Asterisk SVN-branch-1.2-r36377, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark Spencer Asterisk comes with ABSOLUTELY NO WARRANTY; type '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 'show license' for details. ========================================================================= Connected to Asterisk SVN-branch-1.2-r36377 currently running on phone2 (pid = 23506) phone2*CLI> set debug 4 phone2*CLI> Core debug was 0 and is now 4 set Jul 3 01:26:26 DEBUG[23515]: chan_sip.c:3144 sip_alloc: Allocating new SIP dialog for A03FC774-DC9D-472A-BCBE-388C15D77FF5@10.10.11.145 - OPTIONS (No RTP) set Jul 3 01:26:26 DEBUG[23515]: chan_sip.c:11140 handle_request: **** Received OPTIONS (3) - Command in SIP OPTIONS set Jul 3 01:26:26 DEBUG[23515]: chan_sip.c:11341 sipsock_read: SIP message could not be handled, bad request: A03FC774-DC9D-472A-BCBE-388C15D77FF5@10.10.11.145 set veJul 3 01:26:26 DEBUG[23528]: manager.c:1249 process_message: Manager received command 'Command' set veJul 3 01:26:26 DEBUG[23528]: manager.c:1249 process_message: Manager received command 'Command' set veJul 3 01:26:26 DEBUG[23528]: manager.c:1249 process_message: Manager received command 'Command' set verbose 4 phone2*CLI> Verbosity was 0 and is now 4 sip debug phone2*CLI> SIP Debugging enabled <-- SIP read from 10.10.11.200:5060: OPTIONS sip:10.10.11.61 SIP/2.0 Via: SIP/2.0/UDP 10.10.11.200;rport;branch=z9hG4bK0a0a0bc80000001044a83a060000476e000008d7 Content-Length: 0 Call-ID: 4E03B055-DDE1-488D-82FA-B9555A34F797@10.10.11.200 CSeq: 758 OPTIONS From: ;tag=6698214031884 Max-Forwards: 70 To: Jul 3 01:26:29 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 0: OPTIONS sip:10.10.11.61 SIP/2.0 (31) Jul 3 01:26:29 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 1: Via: SIP/2.0/UDP 10.10.11.200;rport;branch=z9hG4bK0a0a0bc80000001044a83a060000476e000008d7 (90) Jul 3 01:26:29 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 2: Content-Length: 0 (17) Jul 3 01:26:29 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 3: Call-ID: 4E03B055-DDE1-488D-82FA-B9555A34F797@10.10.11.200 (58) Jul 3 01:26:29 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 4: CSeq: 758 OPTIONS (17) Jul 3 01:26:29 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 5: From: ;tag=6698214031884 (46) Jul 3 01:26:29 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 6: Max-Forwards: 70 (16) Jul 3 01:26:29 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 7: To: (21) Jul 3 01:26:29 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 8: (0) --- (8 headers 0 lines)--- Jul 3 01:26:29 DEBUG[23515]: chan_sip.c:3144 sip_alloc: Allocating new SIP dialog for 4E03B055-DDE1-488D-82FA-B9555A34F797@10.10.11.200 - OPTIONS (No RTP) Jul 3 01:26:29 DEBUG[23515]: chan_sip.c:11140 handle_request: **** Received OPTIONS (3) - Command in SIP OPTIONS Looking for s in fromoutside (domain 10.10.11.61) Transmitting (NAT) to 10.10.11.200:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.10.11.200;branch=z9hG4bK0a0a0bc80000001044a83a060000476e000008d7;received=10.10.11.200;rport=5060 From: ;tag=6698214031884 To: ;tag=as4b8f9d17 Call-ID: 4E03B055-DDE1-488D-82FA-B9555A34F797@10.10.11.200 CSeq: 758 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Jul 3 01:26:29 DEBUG[23515]: chan_sip.c:11341 sipsock_read: SIP message could not be handled, bad