0 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 1 [ 55]: Via: SIP/2.0/UDP 128.139.26.6:5060;branch=z9hG4bK5AF1FD [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 2 [ 65]: From: "YEHAVI BOURVIN" ;tag=270AED48-1804 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 3 [ 43]: To: ;tag=as268edf01 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 4 [ 35]: Date: Tue, 27 Feb 2007 05:34:38 GMT [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 5 [ 56]: Call-ID: D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 6 [ 16]: Max-Forwards: 70 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 7 [ 13]: CSeq: 101 ACK [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 8 [ 29]: Allow-Events: telephone-event [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 9 [ 17]: Content-Length: 0 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 10 [ 0]: --- (10 headers 0 lines) --- [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4727 find_call: = No match Their Call ID: 7921261c58176f0d7c530e722904adff@132.64.9.163 Their Tag 1268759603 Our tag: as668cf442 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4727 find_call: = Found Their Call ID: D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6 Their Tag 270AED48-1804 Our tag: as268edf01 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:15271 handle_request: **** Received ACK (6) - Command in SIP ACK [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:2223 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #9284 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:2234 __sip_ack: Stopping retransmission on 'D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6' of Response 101: Match Found pbx-gr*CLI> -- Native bridging SIP/gr-pbx-link-0960d288 and SIP/80604-096128e8 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:17627 sip_set_rtp_peer: Sending reinvite on SIP 'D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6' - It's audio soon redirected to IP 132.64.4.128 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:6050 reqprep: Strict routing enforced for session D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6 set_destination: Parsing for address/port to send to set_destination: set destination to 128.139.26.6, port 5060 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:6674 add_sdp: ** Our capability: 0x8 (alaw) Video flag: True Text flag: True [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:6676 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 132.64.9.163 port 27044 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:6837 add_sdp: -- Done with adding codecs to SDP [Feb 27 07:30:40] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:6897 add_sdp: Done building SDP. Settling with this capability: 0x8 (alaw) [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:1716 initialize_initreq: Initializing already initialized SIP dialog D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6 (presumably reinvite) [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 0 [ 42]: INVITE sip:89444@128.139.26.6:5060 SIP/2.0 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 1 [ 63]: Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK476a495c;rport [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 2 [ 16]: Max-Forwards: 70 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 3 [ 45]: From: ;tag=as268edf01 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 4 [ 63]: To: "YEHAVI BOURVIN" ;tag=270AED48-1804 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 5 [ 33]: Contact: [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 6 [ 56]: Call-ID: D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 7 [ 16]: CSeq: 102 INVITE [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 8 [ 42]: User-Agent: Asterisk PBX SVN-trunk-r56126M [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 9 [ 66]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 10 [ 19]: Supported: replaces [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 12 [ 29]: Content-Type: application/sdp [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 13 [ 19]: Content-Length: 240 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 14 [ 0]: [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 0 [ 3]: v=0 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 1 [ 38]: o=root 19178 19179 IN IP4 132.64.4.128 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 2 [ 9]: s=session [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 3 [ 21]: c=IN IP4 132.64.4.128 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 4 [ 5]: t=0 0 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 5 [ 27]: m=audio 16384 RTP/AVP 8 101 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 7 [ 33]: a=rtpmap:101 telephone-event/8000 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 8 [ 15]: a=fmtp:101 0-16 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 9 [ 25]: a=silenceSupp:off - - - - [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 10 [ 10]: a=ptime:20 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 11 [ 10]: a=sendrecv Reliably Transmitting (no NAT) to 128.139.26.6:5060: INVITE sip:89444@128.139.26.6:5060 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK476a495c;rport Max-Forwards: 70 From: ;tag=as268edf01 To: "YEHAVI BOURVIN" ;tag=270AED48-1804 Contact: Call-ID: D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6 CSeq: 102 INVITE User-Agent: Asterisk PBX SVN-trunk-r56126M 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: 240 v=0 o=root 19178 19179 IN IP4 132.64.4.128 s=session c=IN IP4 132.64.4.128 t=0 0 m=audio 16384 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- pbx-gr*CLI> [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:2115 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #9286 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:17627 sip_set_rtp_peer: Sending reinvite on SIP '7921261c58176f0d7c530e722904adff@132.64.9.163' - It's audio soon redirected to IP 128.139.26.6 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:6050 reqprep: Strict routing enforced for session 7921261c58176f0d7c530e722904adff@132.64.9.163 set_destination: Parsing for address/port to send to set_destination: set destination to 132.64.4.128, port 5060 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:6674 add_sdp: ** Our capability: 0x8 (alaw) Video flag: True Text flag: True [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:6676 add_sdp: ** Our prefcodec: 0x8 (alaw) Audio is at 132.64.9.163 port 64524 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:6837 add_sdp: -- Done with adding codecs to SDP [Feb 27 07:30:40] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:6897 add_sdp: Done building SDP. Settling with this capability: 0x8 (alaw) [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:1716 initialize_initreq: Initializing already initialized SIP dialog 7921261c58176f0d7c530e722904adff@132.64.9.163 (presumably reinvite) [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 0 [ 67]: INVITE sip:80604@132.64.4.128:5060;user=phone;transport=udp SIP/2.0 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 1 [ 63]: Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK1022d9c4;rport [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 2 [ 16]: Max-Forwards: 70 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 3 [ 62]: From: "YEHAVI BOURVIN" ;tag=as668cf442 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 4 [ 73]: To: ;tag=1268759603 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 5 [ 33]: Contact: [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 6 [ 54]: Call-ID: 7921261c58176f0d7c530e722904adff@132.64.9.163 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 7 [ 16]: CSeq: 103 INVITE [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 8 [ 42]: User-Agent: Asterisk PBX SVN-trunk-r56126M [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 9 [ 66]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 10 [ 19]: Supported: replaces [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 12 [ 29]: Content-Type: application/sdp [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 13 [ 19]: Content-Length: 240 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 14 [ 0]: [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 0 [ 3]: v=0 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 1 [ 38]: o=root 19178 19179 IN IP4 128.139.26.6 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 2 [ 9]: s=session [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 3 [ 21]: c=IN IP4 128.139.26.6 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 4 [ 5]: t=0 0 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 5 [ 27]: m=audio 19256 RTP/AVP 8 101 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 7 [ 33]: a=rtpmap:101 telephone-event/8000 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 8 [ 15]: a=fmtp:101 0-16 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 9 [ 25]: a=silenceSupp:off - - - - [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 10 [ 10]: a=ptime:20 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 11 [ 10]: a=sendrecv Reliably Transmitting (no NAT) to 132.64.4.128:5060: INVITE sip:80604@132.64.4.128:5060;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK1022d9c4;rport Max-Forwards: 70 From: "YEHAVI BOURVIN" ;tag=as668cf442 To: ;tag=1268759603 Contact: Call-ID: 7921261c58176f0d7c530e722904adff@132.64.9.163 CSeq: 103 INVITE User-Agent: Asterisk PBX SVN-trunk-r56126M 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: 240 v=0 o=root 19178 19179 IN IP4 128.139.26.6 s=session c=IN IP4 128.139.26.6 t=0 0 m=audio 19256 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:2115 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #9287 [Feb 27 07:30:40] DEBUG[19189]: rtp.c:2792 ast_rtp_write: Ooh, format changed from unknown to alaw [Feb 27 07:30:40] DEBUG[19189]: rtp.c:2809 ast_rtp_write: Created smoother: format: 8 ms: 20 len: 160 pbx-gr*CLI> [Feb 27 07:30:40] DEBUG[19189]: rtp.c:2792 ast_rtp_write: Ooh, format changed from unknown to alaw [Feb 27 07:30:40] DEBUG[19189]: rtp.c:2809 ast_rtp_write: Created smoother: format: 8 ms: 20 len: 160 pbx-gr*CLI> [Feb 27 07:30:40] DEBUG[19189]: rtp.c:941 ast_rtcp_read: Got RTCP report of 72 bytes pbx-gr*CLI> <--- SIP read from 128.139.26.6:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK476a495c;rport From: ;tag=as268edf01 To: "YEHAVI BOURVIN" ;tag=270AED48-1804 Date: Tue, 27 Feb 2007 05:34:42 GMT Call-ID: D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6 Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow-Events: telephone-event Remote-Party-ID: "YEHAVI BOURVIN" ;party=called;screen=no;privacy=off Content-Length: 0 <-------------> [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 0 [ 18]: SIP/2.0 100 Trying [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 1 [ 63]: Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK476a495c;rport [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 2 [ 45]: From: ;tag=as268edf01 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 3 [ 63]: To: "YEHAVI BOURVIN" ;tag=270AED48-1804 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 4 [ 35]: Date: Tue, 27 Feb 2007 05:34:42 GMT [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 5 [ 56]: Call-ID: D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 6 [ 33]: Server: Cisco-SIPGateway/IOS-12.x [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 7 [ 16]: CSeq: 102 INVITE [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 8 [ 29]: Allow-Events: telephone-event [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 9 [ 93]: Remote-Party-ID: "YEHAVI BOURVIN" ;party=called;screen=no;privacy=off [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 10 [ 17]: Content-Length: 0 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 11 [ 0]: --- (11 headers 0 lines) --- [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4727 find_call: = No match Their Call ID: 7921261c58176f0d7c530e722904adff@132.64.9.163 Their Tag 1268759603 Our tag: as668cf442 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4727 find_call: = Found Their Call ID: D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6 Their Tag 270AED48-1804 Our tag: as268edf01 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:2268 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #9286 - INVITE (got response) [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:2277 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on 'D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6' Request 102: Found [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:12285 handle_response_invite: SIP response 100 to RE-invite on outgoing call D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6 pbx-gr*CLI> <--- SIP read from 128.139.26.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK476a495c;rport From: ;tag=as268edf01 To: "YEHAVI BOURVIN" ;tag=270AED48-1804 Date: Tue, 27 Feb 2007 05:34:42 GMT Call-ID: D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6 Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER Allow-Events: telephone-event Remote-Party-ID: "YEHAVI BOURVIN" ;party=called;screen=no;privacy=off Contact: Supported: replaces Content-Type: application/sdp Content-Length: 218 v=0 o=CiscoSystemsSIP-GW-UserAgent 8474 8950 IN IP4 128.139.26.6 s=SIP Call c=IN IP4 128.139.26.6 t=0 0 m=audio 19256 RTP/AVP 8 c=IN IP4 128.139.26.6 a=rtpmap:8 PCMA/8000 a=ptime:20 a=silenceSupp:off - - - - <-------------> [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 0 [ 14]: SIP/2.0 200 OK [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 1 [ 63]: Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK476a495c;rport [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 2 [ 45]: From: ;tag=as268edf01 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 3 [ 63]: To: "YEHAVI BOURVIN" ;tag=270AED48-1804 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 4 [ 35]: Date: Tue, 27 Feb 2007 05:34:42 GMT [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 5 [ 56]: Call-ID: D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 6 [ 33]: Server: Cisco-SIPGateway/IOS-12.x [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 7 [ 16]: CSeq: 102 INVITE [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 8 [ 97]: Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 9 [ 29]: Allow-Events: telephone-event [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 10 [ 93]: Remote-Party-ID: "YEHAVI BOURVIN" ;party=called;screen=no;privacy=off [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 11 [ 38]: Contact: [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 12 [ 19]: Supported: replaces [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 13 [ 29]: Content-Type: application/sdp [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 14 [ 19]: Content-Length: 218 pbx-gr*CLI> [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 15 [ 0]: [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 0 [ 3]: v=0 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 1 [ 60]: o=CiscoSystemsSIP-GW-UserAgent 8474 8950 IN IP4 128.139.26.6 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 2 [ 10]: s=SIP Call [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 3 [ 21]: c=IN IP4 128.139.26.6 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 4 [ 5]: t=0 0 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 5 [ 23]: m=audio 19256 RTP/AVP 8 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 6 [ 21]: c=IN IP4 128.139.26.6 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 8 [ 10]: a=ptime:20 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 9 [ 25]: a=silenceSupp:off - - - - --- (15 headers 10 lines) --- [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4727 find_call: = No match Their Call ID: 7921261c58176f0d7c530e722904adff@132.64.9.163 Their Tag 1268759603 Our tag: as668cf442 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4727 find_call: = Found Their Call ID: D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6 Their Tag 270AED48-1804 Our tag: as268edf01 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:2218 __sip_ack: Acked pending invite 102 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:2234 __sip_ack: Stopping retransmission on 'D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6' of Request 102: Match Found [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:12285 handle_response_invite: SIP response 200 to RE-invite on outgoing call D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6 Found RTP audio format 8 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:5282 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 128.139.26.6:19256 Found description format PCMA for ID 8 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:5524 process_sdp: T38 state changed to 0 on channel SIP/gr-pbx-link-0960d288 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 128.139.26.6:19256 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:5615 process_sdp: We're settling with these formats: 0x8 (alaw) [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:5622 process_sdp: We have an owner, now see if we need to change this call [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:3305 update_call_counter: Updating call counter for incoming call --- set_address_from_contact host '128.139.26.6' [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:8478 build_route: build_route: Contact hop: list_route: hop: [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:6050 reqprep: Strict routing enforced for session D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6 set_destination: Parsing for address/port to send to set_destination: set destination to 128.139.26.6, port 5060 Transmitting (no NAT) to 128.139.26.6:5060: ACK sip:89444@128.139.26.6:5060 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK7cb7dcb7;rport Max-Forwards: 70 From: ;tag=as268edf01 To: "YEHAVI BOURVIN" ;tag=270AED48-1804 Contact: Call-ID: D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6 CSeq: 102 ACK User-Agent: Asterisk PBX SVN-trunk-r56126M Content-Length: 0 --- pbx-gr*CLI> <--- SIP read from 132.64.4.128:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK1022d9c4;rport From: "YEHAVI BOURVIN" ;tag=as668cf442 To: ;tag=1268759603 Call-ID: 7921261c58176f0d7c530e722904adff@132.64.9.163 CSeq: 103 INVITE Contact: Yehavi 7912 Server: Cisco-CP7912/8.0.1-060412A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces Content-Length: 204 Content-Type: application/sdp v=0 o=80604 52605 52605 IN IP4 132.64.4.128 s=Cisco 7912 SIP Call c=IN IP4 132.64.4.128 t=0 0 m=audio 16384 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 0 [ 14]: SIP/2.0 200 OK [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 1 [ 63]: Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK1022d9c4;rport [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 2 [ 62]: From: "YEHAVI BOURVIN" ;tag=as668cf442 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 3 [ 73]: To: ;tag=1268759603 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 4 [ 54]: Call-ID: 7921261c58176f0d7c530e722904adff@132.64.9.163 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 5 [ 16]: CSeq: 103 INVITE [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 6 [ 75]: Contact: Yehavi 7912 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 7 [ 34]: Server: Cisco-CP7912/8.0.1-060412A [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 8 [ 80]: Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 9 [ 19]: Supported: replaces [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 10 [ 19]: Content-Length: 204 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 11 [ 29]: Content-Type: application/sdp [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 12 [ 0]: [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 0 [ 3]: v=0 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 1 [ 39]: o=80604 52605 52605 IN IP4 132.64.4.128 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 2 [ 21]: s=Cisco 7912 SIP Call [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 3 [ 21]: c=IN IP4 132.64.4.128 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 4 [ 5]: t=0 0 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 5 [ 27]: m=audio 16384 RTP/AVP 8 101 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 6 [ 22]: a=rtpmap:8 PCMA/8000/1 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 7 [ 33]: a=rtpmap:101 telephone-event/8000 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 8 [ 15]: a=fmtp:101 0-15 --- (12 headers 9 lines) --- [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:4727 find_call: = Found Their Call ID: 7921261c58176f0d7c530e722904adff@132.64.9.163 Their Tag 1268759603 Our tag: as668cf442 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:2218 __sip_ack: Acked pending invite 103 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:2223 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #9287 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:2234 __sip_ack: Stopping retransmission on '7921261c58176f0d7c530e722904adff@132.64.9.163' of Request 103: Match Found [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:12285 handle_response_invite: SIP response 200 to RE-invite on outgoing call 7921261c58176f0d7c530e722904adff@132.64.9.163 Found RTP audio format 8 Found RTP audio format 101 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:5282 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 132.64.4.128:16384 Found description format PCMA for ID 8 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:5524 process_sdp: T38 state changed to 0 on channel SIP/80604-096128e8 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=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 132.64.4.128:16384 [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:5615 process_sdp: We're settling with these formats: 0x8 (alaw) [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:5622 process_sdp: We have an owner, now see if we need to change this call [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:3305 update_call_counter: Updating call counter for outgoing call --- set_address_from_contact host '132.64.4.128' [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:8417 build_route: build_route: Retaining previous route: [Feb 27 07:30:40] DEBUG[19189]: chan_sip.c:6050 reqprep: Strict routing enforced for session 7921261c58176f0d7c530e722904adff@132.64.9.163 set_destination: Parsing for address/port to send to set_destination: set destination to 132.64.4.128, port 5060 Transmitting (no NAT) to 132.64.4.128:5060: ACK sip:80604@132.64.4.128:5060;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK5306ec86;rport Max-Forwards: 70 From: "YEHAVI BOURVIN" ;tag=as668cf442 To: ;tag=1268759603 Contact: Call-ID: 7921261c58176f0d7c530e722904adff@132.64.9.163 CSeq: 103 ACK User-Agent: Asterisk PBX SVN-trunk-r56126M Content-Length: 0 --- pbx-gr*CLI> pbx-gr*CLI> pbx-gr*CLI> ~~~~~~ Here comes the HOLD press pbx-gr*CLI> No such command '~~~~~~' (type 'help' for help) pbx-gr*CLI> pbx-gr*CLI> <--- SIP read from 132.64.4.128:5060 ---> INVITE sip:89444@132.64.9.163 SIP/2.0 Via: SIP/2.0/UDP 132.64.4.128:5060;branch=z9hG4bK26c121d439f9c84a From: ;tag=1268759603 To: "YEHAVI BOURVIN" ;tag=as668cf442 Call-ID: 7921261c58176f0d7c530e722904adff@132.64.9.163 CSeq: 1 INVITE Max-Forwards: 70 Contact: Yehavi 7912 User-Agent: Cisco-CP7912/8.0.1-060412A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces, 100rel Expires: 10 Content-Length: 211 Content-Type: application/sdp v=0 o=80604 53165 