request: 4E03B055-DDE1-488D-82FA-B9555A34F797@10.10.11.200 Destroying call '4E03B055-DDE1-488D-82FA-B9555A34F797@10.10.11.200' <-- SIP read from 195.14.53.142:5060: OPTIONS sip:phone2.masterhost.ru SIP/2.0 Via: SIP/2.0/UDP 192.168.200.218;rport;branch=z9hG4bKc0a8c8da0000001044a839ff000015bf00001fcc Content-Length: 0 Call-ID: 169469BD-4B0D-4AEC-B64B-77E6BB62D403@192.168.200.218 CSeq: 4242 OPTIONS From: ;tag=22463925014166 Max-Forwards: 70 To: Jul 3 01:26:32 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 0: OPTIONS sip:phone2.masterhost.ru SIP/2.0 (40) Jul 3 01:26:32 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.200.218;rport;branch=z9hG4bKc0a8c8da0000001044a839ff000015bf00001fcc (93) Jul 3 01:26:32 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 2: Content-Length: 0 (17) Jul 3 01:26:32 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 3: Call-ID: 169469BD-4B0D-4AEC-B64B-77E6BB62D403@192.168.200.218 (61) Jul 3 01:26:32 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 4: CSeq: 4242 OPTIONS (18) Jul 3 01:26:32 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 5: From: ;tag=22463925014166 (56) Jul 3 01:26:32 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 6: Max-Forwards: 70 (16) Jul 3 01:26:32 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 7: To: (30) Jul 3 01:26:32 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 8: (0) --- (8 headers 0 lines)--- Jul 3 01:26:32 DEBUG[23515]: chan_sip.c:3144 sip_alloc: Allocating new SIP dialog for 169469BD-4B0D-4AEC-B64B-77E6BB62D403@192.168.200.218 - OPTIONS (No RTP) Jul 3 01:26:32 DEBUG[23515]: chan_sip.c:11140 handle_request: **** Received OPTIONS (3) - Command in SIP OPTIONS Looking for s in fromoutside (domain phone2.masterhost.ru) Transmitting (NAT) to 195.14.53.142:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.200.218;branch=z9hG4bKc0a8c8da0000001044a839ff000015bf00001fcc;received=195.14.53.142;rport=5060 From: ;tag=22463925014166 To: ;tag=as0887996b Call-ID: 169469BD-4B0D-4AEC-B64B-77E6BB62D403@192.168.200.218 CSeq: 4242 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Jul 3 01:26:32 DEBUG[23515]: chan_sip.c:11341 sipsock_read: SIP message could not be handled, bad request: 169469BD-4B0D-4AEC-B64B-77E6BB62D403@192.168.200.218 Destroying call '169469BD-4B0D-4AEC-B64B-77E6BB62D403@192.168.200.218' <-- SIP read from 10.10.11.134:5060: OPTIONS sip:10.10.11.61 SIP/2.0 Via: SIP/2.0/UDP 10.10.11.134;rport;branch=z9hG4bK0a0a0b860000001044a83a0b0000080700010967 Content-Length: 0 Call-ID: 23D02ACD-0BD6-43A6-B6CB-DF5066A7D3AA@10.10.11.134 CSeq: 28472 OPTIONS From: ;tag=80349531229070 Max-Forwards: 70 To: Jul 3 01:26:35 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 0: OPTIONS sip:10.10.11.61 SIP/2.0 (31) Jul 3 01:26:35 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 1: Via: SIP/2.0/UDP 10.10.11.134;rport;branch=z9hG4bK0a0a0b860000001044a83a0b0000080700010967 (90) Jul 3 01:26:35 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 2: Content-Length: 0 (17) Jul 3 01:26:35 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 3: Call-ID: 23D02ACD-0BD6-43A6-B6CB-DF5066A7D3AA@10.10.11.134 (58) Jul 3 01:26:35 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 4: CSeq: 28472 OPTIONS (19) Jul 3 01:26:35 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 5: From: ;tag=80349531229070 (47) Jul 3 01:26:35 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 6: Max-Forwards: 70 (16) Jul 3 01:26:35 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 7: To: (21) Jul 3 01:26:35 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 8: (0) --- (8 headers 0 lines)--- Jul 3 01:26:35 DEBUG[23515]: chan_sip.c:3144 sip_alloc: Allocating new SIP dialog for 23D02ACD-0BD6-43A6-B6CB-DF5066A7D3AA@10.10.11.134 - OPTIONS (No RTP) Jul 3 01:26:35 DEBUG[23515]: chan_sip.c:11140 handle_request: **** Received OPTIONS (3) - Command in SIP OPTIONS Looking for s in fromoutside (domain 10.10.11.61) Transmitting (NAT) to 10.10.11.134:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.10.11.134;branch=z9hG4bK0a0a0b860000001044a83a0b0000080700010967;received=10.10.11.134;rport=5060 From: ;tag=80349531229070 To: ;tag=as44959c93 Call-ID: 23D02ACD-0BD6-43A6-B6CB-DF5066A7D3AA@10.10.11.134 CSeq: 28472 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Jul 3 01:26:35 DEBUG[23515]: chan_sip.c:11341 sipsock_read: SIP message could not be handled, bad request: 23D02ACD-0BD6-43A6-B6CB-DF5066A7D3AA@10.10.11.134 Destroying call '23D02ACD-0BD6-43A6-B6CB-DF5066A7D3AA@10.10.11.134' Jul 3 01:26:36 DEBUG[23515]: chan_sip.c:1323 __sip_autodestruct: Auto destroying call 'f56ccabb1757e7656efc62192e9f4ff0@10.15.6.248' Destroying call 'f56ccabb1757e7656efc62192e9f4ff0@10.15.6.248' <-- SIP read from 85.140.10.142:62443: OPTIONS sip:phone2.masterhost.ru SIP/2.0 Via: SIP/2.0/UDP 10.10.10.7;rport;branch=z9hG4bK0a0a0a070000000b44a83a0b0000711f00000210 Content-Length: 0 Call-ID: 1F1AB8A8-761A-4E01-9221-D9AB61AC8B76@10.10.10.7 CSeq: 190 OPTIONS From: ;tag=4797165352479 Max-Forwards: 70 To: Jul 3 01:26:36 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 0: OPTIONS sip:phone2.masterhost.ru SIP/2.0 (40) Jul 3 01:26:36 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 1: Via: SIP/2.0/UDP 10.10.10.7;rport;branch=z9hG4bK0a0a0a070000000b44a83a0b0000711f00000210 (88) Jul 3 01:26:36 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 2: Content-Length: 0 (17) Jul 3 01:26:36 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 3: Call-ID: 1F1AB8A8-761A-4E01-9221-D9AB61AC8B76@10.10.10.7 (56) Jul 3 01:26:36 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 4: CSeq: 190 OPTIONS (17) Jul 3 01:26:36 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 5: From: ;tag=4797165352479 (55) Jul 3 01:26:36 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 6: Max-Forwards: 70 (16) Jul 3 01:26:36 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 7: To: (30) Jul 3 01:26:36 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 8: (0) --- (8 headers 0 lines)--- Jul 3 01:26:36 DEBUG[23515]: chan_sip.c:3144 sip_alloc: Allocating new SIP dialog for 1F1AB8A8-761A-4E01-9221-D9AB61AC8B76@10.10.10.7 - OPTIONS (No RTP) Jul 3 01:26:36 DEBUG[23515]: chan_sip.c:11140 handle_request: **** Received OPTIONS (3) - Command in SIP OPTIONS Looking for s in fromoutside (domain phone2.masterhost.ru) Transmitting (NAT) to 85.140.10.142:62443: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.10.10.7;branch=z9hG4bK0a0a0a070000000b44a83a0b0000711f00000210;received=85.140.10.142;rport=62443 From: ;tag=4797165352479 To: ;tag=as2cb22832 Call-ID: 1F1AB8A8-761A-4E01-9221-D9AB61AC8B76@10.10.10.7 CSeq: 190 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Jul 