53165 IN IP4 132.64.4.128 s=Cisco 7912 SIP Call c=IN IP4 0.0.0.0 t=0 0 m=audio 16384 RTP/AVP 8 101 a=sendonly a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 0 [ 37]: INVITE sip:89444@132.64.9.163 SIP/2.0 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 1 [ 65]: Via: SIP/2.0/UDP 132.64.4.128:5060;branch=z9hG4bK26c121d439f9c84a [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 2 [ 70]: From: ;tag=1268759603 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 3 [ 60]: To: "YEHAVI BOURVIN" ;tag=as668cf442 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 4 [ 54]: Call-ID: 7921261c58176f0d7c530e722904adff@132.64.9.163 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 5 [ 14]: CSeq: 1 INVITE [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 6 [ 16]: Max-Forwards: 70 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 7 [ 75]: Contact: Yehavi 7912 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 8 [ 38]: User-Agent: Cisco-CP7912/8.0.1-060412A [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 9 [ 80]: Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 10 [ 27]: Supported: replaces, 100rel [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 11 [ 11]: Expires: 10 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 12 [ 19]: Content-Length: 211 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 13 [ 29]: Content-Type: application/sdp [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 14 [ 0]: [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 0 [ 3]: v=0 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 1 [ 39]: o=80604 53165 53165 IN IP4 132.64.4.128 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 2 [ 21]: s=Cisco 7912 SIP Call [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 3 [ 16]: c=IN IP4 0.0.0.0 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 4 [ 5]: t=0 0 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 5 [ 27]: m=audio 16384 RTP/AVP 8 101 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 6 [ 10]: a=sendonly [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 7 [ 22]: a=rtpmap:8 PCMA/8000/1 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 9 [ 15]: a=fmtp:101 0-15 --- (14 headers 10 lines) --- [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4727 find_call: = Found Their Call ID: 7921261c58176f0d7c530e722904adff@132.64.9.163 Their Tag 1268759603 Our tag: as668cf442 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:15271 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:1829 parse_sip_options: Begin: parsing SIP "Supported: replaces, 100rel" [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:1837 parse_sip_options: Found SIP option: -replaces- [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:1843 parse_sip_options: Matched SIP option: replaces [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:1837 parse_sip_options: Found SIP option: -100rel- [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:1843 parse_sip_options: Matched SIP option: 100rel Sending to 132.64.4.128 : 5060 (no NAT) [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:13975 handle_request_invite: Initializing initreq for method INVITE - callid 7921261c58176f0d7c530e722904adff@132.64.9.163 Found RTP audio format 8 Found RTP audio format 101 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:5282 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 0.0.0.0:16384 Found description format PCMA for ID 8 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:5524 process_sdp: T38 state changed to 0 on channel SIP/80604-096128e8 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=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 0.0.0.0:16384 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:5615 process_sdp: We're settling with these formats: 0x8 (alaw) [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:5622 process_sdp: We have an owner, now see if we need to change this call pbx-gr*CLI> [Feb 27 07:30:46] DEBUG[19189]: devicestate.c:444 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/80604 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:14109 handle_request_invite: Got a SIP re-invite for call 7921261c58176f0d7c530e722904adff@132.64.9.163 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:14204 handle_request_invite: SIP/80604-096128e8: This call is UP.... [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:6952 transmit_response_with_sdp: Setting framing from config on incoming call [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:6674 add_sdp: ** Our capability: 0x8 (alaw) Video flag: True Text flag: True [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:6676 add_sdp: ** Our prefcodec: 0x8 (alaw) Audio is at 132.64.9.163 port 64524 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:6837 add_sdp: -- Done with adding codecs to SDP [Feb 27 07:30:46] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:6897 add_sdp: Done building SDP. Settling with this capability: 0x8 (alaw) <--- Reliably Transmitting (no NAT) to 132.64.4.128:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 132.64.4.128:5060;branch=z9hG4bK26c121d439f9c84a;received=132.64.4.128 From: ;tag=1268759603 To: "YEHAVI BOURVIN" ;tag=as668cf442 Call-ID: 7921261c58176f0d7c530e722904adff@132.64.9.163 CSeq: 1 INVITE User-Agent: Asterisk PBX SVN-trunk-r56126M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 240 v=0 o=root 19178 19180 IN IP4 128.139.26.6 s=session c=IN IP4 128.139.26.6 t=0 0 m=audio 19256 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=recvonly <------------> [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:2115 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #9289 pbx-gr*CLI> [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:17627 sip_set_rtp_peer: Sending reinvite on SIP 'D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6' - It's audio soon redirected to IP 132.64.9.163 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:6050 reqprep: Strict routing enforced for session D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6 set_destination: Parsing for address/port to send to set_destination: set destination to 128.139.26.6, port 5060 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:6674 add_sdp: ** Our capability: 0x8 (alaw) Video flag: True Text flag: True [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:6676 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 132.64.9.163 port 27044 Adding codec 0x8 (alaw) to SDP [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:6837 add_sdp: -- Done with adding codecs to SDP [Feb 27 07:30:46] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:6897 add_sdp: Done building SDP. Settling with this capability: 0x8 (alaw) [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:1716 initialize_initreq: Initializing already initialized SIP dialog D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6 (presumably reinvite) [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 0 [ 42]: INVITE sip:89444@128.139.26.6:5060 SIP/2.0 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 1 [ 63]: Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK0b410140;rport [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 2 [ 16]: Max-Forwards: 70 pbx-gr*CLI> [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 3 [ 45]: From: ;tag=as268edf01 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 4 [ 63]: To: "YEHAVI BOURVIN" ;tag=270AED48-1804 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 5 [ 33]: Contact: [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 6 [ 56]: Call-ID: D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 7 [ 16]: CSeq: 103 INVITE [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 8 [ 42]: User-Agent: Asterisk PBX SVN-trunk-r56126M [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 9 [ 66]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 10 [ 19]: Supported: replaces [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 12 [ 29]: Content-Type: application/sdp [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 13 [ 19]: Content-Length: 184 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 14 [ 0]: [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 0 [ 3]: v=0 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 1 [ 38]: o=root 19178 19180 IN IP4 132.64.9.163 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 2 [ 9]: s=session [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 3 [ 21]: c=IN IP4 132.64.9.163 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 4 [ 5]: t=0 0 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 5 [ 23]: m=audio 27044 RTP/AVP 8 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 7 [ 25]: a=silenceSupp:off - - - - [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 8 [ 10]: a=ptime:20 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 9 [ 10]: a=sendrecv Reliably Transmitting (no NAT) to 128.139.26.6:5060: INVITE sip:89444@128.139.26.6:5060 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK0b410140;rport Max-Forwards: 70 From: ;tag=as268edf01 To: "YEHAVI BOURVIN" ;tag=270AED48-1804 Contact: Call-ID: D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6 CSeq: 103 INVITE User-Agent: Asterisk PBX SVN-trunk-r56126M 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: 184 v=0 o=root 19178 19180 IN IP4 132.64.9.163 s=session c=IN IP4 132.64.9.163 t=0 0 m=audio 27044 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:2115 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #9290 -- Started music on hold, class 'default', on SIP/gr-pbx-link-0960d288 [Feb 27 07:30:46] DEBUG[19189]: rtp.c:2937 bridge_native_loop: Oooh, 'SIP/80604-096128e8' changed end address to 0.0.0.0:0 (format 8) [Feb 27 07:30:46] DEBUG[19189]: rtp.c:2939 bridge_native_loop: Oooh, 'SIP/80604-096128e8' changed end vaddress to 0.0.0.0:0 (format 8) [Feb 27 07:30:46] DEBUG[19189]: rtp.c:2941 bridge_native_loop: Oooh, 'SIP/80604-096128e8' changed end taddress to 0.0.0.0:0 (format 8) [Feb 27 07:30:46] DEBUG[19189]: rtp.c:2943 bridge_native_loop: Oooh, 'SIP/80604-096128e8' was 132.64.4.128:16384/(format 8) [Feb 27 07:30:46] DEBUG[19189]: rtp.c:2945 bridge_native_loop: Oooh, 'SIP/80604-096128e8' was 0.0.0.0:0/(format 8) [Feb 27 07:30:46] DEBUG[19189]: rtp.c:2947 bridge_native_loop: Oooh, 'SIP/80604-096128e8' was 0.0.0.0:0/(format 8) [Feb 27 07:30:46] DEBUG[19189]: devicestate.c:302 ast_device_state: No provider found, checking channel drivers for SIP - 80604 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:15897 sip_devicestate: Checking device state for peer 80604 [Feb 27 07:30:46] DEBUG[19189]: devicestate.c:428 do_state_change: Changing state for SIP/80604 - state 8 (On Hold) [Feb 27 07:30:46] DEBUG[19189]: devicestate.c:302 ast_device_state: No provider found, checking channel drivers for SIP - 80604 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:15897 sip_devicestate: Checking device state for peer 80604 pbx-gr*CLI> [Feb 27 07:30:46] DEBUG[19189]: app_queue.c:568 changethread: Device 'SIP/80604' changed to state '8' (On Hold) but we don't care because they're not a member of any queue. pbx-gr*CLI> <--- SIP read from 132.64.4.128:5060 ---> ACK sip:89444@132.64.9.163 SIP/2.0 Via: SIP/2.0/UDP 132.64.4.128:5060;branch=z9hG4bK2eb40a615f4fee99 From: ;tag=1268759603 To: "YEHAVI BOURVIN" ;tag=as668cf442 Call-ID: 7921261c58176f0d7c530e722904adff@132.64.9.163 CSeq: 1 ACK Max-Forwards: 70 User-Agent: Cisco-CP7912/8.0.1-060412A Content-Length: 0 <-------------> [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 0 [ 34]: ACK sip:89444@132.64.9.163 SIP/2.0 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 1 [ 65]: Via: SIP/2.0/UDP 132.64.4.128:5060;branch=z9hG4bK2eb40a615f4fee99 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 2 [ 70]: From: ;tag=1268759603 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 3 [ 60]: To: "YEHAVI BOURVIN" ;tag=as668cf442 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 4 [ 54]: Call-ID: 7921261c58176f0d7c530e722904adff@132.64.9.163 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 5 [ 11]: CSeq: 1 ACK [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 6 [ 16]: Max-Forwards: 70 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 7 [ 38]: User-Agent: Cisco-CP7912/8.0.1-060412A [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 8 [ 17]: Content-Length: 0 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 9 [ 0]: --- (9 headers 0 lines) --- [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4727 find_call: = Found Their Call ID: 7921261c58176f0d7c530e722904adff@132.64.9.163 Their Tag 1268759603 Our tag: as668cf442 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:15271 handle_request: **** Received ACK (6) - Command in SIP ACK [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:2223 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #9289 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:2234 __sip_ack: Stopping retransmission on '7921261c58176f0d7c530e722904adff@132.64.9.163' of Response 1: Match Found pbx-gr*CLI> <--- SIP read from 128.139.26.6:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK0b410140;rport From: ;tag=as268edf01 To: "YEHAVI BOURVIN" ;tag=270AED48-1804 Date: Tue, 27 Feb 2007 05:34:48 GMT Call-ID: D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6 Server: Cisco-SIPGateway/IOS-12.x CSeq: 103 INVITE Allow-Events: telephone-event Remote-Party-ID: "YEHAVI BOURVIN" ;party=called;screen=no;privacy=off Content-Length: 0 <-------------> [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 0 [ 18]: SIP/2.0 100 Trying [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 1 [ 63]: Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK0b410140;rport [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 2 [ 45]: From: ;tag=as268edf01 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 3 [ 63]: To: "YEHAVI BOURVIN" ;tag=270AED48-1804 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 4 [ 35]: Date: Tue, 27 Feb 2007 05:34:48 GMT [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 5 [ 56]: Call-ID: D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 6 [ 33]: Server: Cisco-SIPGateway/IOS-12.x [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 7 [ 16]: CSeq: 103 INVITE [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 8 [ 29]: Allow-Events: telephone-event [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 9 [ 93]: Remote-Party-ID: "YEHAVI BOURVIN" ;party=called;screen=no;privacy=off [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 10 [ 17]: Content-Length: 0 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 11 [ 0]: --- (11 headers 0 lines) --- [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4727 find_call: = No match Their Call ID: 7921261c58176f0d7c530e722904adff@132.64.9.163 Their Tag 1268759603 Our tag: as668cf442 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4727 find_call: = Found Their Call ID: D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6 Their Tag 270AED48-1804 Our tag: as268edf01 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:2268 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #9290 - INVITE (got response) [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:2277 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on 'D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6' Request 103: Found [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:12285 handle_response_invite: SIP response 100 to RE-invite on outgoing call D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6 <--- SIP read from 128.139.26.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK0b410140;rport From: ;tag=as268edf01 To: "YEHAVI BOURVIN" ;tag=270AED48-1804 Date: Tue, 27 Feb 2007 05:34:48 GMT Call-ID: D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6 Server: Cisco-SIPGateway/IOS-12.x CSeq: 103 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER Allow-Events: telephone-event Remote-Party-ID: "YEHAVI BOURVIN" ;party=called;screen=no;privacy=off Contact: Supported: replaces Content-Type: application/sdp Content-Length: 218 v=0 o=CiscoSystemsSIP-GW-UserAgent 8474 8951 IN IP4 128.139.26.6 s=SIP Call c=IN IP4 128.139.26.6 t=0 0 m=audio 19256 RTP/AVP 8 c=IN IP4 128.139.26.6 a=rtpmap:8 PCMA/8000 a=ptime:20 a=silenceSupp:off - - - - <-------------> [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 0 [ 14]: SIP/2.0 200 OK [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 1 [ 63]: Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK0b410140;rport pbx-gr*CLI> [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 2 [ 45]: From: ;tag=as268edf01 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 3 [ 63]: To: "YEHAVI BOURVIN" ;tag=270AED48-1804 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 4 [ 35]: Date: Tue, 27 Feb 2007 05:34:48 GMT [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 5 [ 56]: Call-ID: D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 6 [ 33]: Server: Cisco-SIPGateway/IOS-12.x [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 7 [ 16]: CSeq: 103 INVITE [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 8 [ 97]: Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 9 [ 29]: Allow-Events: telephone-event [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 10 [ 93]: Remote-Party-ID: "YEHAVI BOURVIN" ;party=called;screen=no;privacy=off [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 11 [ 38]: Contact: [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 12 [ 19]: Supported: replaces [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 13 [ 29]: Content-Type: application/sdp [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 14 [ 19]: Content-Length: 218 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 15 [ 0]: [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 0 [ 3]: v=0 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 1 [ 60]: o=CiscoSystemsSIP-GW-UserAgent 8474 8951 IN IP4 128.139.26.6 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 2 [ 10]: s=SIP Call [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 3 [ 21]: c=IN IP4 128.139.26.6 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 4 [ 5]: t=0 0 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 5 [ 23]: m=audio 19256 RTP/AVP 8 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 6 [ 21]: c=IN IP4 128.139.26.6 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 8 [ 10]: a=ptime:20 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 9 [ 25]: a=silenceSupp:off - - - - --- (15 headers 10 lines) --- [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4727 find_call: = No match Their Call ID: 7921261c58176f0d7c530e722904adff@132.64.9.163 Their Tag 1268759603 Our tag: as668cf442 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:4727 find_call: = Found Their Call ID: D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6 Their Tag 270AED48-1804 Our tag: as268edf01 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:2218 __sip_ack: Acked pending invite 103 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:2234 __sip_ack: Stopping retransmission on 'D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6' of Request 103: Match Found [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:12285 handle_response_invite: SIP response 200 to RE-invite on outgoing call D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6 Found RTP audio format 8 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:5282 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 128.139.26.6:19256 Found description format PCMA for ID 8 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:5524 process_sdp: T38 state changed to 0 on channel SIP/gr-pbx-link-0960d288 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 128.139.26.6:19256 [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:5615 process_sdp: We're settling with these formats: 0x8 (alaw) [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:5622 process_sdp: We have an owner, now see if we need to change this call [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:3305 update_call_counter: Updating call counter for incoming call --- set_address_from_contact host '128.139.26.6' [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:8417 build_route: build_route: Retaining previous route: [Feb 27 07:30:46] DEBUG[19189]: chan_sip.c:6050 reqprep: Strict routing enforced for session D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6 set_destination: Parsing for address/port to send to set_destination: set destination to 128.139.26.6, port 5060 Transmitting (no NAT) to 128.139.26.6:5060: ACK sip:89444@128.139.26.6:5060 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK0a7f57db;rport Max-Forwards: 70 From: ;tag=as268edf01 To: "YEHAVI BOURVIN" ;tag=270AED48-1804 Contact: Call-ID: D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6 CSeq: 103 ACK User-Agent: Asterisk PBX SVN-trunk-r56126M Content-Length: 0 --- pbx-gr*CLI> [Feb 27 07:30:46] DEBUG[19189]: rtp.c:1241 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Feb 27 07:30:46] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) [Feb 27 07:30:46] DEBUG[19189]: channel.c:2760 set_format: Set channel SIP/gr-pbx-link-0960d288 to write format slin [Feb 27 07:30:46] DEBUG[19189]: res_musiconhold.c:252 ast_moh_files_next: SIP/gr-pbx-link-0960d288 Opened file 1 '/var/lib/asterisk/moh/fpm-world-mix' [Feb 27 07:30:46] DEBUG[19189]: rtp.c:2645 ast_rtp_raw_write: Difference is 40848, ms is 5126 pbx-gr*CLI> [Feb 27 07:30:46] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:46] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:46] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:46] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:46] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:46] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:46] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:46] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:46] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:46] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:46] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:46] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:46] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:46] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:46] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:46] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:46] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:46] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:46] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:46] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:46] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:46] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:46] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:46] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:46] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:46] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:46] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:46] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:46] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:46] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:46] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:46] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:46] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:47] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:47] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:47] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:47] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:47] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