3 01:26:36 DEBUG[23515]: chan_sip.c:11341 sipsock_read: SIP message could not be handled, bad request: 1F1AB8A8-761A-4E01-9221-D9AB61AC8B76@10.10.10.7 Destroying call '1F1AB8A8-761A-4E01-9221-D9AB61AC8B76@10.10.10.7' <-- SIP read from 10.10.11.201:5060: OPTIONS sip:10.10.11.61 SIP/2.0 Via: SIP/2.0/UDP 10.10.11.201;rport;branch=z9hG4bK0a0a0bc90000001044a83a110000158000011807 Content-Length: 0 Call-ID: 75817351-940B-4522-A96B-ABF0B97EDC36@10.10.11.201 CSeq: 23475 OPTIONS From: ;tag=8041500269921 Max-Forwards: 70 To: Jul 3 01:26:41 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 0: OPTIONS sip:10.10.11.61 SIP/2.0 (31) Jul 3 01:26:41 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 1: Via: SIP/2.0/UDP 10.10.11.201;rport;branch=z9hG4bK0a0a0bc90000001044a83a110000158000011807 (90) Jul 3 01:26:41 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 2: Content-Length: 0 (17) Jul 3 01:26:41 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 3: Call-ID: 75817351-940B-4522-A96B-ABF0B97EDC36@10.10.11.201 (58) Jul 3 01:26:41 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 4: CSeq: 23475 OPTIONS (19) Jul 3 01:26:41 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 5: From: ;tag=8041500269921 (46) Jul 3 01:26:41 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 6: Max-Forwards: 70 (16) Jul 3 01:26:41 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 7: To: (21) Jul 3 01:26:41 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 8: (0) --- (8 headers 0 lines)--- Jul 3 01:26:41 DEBUG[23515]: chan_sip.c:3144 sip_alloc: Allocating new SIP dialog for 75817351-940B-4522-A96B-ABF0B97EDC36@10.10.11.201 - OPTIONS (No RTP) Jul 3 01:26:41 DEBUG[23515]: chan_sip.c:11140 handle_request: **** Received OPTIONS (3) - Command in SIP OPTIONS Looking for s in fromoutside (domain 10.10.11.61) Transmitting (NAT) to 10.10.11.201:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.10.11.201;branch=z9hG4bK0a0a0bc90000001044a83a110000158000011807;received=10.10.11.201;rport=5060 From: ;tag=8041500269921 To: ;tag=as42d88a65 Call-ID: 75817351-940B-4522-A96B-ABF0B97EDC36@10.10.11.201 CSeq: 23475 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Jul 3 01:26:41 DEBUG[23515]: chan_sip.c:11341 sipsock_read: SIP message could not be handled, bad request: 75817351-940B-4522-A96B-ABF0B97EDC36@10.10.11.201 Destroying call '75817351-940B-4522-A96B-ABF0B97EDC36@10.10.11.201' -- Accepting call from '9166104314' to '7729720' on channel 0/1, span 2 Jul 3 01:26:43 DEBUG[23510]: devicestate.c:187 do_state_change: Changing state for Zap/32 - state 2 (In use) Jul 3 01:26:43 DEBUG[23519]: chan_zap.c:1408 zt_enable_ec: No echo cancellation requested Jul 3 01:26:43 DEBUG[23529]: pbx.c:1677 pbx_extension_helper: Launching 'Answer' -- Executing Answer("Zap/32-1", "") in new stack Jul 3 01:26:43 DEBUG[23530]: app_queue.c:490 changethread: Device 'Zap/32' changed to state '2' (In use) but we don't care because they're not a member of any queue. Jul 3 01:26:43 DEBUG[23510]: channel.c:777 channel_find_locked: Avoiding initial deadlock for 'Zap/32-1' Jul 3 01:26:43 DEBUG[23529]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '20060703-012643' Jul 3 01:26:43 DEBUG[23529]: pbx.c:1677 pbx_extension_helper: Launching 'Monitor' -- Executing Monitor("Zap/32-1", "|in-20060703-012643-9166104314|mb") in new stack