:47] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:47] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:47] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:47] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:47] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:47] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:47] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:47] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:47] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:47] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:47] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:47] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:47] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:47] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:47] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:47] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:47] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:47] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:47] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:47] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:47] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:47] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:47] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:47] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:47] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:47] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:47] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:47] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:47] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:47] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:47] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:47] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:47] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:47] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:47] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:47] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:47] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:47] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:47] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:47] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:47] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:47] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:47] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:47] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:47] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:48] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:48] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:48] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:48] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:48] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:48] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> [Feb 27 07:30:48] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) pbx-gr*CLI> <--- SIP read from 132.64.4.128:5060 ---> INVITE sip:89444@132.64.9.163 SIP/2.0 Via: SIP/2.0/UDP 132.64.4.128:5060;branch=z9hG4bK77c6d98f44e913fa From: ;tag=1268759603 To: "YEHAVI BOURVIN" ;tag=as668cf442 Call-ID: 7921261c58176f0d7c530e722904adff@132.64.9.163 CSeq: 2 INVITE Max-Forwards: 70 Contact: Yehavi 7912 User-Agent: Cisco-CP7912/8.0.1-060412A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces, 100rel Expires: 10 Content-Length: 279 Content-Type: application/sdp v=0 o=80604 53350 53350 IN IP4 132.64.4.128 s=Cisco 7912 SIP Call c=IN IP4 132.64.4.128 t=0 0 m=audio 16384 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:18 G729/8000/1 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 0 [ 37]: INVITE sip:89444@132.64.9.163 SIP/2.0 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 1 [ 65]: Via: SIP/2.0/UDP 132.64.4.128:5060;branch=z9hG4bK77c6d98f44e913fa [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 2 [ 70]: From: ;tag=1268759603 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 3 [ 60]: To: "YEHAVI BOURVIN" ;tag=as668cf442 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 4 [ 54]: Call-ID: 7921261c58176f0d7c530e722904adff@132.64.9.163 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 5 [ 14]: CSeq: 2 INVITE [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 6 [ 16]: Max-Forwards: 70 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 7 [ 75]: Contact: Yehavi 7912 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 8 [ 38]: User-Agent: Cisco-CP7912/8.0.1-060412A [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 9 [ 80]: Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 10 [ 27]: Supported: replaces, 100rel [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 11 [ 11]: Expires: 10 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 12 [ 19]: Content-Length: 279 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 13 [ 29]: Content-Type: application/sdp [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 14 [ 0]: [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 0 [ 3]: v=0 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 1 [ 39]: o=80604 53350 53350 IN IP4 132.64.4.128 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 2 [ 21]: s=Cisco 7912 SIP Call [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 3 [ 21]: c=IN IP4 132.64.4.128 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 4 [ 5]: t=0 0 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 5 [ 32]: m=audio 16384 RTP/AVP 8 0 18 101 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 6 [ 22]: a=rtpmap:8 PCMA/8000/1 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 7 [ 22]: a=rtpmap:0 PCMU/8000/1 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 8 [ 23]: a=rtpmap:18 G729/8000/1 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 9 [ 19]: a=fmtp:18 annexb=no [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 10 [ 33]: a=rtpmap:101 telephone-event/8000 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 11 [ 15]: a=fmtp:101 0-15 --- (14 headers 12 lines) --- [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4727 find_call: = Found Their Call ID: 7921261c58176f0d7c530e722904adff@132.64.9.163 Their Tag 1268759603 Our tag: as668cf442 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:15271 handle_request: **** Received INVITE (5) - Command in SIP INVITE Sending to 132.64.4.128 : 5060 (no NAT) [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:13975 handle_request_invite: Initializing initreq for method INVITE - callid 7921261c58176f0d7c530e722904adff@132.64.9.163 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 101 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:5282 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 132.64.4.128:16384 Found description format PCMA for ID 8 Found description format PCMU for ID 0 Found description format G729 for ID 18 Got unsupported a:fmtp in SDP offer Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:5524 process_sdp: T38 state changed to 0 on channel SIP/80604-096128e8 Capabilities: us - 0x8 (alaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=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 132.64.4.128:16384 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:5615 process_sdp: We're settling with these formats: 0x8 (alaw) [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:5622 process_sdp: We have an owner, now see if we need to change this call [Feb 27 07:30:48] DEBUG[19189]: devicestate.c:444 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/80604 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:14109 handle_request_invite: Got a SIP re-invite for call 7921261c58176f0d7c530e722904adff@132.64.9.163 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:14204 handle_request_invite: SIP/80604-096128e8: This call is UP.... [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:6952 transmit_response_with_sdp: Setting framing from config on incoming call [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:6674 add_sdp: ** Our capability: 0x8 (alaw) Video flag: True Text flag: True [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:6676 add_sdp: ** Our prefcodec: 0x8 (alaw) Audio is at 132.64.9.163 port 64524 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:6837 add_sdp: -- Done with adding codecs to SDP [Feb 27 07:30:48] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:6897 add_sdp: Done building SDP. Settling with this capability: 0x8 (alaw) <--- Reliably Transmitting (no NAT) to 132.64.4.128:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 132.64.4.128:5060;branch=z9hG4bK77c6d98f44e913fa;received=132.64.4.128 From: ;tag=1268759603 To: "YEHAVI BOURVIN" ;tag=as668cf442 Call-ID: 7921261c58176f0d7c530e722904adff@132.64.9.163 CSeq: 2 INVITE User-Agent: Asterisk PBX SVN-trunk-r56126M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 240 v=0 o=root 19178 19181 IN IP4 128.139.26.6 s=session c=IN IP4 128.139.26.6 t=0 0 m=audio 19256 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:2115 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #9291 pbx-gr*CLI> [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:17627 sip_set_rtp_peer: Sending reinvite on SIP 'D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6' - It's audio soon redirected to IP 132.64.4.128 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:6050 reqprep: Strict routing enforced for session D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6 set_destination: Parsing for address/port to send to set_destination: set destination to 128.139.26.6, port 5060 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:6674 add_sdp: ** Our capability: 0x8 (alaw) Video flag: True Text flag: True [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:6676 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 132.64.9.163 port 27044 Adding codec 0x8 (alaw) to SDP [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:6837 add_sdp: -- Done with adding codecs to SDP [Feb 27 07:30:48] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:6897 add_sdp: Done building SDP. Settling with this capability: 0x8 (alaw) [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:1716 initialize_initreq: Initializing already initialized SIP dialog D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6 (presumably reinvite) [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 0 [ 42]: INVITE sip:89444@128.139.26.6:5060 SIP/2.0 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 1 [ 63]: Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK32ca10ba;rport [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 2 [ 16]: Max-Forwards: 70 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 3 [ 45]: From: ;tag=as268edf01 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 4 [ 63]: To: "YEHAVI BOURVIN" ;tag=270AED48-1804 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 5 [ 33]: Contact: [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 6 [ 56]: Call-ID: D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 7 [ 16]: CSeq: 104 INVITE [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 8 [ 42]: User-Agent: Asterisk PBX SVN-trunk-r56126M [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 9 [ 66]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY pbx-gr*CLI> [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 10 [ 19]: Supported: replaces [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 12 [ 29]: Content-Type: application/sdp [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 13 [ 19]: Content-Length: 184 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 14 [ 0]: [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 0 [ 3]: v=0 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 1 [ 38]: o=root 19178 19181 IN IP4 132.64.4.128 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 2 [ 9]: s=session [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 3 [ 21]: c=IN IP4 132.64.4.128 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 4 [ 5]: t=0 0 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 5 [ 23]: m=audio 16384 RTP/AVP 8 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 7 [ 25]: a=silenceSupp:off - - - - [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 8 [ 10]: a=ptime:20 pbx-gr*CLI> [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 9 [ 10]: a=sendrecv Reliably Transmitting (no NAT) to 128.139.26.6:5060: INVITE sip:89444@128.139.26.6:5060 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK32ca10ba;rport Max-Forwards: 70 From: ;tag=as268edf01 To: "YEHAVI BOURVIN" ;tag=270AED48-1804 Contact: Call-ID: D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6 CSeq: 104 INVITE User-Agent: Asterisk PBX SVN-trunk-r56126M 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: 184 v=0 o=root 19178 19181 IN IP4 132.64.4.128 s=session c=IN IP4 132.64.4.128 t=0 0 m=audio 16384 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:2115 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #9292 -- Stopped music on hold on SIP/gr-pbx-link-0960d288 [Feb 27 07:30:48] DEBUG[19189]: channel.c:2760 set_format: Set channel SIP/gr-pbx-link-0960d288 to write format alaw [Feb 27 07:30:48] DEBUG[19189]: rtp.c:2937 bridge_native_loop: Oooh, 'SIP/80604-096128e8' changed end address to 132.64.4.128:16384 (format 268) [Feb 27 07:30:48] DEBUG[19189]: rtp.c:2939 bridge_native_loop: Oooh, 'SIP/80604-096128e8' changed end vaddress to 0.0.0.0:0 (format 268) [Feb 27 07:30:48] DEBUG[19189]: rtp.c:2941 bridge_native_loop: Oooh, 'SIP/80604-096128e8' changed end taddress to 0.0.0.0:0 (format 268) [Feb 27 07:30:48] DEBUG[19189]: rtp.c:2943 bridge_native_loop: Oooh, 'SIP/80604-096128e8' was 0.0.0.0:0/(format 8) [Feb 27 07:30:48] DEBUG[19189]: rtp.c:2945 bridge_native_loop: Oooh, 'SIP/80604-096128e8' was 0.0.0.0:0/(format 8) [Feb 27 07:30:48] DEBUG[19189]: rtp.c:2947 bridge_native_loop: Oooh, 'SIP/80604-096128e8' was 0.0.0.0:0/(format 8) [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:17632 sip_set_rtp_peer: Deferring reinvite on SIP 'D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6' - It's audio will be redirected to IP 132.64.4.128 [Feb 27 07:30:48] DEBUG[19189]: devicestate.c:302 ast_device_state: No provider found, checking channel drivers for SIP - 80604 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:15897 sip_devicestate: Checking device state for peer 80604 [Feb 27 07:30:48] DEBUG[19189]: devicestate.c:428 do_state_change: Changing state for SIP/80604 - state 8 (On Hold) [Feb 27 07:30:48] DEBUG[19189]: devicestate.c:302 ast_device_state: No provider found, checking channel drivers for SIP - 80604 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:15897 sip_devicestate: Checking device state for peer 80604 [Feb 27 07:30:48] DEBUG[19189]: rtp.c:2809 ast_rtp_write: Created smoother: format: 8 ms: 20 len: 160 [Feb 27 07:30:48] DEBUG[19189]: app_queue.c:568 changethread: Device 'SIP/80604' changed to state '8' (On Hold) but we don't care because they're not a member of any queue. pbx-gr*CLI> [Feb 27 07:30:48] DEBUG[19189]: rtp.c:941 ast_rtcp_read: Got RTCP report of 72 bytes pbx-gr*CLI> <--- SIP read from 128.139.26.6:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK32ca10ba;rport From: ;tag=as268edf01 To: "YEHAVI BOURVIN" ;tag=270AED48-1804 Date: Tue, 27 Feb 2007 05:34:50 GMT Call-ID: D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6 Server: Cisco-SIPGateway/IOS-12.x CSeq: 104 INVITE Allow-Events: telephone-event Remote-Party-ID: "YEHAVI BOURVIN" ;party=called;screen=no;privacy=off Content-Length: 0 <-------------> [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 0 [ 18]: SIP/2.0 100 Trying [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 1 [ 63]: Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK32ca10ba;rport [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 2 [ 45]: From: ;tag=as268edf01 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 3 [ 63]: To: "YEHAVI BOURVIN" ;tag=270AED48-1804 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 4 [ 35]: Date: Tue, 27 Feb 2007 05:34:50 GMT [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 5 [ 56]: Call-ID: D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 6 [ 33]: Server: Cisco-SIPGateway/IOS-12.x [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 7 [ 16]: CSeq: 104 INVITE [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 8 [ 29]: Allow-Events: telephone-event [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 9 [ 93]: Remote-Party-ID: "YEHAVI BOURVIN" ;party=called;screen=no;privacy=off [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 10 [ 17]: Content-Length: 0 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 11 [ 0]: --- (11 headers 0 lines) --- [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4727 find_call: = No match Their Call ID: 7921261c58176f0d7c530e722904adff@132.64.9.163 Their Tag 1268759603 Our tag: as668cf442 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4727 find_call: = Found Their Call ID: D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6 Their Tag 270AED48-1804 Our tag: as268edf01 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:2268 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #9292 - INVITE (got response) [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:2277 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on 'D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6' Request 104: Found [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:12285 handle_response_invite: SIP response 100 to RE-invite on outgoing call D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:12268 check_pendings: Sending pending reinvite on 'D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6' [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:6050 reqprep: Strict routing enforced for session D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6 set_destination: Parsing for address/port to send to set_destination: set destination to 128.139.26.6, port 5060 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:6674 add_sdp: ** Our capability: 0x8 (alaw) Video flag: True Text flag: True [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:6676 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 132.64.9.163 port 27044 Adding codec 0x8 (alaw) to SDP [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:6837 add_sdp: -- Done with adding codecs to SDP [Feb 27 07:30:48] DEBUG[19189]: channel.c:2294 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:6897 add_sdp: Done building SDP. Settling with this capability: 0x8 (alaw) [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:1716 initialize_initreq: Initializing already initialized SIP dialog D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6 (presumably reinvite) [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 0 [ 42]: INVITE sip:89444@128.139.26.6:5060 SIP/2.0 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 1 [ 63]: Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK19a23375;rport [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 2 [ 16]: Max-Forwards: 70 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 3 [ 45]: From: ;tag=as268edf01 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 4 [ 63]: To: "YEHAVI BOURVIN" ;tag=270AED48-1804 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 5 [ 33]: Contact: [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 6 [ 56]: Call-ID: D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 7 [ 16]: CSeq: 105 INVITE [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 8 [ 42]: User-Agent: Asterisk PBX SVN-trunk-r56126M [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 9 [ 66]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 10 [ 19]: Supported: replaces [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 12 [ 29]: Content-Type: application/sdp [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 13 [ 19]: Content-Length: 184 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 14 [ 0]: [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 0 [ 3]: v=0 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 1 [ 38]: o=root 19178 19182 IN IP4 132.64.4.128 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 2 [ 9]: s=session [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 3 [ 21]: c=IN IP4 132.64.4.128 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 4 [ 5]: t=0 0 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 5 [ 23]: m=audio 16384 RTP/AVP 8 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 7 [ 25]: a=silenceSupp:off - - - - [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 8 [ 10]: a=ptime:20 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 9 [ 10]: a=sendrecv Reliably Transmitting (no NAT) to 128.139.26.6:5060: INVITE sip:89444@128.139.26.6:5060 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK19a23375;rport Max-Forwards: 70 From: ;tag=as268edf01 To: "YEHAVI BOURVIN" ;tag=270AED48-1804 Contact: Call-ID: D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6 CSeq: 105 INVITE User-Agent: Asterisk PBX SVN-trunk-r56126M 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: 184 v=0 o=root 19178 19182 IN IP4 132.64.4.128 s=session c=IN IP4 132.64.4.128 t=0 0 m=audio 16384 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:2115 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #9294 <--- SIP read from 128.139.26.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK32ca10ba;rport From: ;tag=as268edf01 To: "YEHAVI BOURVIN" ;tag=270AED48-1804 Date: Tue, 27 Feb 2007 05:34:50 GMT Call-ID: D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6 Server: Cisco-SIPGateway/IOS-12.x CSeq: 104 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER Allow-Events: telephone-event Remote-Party-ID: "YEHAVI BOURVIN" ;party=called;screen=no;privacy=off Contact: Supported: replaces Content-Type: application/sdp Content-Length: 218 v=0 o=CiscoSystemsSIP-GW-UserAgent 8474 8952 IN IP4 128.139.26.6 s=SIP Call c=IN IP4 128.139.26.6 t=0 0 m=audio 19256 RTP/AVP 8 c=IN IP4 128.139.26.6 a=rtpmap:8 PCMA/8000 a=ptime:20 a=silenceSupp:off - - - - <-------------> [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 0 [ 14]: SIP/2.0 200 OK [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 1 [ 63]: Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK32ca10ba;rport [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 2 [ 45]: From: ;tag=as268edf01 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 3 [ 63]: To: "YEHAVI BOURVIN" ;tag=270AED48-1804 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 4 [ 35]: Date: Tue, 27 Feb 2007 05:34:50 GMT [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 5 [ 56]: Call-ID: D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 6 [ 33]: Server: Cisco-SIPGateway/IOS-12.x [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 7 [ 16]: CSeq: 104 INVITE [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 8 [ 97]: Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 9 [ 29]: Allow-Events: telephone-event [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 10 [ 93]: Remote-Party-ID: "YEHAVI BOURVIN" ;party=called;screen=no;privacy=off [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 11 [ 38]: Contact: [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 12 [ 19]: Supported: replaces [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 13 [ 29]: Content-Type: application/sdp [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 14 [ 19]: Content-Length: 218 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 15 [ 0]: [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 0 [ 3]: v=0 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 1 [ 60]: o=CiscoSystemsSIP-GW-UserAgent 8474 8952 IN IP4 128.139.26.6 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 2 [ 10]: s=SIP Call [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 3 [ 21]: c=IN IP4 128.139.26.6 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 4 [ 5]: t=0 0 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 5 [ 23]: m=audio 19256 RTP/AVP 8 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 6 [ 21]: c=IN IP4 128.139.26.6 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 8 [ 10]: a=ptime:20 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 9 [ 25]: a=silenceSupp:off - - - - --- (15 headers 10 lines) --- [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4727 find_call: = No match Their Call ID: 7921261c58176f0d7c530e722904adff@132.64.9.163 Their Tag 1268759603 Our tag: as668cf442 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4727 find_call: = Found Their Call ID: D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6 Their Tag 270AED48-1804 Our tag: as268edf01 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:2234 __sip_ack: Stopping retransmission on 'D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6' of Request 104: Match Found [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:12285 handle_response_invite: SIP response 200 to RE-invite on outgoing call D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6 Found RTP audio format 8 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:5282 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 128.139.26.6:19256 pbx-gr*CLI> Found description format PCMA for ID 8 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:5524 process_sdp: T38 state changed to 0 on channel SIP/gr-pbx-link-0960d288 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 128.139.26.6:19256 pbx-gr*CLI> [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:5615 process_sdp: We're settling with these formats: 0x8 (alaw) [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:5622 process_sdp: We have an owner, now see if we need to change this call [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:3305 update_call_counter: Updating call counter for incoming call --- set_address_from_contact host '128.139.26.6' [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:8417 build_route: build_route: Retaining previous route: [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:6050 reqprep: Strict routing enforced for session D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6 set_destination: Parsing for address/port to send to set_destination: set destination to 128.139.26.6, port 5060 Transmitting (no NAT) to 128.139.26.6:5060: ACK sip:89444@128.139.26.6:5060 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK7328ffa5;rport Max-Forwards: 70 From: ;tag=as268edf01 To: "YEHAVI BOURVIN" ;tag=270AED48-1804 Contact: Call-ID: D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6 CSeq: 104 ACK User-Agent: Asterisk PBX SVN-trunk-r56126M Content-Length: 0 --- <--- SIP read from 132.64.4.128:5060 ---> ACK sip:89444@132.64.9.163 SIP/2.0 Via: SIP/2.0/UDP 132.64.4.128:5060;branch=z9hG4bK513bcd2c428ac18 From: ;tag=1268759603 To: "YEHAVI BOURVIN" ;tag=as668cf442 Call-ID: 7921261c58176f0d7c530e722904adff@132.64.9.163 CSeq: 2 ACK Max-Forwards: 70 User-Agent: Cisco-CP7912/8.0.1-060412A Content-Length: 0 <-------------> [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 0 [ 34]: ACK sip:89444@132.64.9.163 SIP/2.0 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 1 [ 64]: Via: SIP/2.0/UDP 132.64.4.128:5060;branch=z9hG4bK513bcd2c428ac18 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 2 [ 70]: From: ;tag=1268759603 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 3 [ 60]: To: "YEHAVI BOURVIN" ;tag=as668cf442 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 4 [ 54]: Call-ID: 7921261c58176f0d7c530e722904adff@132.64.9.163 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 5 [ 11]: CSeq: 2 ACK [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 6 [ 16]: Max-Forwards: 70 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 7 [ 38]: User-Agent: Cisco-CP7912/8.0.1-060412A [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 8 [ 17]: Content-Length: 0 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 9 [ 0]: --- (9 headers 0 lines) --- [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4727 find_call: = Found Their Call ID: 7921261c58176f0d7c530e722904adff@132.64.9.163 Their Tag 1268759603 Our tag: as668cf442 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:15271 handle_request: **** Received ACK (6) - Command in SIP ACK [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:2223 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #9291 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:2234 __sip_ack: Stopping retransmission on '7921261c58176f0d7c530e722904adff@132.64.9.163' of Response 2: Match Found pbx-gr*CLI> <--- SIP read from 128.139.26.6:5060 ---> SIP/2.0 500 Internal Server Error Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK19a23375;rport From: ;tag=as268edf01 To: "YEHAVI BOURVIN" ;tag=270AED48-1804 Call-ID: D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6 CSeq: 105 INVITE Reason: Q.850;cause=100 Content-Length: 0 <-------------> [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 0 [ 33]: SIP/2.0 500 Internal Server Error [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 1 [ 63]: Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK19a23375;rport [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 2 [ 45]: From: ;tag=as268edf01 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 3 [ 63]: To: "YEHAVI BOURVIN" ;tag=270AED48-1804 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 4 [ 56]: Call-ID: D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 5 [ 16]: CSeq: 105 INVITE [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 6 [ 23]: Reason: Q.850;cause=100 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 7 [ 17]: Content-Length: 0 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 8 [ 0]: --- (8 headers 0 lines) --- [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4727 find_call: = No match Their Call ID: 7921261c58176f0d7c530e722904adff@132.64.9.163 Their Tag 1268759603 Our tag: as668cf442 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4727 find_call: = Found Their Call ID: D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6 Their Tag 270AED48-1804 Our tag: as268edf01 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:2218 __sip_ack: Acked pending invite 105 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:2223 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #9294 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:2234 __sip_ack: Stopping retransmission on 'D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6' of Request 105: Match Found -- Got SIP response 500 "Internal Server Error" back from 128.139.26.6 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:6050 reqprep: Strict routing enforced for session D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6 set_destination: Parsing for address/port to send to set_destination: set destination to 128.139.26.6, port 5060 Transmitting (no NAT) to 128.139.26.6:5060: ACK sip:89444@128.139.26.6:5060 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK7328ffa5;rport Max-Forwards: 70 From: ;tag=as268edf01 To: "YEHAVI BOURVIN" ;tag=270AED48-1804 Contact: Call-ID: D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6 CSeq: 105 ACK User-Agent: Asterisk PBX SVN-trunk-r56126M Content-Length: 0 --- [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:1731 sip_alreadygone: Setting SIP_ALREADYGONE on dialog D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6 pbx-gr*CLI> [Feb 27 07:30:48] DEBUG[19189]: rtp.c:3024 bridge_native_loop: Got a FRAME_CONTROL (8) frame on channel SIP/gr-pbx-link-0960d288 [Feb 27 07:30:48] DEBUG[19189]: channel.c:3971 ast_channel_bridge: Returning from native bridge, channels: SIP/gr-pbx-link-0960d288, SIP/80604-096128e8 [Feb 27 07:30:48] DEBUG[19189]: channel.c:1591 ast_hangup: Hanging up channel 'SIP/80604-096128e8' [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:3603 sip_hangup: Hangup call SIP/80604-096128e8, SIP callid 7921261c58176f0d7c530e722904adff@132.64.9.163) Scheduling destruction of SIP dialog '7921261c58176f0d7c530e722904adff@132.64.9.163' in 32000 ms (Method: ACK) [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:6050 reqprep: Strict routing enforced for session 7921261c58176f0d7c530e722904adff@132.64.9.163 set_destination: Parsing for address/port to send to set_destination: set destination to 132.64.4.128, port 5060 Reliably Transmitting (no NAT) to 132.64.4.128:5060: BYE sip:80604@132.64.4.128:5060;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK1947ad99;rport Max-Forwards: 70 From: "YEHAVI BOURVIN" ;tag=as668cf442 To: ;tag=1268759603 Call-ID: 7921261c58176f0d7c530e722904adff@132.64.9.163 CSeq: 104 BYE User-Agent: Asterisk PBX SVN-trunk-r56126M Content-Length: 0 --- [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:2115 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #9296 [Feb 27 07:30:48] DEBUG[19189]: devicestate.c:444 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/80604-096128e8 [Feb 27 07:30:48] DEBUG[19189]: rtp.c:1566 ast_rtp_early_bridge: Channel '' has no RTP, not doing anything [Feb 27 07:30:48] DEBUG[19189]: app_dial.c:1710 dial_exec_full: Exiting with DIALSTATUS=ANSWER. [Feb 27 07:30:48] DEBUG[19189]: pbx.c:2289 __ast_pbx_run: Spawn extension (huji-remote-gr,80604,6) exited non-zero on 'SIP/gr-pbx-link-0960d288' == Spawn extension (huji-remote-gr, 80604, 6) exited non-zero on 'SIP/gr-pbx-link-0960d288' [Feb 27 07:30:48] DEBUG[19189]: pbx.c:1693 pbx_extension_helper: Launching 'NoOp' -- Executing [h@huji-remote-gr:1] NoOp("SIP/gr-pbx-link-0960d288", "89444 80604 h") in new stack [Feb 27 07:30:48] DEBUG[19189]: db.c:198 ast_db_get: Unable to find key 'CallBack' in family '89444' [Feb 27 07:30:48] DEBUG[19189]: func_db.c:73 function_db_read: DB: 89444/CallBack not found in database. [Feb 27 07:30:48] DEBUG[19189]: pbx.c:1543 pbx_substitute_variables_helper_full: Function result is '' [Feb 27 07:30:48] DEBUG[19189]: pbx.c:1693 pbx_extension_helper: Launching 'Set' -- Executing [h@huji-remote-gr:2] Set("SIP/gr-pbx-link-0960d288", "tmp=") in new stack [Feb 27 07:30:48] DEBUG[19189]: pbx.c:1693 pbx_extension_helper: Launching 'NoOp' -- Executing [h@huji-remote-gr:3] NoOp("SIP/gr-pbx-link-0960d288", "89444 ") in new stack [Feb 27 07:30:48] DEBUG[19189]: pbx.c:1693 pbx_extension_helper: Launching 'GotoIf' -- Executing [h@huji-remote-gr:4] GotoIf("SIP/gr-pbx-link-0960d288", "?5:103") in new stack -- Goto (huji-remote-gr,h,103) [Feb 27 07:30:48] DEBUG[19189]: pbx.c:1693 pbx_extension_helper: Launching 'NoOp' -- Executing [h@huji-remote-gr:103] NoOp("SIP/gr-pbx-link-0960d288", "Nothing to call") in new stack [Feb 27 07:30:48] DEBUG[19189]: pbx.c:1543 pbx_substitute_variables_helper_full: Function result is '"YEHAVI BOURVIN" <89444>' [Feb 27 07:30:48] DEBUG[19189]: pbx.c:1543 pbx_substitute_variables_helper_full: Function result is '89444' [Feb 27 07:30:48] DEBUG[19189]: pbx.c:1543 pbx_substitute_variables_helper_full: Function result is '80604' [Feb 27 07:30:48] DEBUG[19189]: pbx.c:1543 pbx_substitute_variables_helper_full: Function result is 'huji-remote-gr' [Feb 27 07:30:48] DEBUG[19189]: pbx.c:1543 pbx_substitute_variables_helper_full: Function result is 'SIP/gr-pbx-link-0960d288' [Feb 27 07:30:48] DEBUG[19189]: pbx.c:1543 pbx_substitute_variables_helper_full: Function result is 'SIP/80604-096128e8' [Feb 27 07:30:48] DEBUG[19189]: pbx.c:1543 pbx_substitute_variables_helper_full: Function result is 'Dial' [Feb 27 07:30:48] DEBUG[19189]: pbx.c:1543 pbx_substitute_variables_helper_full: Function result is 'SIP/80604|20|' [Feb 27 07:30:48] DEBUG[19189]: pbx.c:1543 pbx_substitute_variables_helper_full: Function result is '2007-02-27 07:30:37' [Feb 27 07:30:48] DEBUG[19189]: pbx.c:1543 pbx_substitute_variables_helper_full: Function result is '2007-02-27 07:30:40' [Feb 27 07:30:48] DEBUG[19189]: pbx.c:1543 pbx_substitute_variables_helper_full: Function result is '2007-02-27 07:30:48' [Feb 27 07:30:48] DEBUG[19189]: pbx.c:1543 pbx_substitute_variables_helper_full: Function result is '11' [Feb 27 07:30:48] DEBUG[19189]: pbx.c:1543 pbx_substitute_variables_helper_full: Function result is '8' [Feb 27 07:30:48] DEBUG[19189]: pbx.c:1543 pbx_substitute_variables_helper_full: Function result is 'ANSWERED' [Feb 27 07:30:48] DEBUG[19189]: pbx.c:1543 pbx_substitute_variables_helper_full: Function result is 'DOCUMENTATION' [Feb 27 07:30:48] DEBUG[19189]: pbx.c:1543 pbx_substitute_variables_helper_full: Function result is '' [Feb 27 07:30:48] DEBUG[19189]: pbx.c:1543 pbx_substitute_variables_helper_full: Function result is '1172554237.181' pbx-gr*CLI> [Feb 27 07:30:48] DEBUG[19189]: pbx.c:1543 pbx_substitute_variables_helper_full: Function result is '' [Feb 27 07:30:48] DEBUG[19189]: channel.c:1591 ast_hangup: Hanging up channel 'SIP/gr-pbx-link-0960d288' [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:3603 sip_hangup: Hangup call SIP/gr-pbx-link-0960d288, SIP callid D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6) [Feb 27 07:30:48] DEBUG[19189]: devicestate.c:444 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/gr-pbx-link-0960d288 [Feb 27 07:30:48] DEBUG[19189]: devicestate.c:302 ast_device_state: No provider found, checking channel drivers for SIP - 80604 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:15897 sip_devicestate: Checking device state for peer 80604 [Feb 27 07:30:48] DEBUG[19189]: devicestate.c:428 do_state_change: Changing state for SIP/80604 - state 8 (On Hold) [Feb 27 07:30:48] DEBUG[19189]: devicestate.c:302 ast_device_state: No provider found, checking channel drivers for SIP - 80604 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:15897 sip_devicestate: Checking device state for peer 80604 [Feb 27 07:30:48] DEBUG[19189]: devicestate.c:302 ast_device_state: No provider found, checking channel drivers for SIP - gr-pbx-link [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:15897 sip_devicestate: Checking device state for peer gr-pbx-link [Feb 27 07:30:48] DEBUG[19189]: app_queue.c:568 changethread: Device 'SIP/80604' changed to state '8' (On Hold) but we don't care because they're not a member of any queue. pbx-gr*CLI> [Feb 27 07:30:48] DEBUG[19189]: res_config_mysql.c:651 mysql_reconnect: MySQL RealTime: Everything is fine. [Feb 27 07:30:48] DEBUG[19189]: res_config_mysql.c:139 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_users WHERE name = 'gr-pbx-link' pbx-gr*CLI> [Feb 27 07:30:48] DEBUG[19189]: devicestate.c:428 do_state_change: Changing state for SIP/gr-pbx-link - state 4 (Invalid) pbx-gr*CLI> [Feb 27 07:30:48] DEBUG[19189]: app_queue.c:568 changethread: Device 'SIP/gr-pbx-link' changed to state '4' (Invalid) but we don't care because they're not a member of any queue. pbx-gr*CLI> <--- SIP read from 132.64.4.128:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK1947ad99;rport From: "YEHAVI BOURVIN" ;tag=as668cf442 To: ;tag=1268759603 Call-ID: 7921261c58176f0d7c530e722904adff@132.64.9.163 CSeq: 104 BYE Server: Cisco-CP7912/8.0.1-060412A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces Content-Length: 0 <-------------> [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 0 [ 14]: SIP/2.0 200 OK [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 1 [ 63]: Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK1947ad99;rport [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 2 [ 62]: From: "YEHAVI BOURVIN" ;tag=as668cf442 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 3 [ 68]: To: ;tag=1268759603 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 4 [ 54]: Call-ID: 7921261c58176f0d7c530e722904adff@132.64.9.163 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 5 [ 13]: CSeq: 104 BYE [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 6 [ 34]: Server: Cisco-CP7912/8.0.1-060412A [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 7 [ 80]: Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 8 [ 19]: Supported: replaces [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 9 [ 17]: Content-Length: 0 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 10 [ 0]: --- (10 headers 0 lines) --- [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:4727 find_call: = Found Their Call ID: 7921261c58176f0d7c530e722904adff@132.64.9.163 Their Tag 1268759603 Our tag: as668cf442 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:2223 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #9296 [Feb 27 07:30:48] DEBUG[19189]: chan_sip.c:2234 __sip_ack: Stopping retransmission on '7921261c58176f0d7c530e722904adff@132.64.9.163' of Request 104: Match Found SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog '7921261c58176f0d7c530e722904adff@132.64.9.163' Method: ACK Really destroying SIP dialog 'D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6' Method: ACK pbx-gr*CLI> <--- SIP read from 132.64.128.105:43590 ---> <-------------> [Feb 27 07:30:49] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 0 [ 0]: [Feb 27 07:30:49] DEBUG[19189]: chan_sip.c:4941 parse_request: Body 0 [ 0]: --- (0 headers 1 lines) --- pbx-gr*CLI> <--- SIP read from 128.139.26.6:50245 ---> BYE sip:80604@132.64.9.163:5060 SIP/2.0 Via: SIP/2.0/UDP 128.139.26.6:5060;branch=z9hG4bK5B02481 From: "YEHAVI BOURVIN" ;tag=270AED48-1804 To: ;tag=as268edf01 Date: Tue, 27 Feb 2007 05:34:50 GMT Call-ID: D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6 User-Agent: Cisco-SIPGateway/IOS-12.x Max-Forwards: 70 Timestamp: 1172554492 CSeq: 102 BYE Content-Length: 0 <-------------> [Feb 27 07:30:50] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 0 [ 39]: BYE sip:80604@132.64.9.163:5060 SIP/2.0 [Feb 27 07:30:50] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 1 [ 56]: Via: SIP/2.0/UDP 128.139.26.6:5060;branch=z9hG4bK5B02481 [Feb 27 07:30:50] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 2 [ 65]: From: "YEHAVI BOURVIN" ;tag=270AED48-1804 [Feb 27 07:30:50] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 3 [ 43]: To: ;tag=as268edf01 [Feb 27 07:30:50] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 4 [ 35]: Date: Tue, 27 Feb 2007 05:34:50 GMT [Feb 27 07:30:50] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 5 [ 56]: Call-ID: D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6 [Feb 27 07:30:50] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 6 [ 37]: User-Agent: Cisco-SIPGateway/IOS-12.x [Feb 27 07:30:50] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 7 [ 16]: Max-Forwards: 70 [Feb 27 07:30:50] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 8 [ 21]: Timestamp: 1172554492 [Feb 27 07:30:50] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 9 [ 13]: CSeq: 102 BYE [Feb 27 07:30:50] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 10 [ 17]: Content-Length: 0 [Feb 27 07:30:50] DEBUG[19189]: chan_sip.c:4941 parse_request: Header 11 [ 0]: --- (11 headers 0 lines) --- [Feb 27 07:30:50] DEBUG[19189]: chan_sip.c:4727 find_call: = No match Their Call ID: 3c26700d7c83-qr0f9ckwnsm1@snom360-0004132384F8 Their Tag mi5g3aesmx Our tag: as27f232fc [Feb 27 07:30:50] DEBUG[19189]: chan_sip.c:4727 find_call: = No match Their Call ID: 3c26700d7c83-m8j3bh85uu3m@snom360-0004132384F8 Their Tag 1ipswbr4bt Our tag: as4af57878 [Feb 27 07:30:50] DEBUG[19189]: chan_sip.c:4727 find_call: = No match Their Call ID: 3c26700d7c83-60hzuh819b4l@snom360-0004132384F8 Their Tag 27tc0487d3 Our tag: as3a18e567 [Feb 27 07:30:50] DEBUG[19189]: chan_sip.c:4727 find_call: = No match Their Call ID: 3c26700c5091-h3po9dsz8ull@snom320-0004132480B4 Their Tag 0xqyctmw0w Our tag: as385f4cf3 [Feb 27 07:30:50] DEBUG[19189]: chan_sip.c:4727 find_call: = No match Their Call ID: 3c26700c5091-z9why8kfv9np@snom320-0004132480B4 Their Tag st7fjxq5un Our tag: as0c33aea5 [Feb 27 07:30:50] DEBUG[19189]: chan_sip.c:4727 find_call: = No match Their Call ID: 3c26700c5091-6fozttksacr3@snom320-0004132480B4 Their Tag xriv93rx4h Our tag: as0d02ce20 [Feb 27 07:30:50] DEBUG[19189]: chan_sip.c:4727 find_call: = No match Their Call ID: 3dacddc4-e99e4206-6bcb53c5@132.64.4.121 Their Tag 3755C748-BA41B887 Our tag: as01510987 [Feb 27 07:30:50] DEBUG[19189]: chan_sip.c:4727 find_call: = No match Their Call ID: 64aeec39-22798afb-f4d0577a@132.64.4.121 Their Tag D7A78ABD-FB1A96BC Our tag: as760e1762 [Feb 27 07:30:50] DEBUG[19189]: chan_sip.c:4727 find_call: = No match Their Call ID: d8620a30-ea21a472-fe5c3b31@132.64.4.121 Their Tag 20C25FB4-AE7E55F3 Our tag: as7672f197 [Feb 27 07:30:50] DEBUG[19189]: chan_sip.c:4727 find_call: = No match Their Call ID: 8004e467-8179a29-4fa4eb28@132.64.4.121 Their Tag 519930EB-5285016A Our tag: as5302e30a [Feb 27 07:30:50] DEBUG[19189]: chan_sip.c:4727 find_call: = No match Their Call ID: 8d5eade-28d1dc20-204d75f@132.64.4.121 Their Tag B1DA6E62-DC10921 Our tag: as7b7a15e6 [Feb 27 07:30:50] DEBUG[19189]: chan_sip.c:4727 find_call: = No match Their Call ID: 6e178d95-48cada57-ddda6fd6@132.64.4.121 Their Tag DB388819-37C8DD18 Our tag: as3b303d11 [Feb 27 07:30:50] DEBUG[19189]: chan_sip.c:4727 find_call: = No match Their Call ID: d0103c8c-b8104ce-f188248d@132.64.4.121 Their Tag 9069EE10-7036ED4F Our tag: as069fd33e [Feb 27 07:30:50] DEBUG[19189]: chan_sip.c:4727 find_call: = No match Their Call ID: 3b280b12-7a0a228d-8e47726c@132.64.4.135 Their Tag 49667ADF-A4719856 Our tag: as5557dee1 [Feb 27 07:30:50] DEBUG[19189]: chan_sip.c:4727 find_call: = No match Their Call ID: 8b0a67c3-5276cb85-68746584@132.64.4.121 Their Tag D4291047-F5A9A9C6 Our tag: as45c7a8f4 [Feb 27 07:30:50] DEBUG[19189]: chan_sip.c:4727 find_call: = No match Their Call ID: 3e849123-fbaf994a-7e2c465@132.64.4.135 Their Tag 16D7DF64-1DDA1477 Our tag: as24fb17f2 [Feb 27 07:30:50] DEBUG[19189]: chan_sip.c:4727 find_call: = No match Their Call ID: 9b324fb8-b84e063b-fc90d882@132.64.4.135 Their Tag 8EBD433D-EA6E55C Our tag: as29972d1c [Feb 27 07:30:50] DEBUG[19189]: chan_sip.c:4727 find_call: = No match Their Call ID: ec263f3f-92e9c536-331b9e1@132.64.4.135 Their Tag A4477F30-85603053 Our tag: as7669c5c0 [Feb 27 07:30:50] DEBUG[19189]: chan_sip.c:4727 find_call: = No match Their Call ID: a5ea4d44-f2eeacd7-89f392ee@132.64.4.135 Their Tag BE7AA539-DE8177A8 Our tag: as2b2c3ef9 [Feb 27 07:30:50] DEBUG[19189]: chan_sip.c:4727 find_call: = No match Their Call ID: 1441609d-20c6773c-c0e53f6f@132.64.4.135 Their Tag E3C2A1A6-6469BD91 Our tag: as790f216d [Feb 27 07:30:50] DEBUG[19189]: chan_sip.c:4727 find_call: = No match Their Call ID: f959139a-faf1d075-45de9a34@132.64.4.135 Their Tag 88E25707-5607D15E Our tag: as3fb45c9c [Feb 27 07:30:50] DEBUG[19189]: chan_sip.c:4727 find_call: = No match Their Call ID: 9948bdcb-e36cd8d2-acc85d4d@132.64.4.135 Their Tag 5C04962C-6412539F Our tag: as5ad1b515 <--- Transmitting (no NAT) to 128.139.26.6:50245 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 128.139.26.6:5060;branch=z9hG4bK5B02481;received=128.139.26.6 From: "YEHAVI BOURVIN" ;tag=270AED48-1804 To: ;tag=as268edf01 Call-ID: D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6 CSeq: 102 BYE User-Agent: Asterisk PBX SVN-trunk-r56126M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> [Feb 27 07:30:50] DEBUG[19189]: chan_sip.c:4791 find_call: That's odd... Got a request in unknown dialog. Callid D12B806-C55B11DB-94F086A5-87E354FD@128.139.26.6 [Feb 27 07:30:50] DEBUG[19189]: chan_sip.c:15450 sipsock_read: Invalid SIP message - rejected , no callid, len 422 pbx-gr*CLI> quit Executing last minute cleanups Asterisk ending (0).