Jul 3 01:26:43 DEBUG[23529]: pbx.c:1677 pbx_extension_helper: Launching 'Wait' -- Executing Wait("Zap/32-1", "0.5") in new stack Jul 3 01:26:43 DEBUG[23510]: devicestate.c:187 do_state_change: Changing state for Zap/32 - state 2 (In use) Jul 3 01:26:43 DEBUG[23531]: app_queue.c:490 changethread: Device 'Zap/32' changed to state '2' (In use) but we don't care because they're not a member of any queue. Jul 3 01:26:43 DEBUG[23529]: pbx.c:1677 pbx_extension_helper: Launching 'Queue' -- Executing Queue("Zap/32-1", "callcenterq|tw") in new stack Jul 3 01:26:43 DEBUG[23529]: app_queue.c:3046 queue_exec: NO QUEUE_PRIO variable found. Using default. Jul 3 01:26:43 DEBUG[23529]: app_queue.c:3054 queue_exec: queue: callcenterq, options: tw, url: (null), announce: (null), expires: 0, priority: 0 Jul 3 01:26:43 DEBUG[23529]: app_queue.c:1153 join_queue: Queue 'callcenterq' Join, Channel 'Zap/32-1', Position '1' -- Started music on hold, class 'default', on Zap/32-1 Jul 3 01:26:43 DEBUG[23529]: channel.c:1724 ast_settimeout: Scheduling timer at 160 sample intervals Jul 3 01:26:43 DEBUG[23529]: app_queue.c:1997 is_our_turn: It's our turn (Zap/32-1). Jul 3 01:26:43 DEBUG[23529]: app_queue.c:2193 try_calling: Zap/32-1 is trying to call a queue member. Jul 3 01:26:43 DEBUG[23529]: app_queue.c:2217 try_calling: Simple queue (no URL) Jul 3 01:26:43 DEBUG[23529]: app_queue.c:1610 ring_one: (Parallel) Trying 'Local/1211@toagent' with metric 0 Jul 3 01:26:43 DEBUG[23510]: devicestate.c:187 do_state_change: Changing state for Local/1211@toagent - state 2 (In use) Jul 3 01:26:43 DEBUG[23529]: channel.c:2836 ast_channel_inherit_variables: Not copying variable STACK-fromoutside-7729720-4. Jul 3 01:26:43 DEBUG[23532]: app_queue.c:496 changethread: Device 'Local/1211@toagent' changed to state '2' (In use) Jul 3 01:26:43 DEBUG[23529]: channel.c:2836 ast_channel_inherit_variables: Not copying variable STACK-fromoutside-7729720-3. Jul 3 01:26:43 DEBUG[23529]: channel.c:2836 ast_channel_inherit_variables: Not copying variable AUTO_MONITOR. Jul 3 01:26:43 DEBUG[23529]: channel.c:2836 ast_channel_inherit_variables: Not copying variable STACK-fromoutside-7729720-2. Jul 3 01:26:43 DEBUG[23529]: channel.c:2836 ast_channel_inherit_variables: Not copying variable STACK-fromoutside-7729720-1. Jul 3 01:26:43 DEBUG[23529]: channel.c:2836 ast_channel_inherit_variables: Not copying variable CALLEDTON. Jul 3 01:26:43 DEBUG[23529]: channel.c:2836 ast_channel_inherit_variables: Not copying variable ANI2. Jul 3 01:26:43 DEBUG[23529]: channel.c:2836 ast_channel_inherit_variables: Not copying variable TRANSFERCAPABILITY. -- Called Local/1211@toagent Jul 3 01:26:43 DEBUG[23533]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'SIP/1211' Jul 3 01:26:43 DEBUG[23533]: pbx.c:1677 pbx_extension_helper: Launching 'Set' -- Executing Set("Local/1211@toagent-8456,2", "chan=SIP/1211") in new stack Jul 3 01:26:43 DEBUG[23533]: pbx.c:1677 pbx_extension_helper: Launching 'ChanIsAvail' -- Executing ChanIsAvail("Local/1211@toagent-8456,2", "SIP/1211|s") in new stack Jul 3 01:26:43 DEBUG[23533]: chan_sip.c:11671 sip_devicestate: Checking device state for peer 1211 Jul 3 01:26:43 DEBUG[23533]: chan_sip.c:3144 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) Jul 3 01:26:43 DEBUG[23533]: chan_sip.c:1874 create_addr_from_peer: Setting NAT on RTP to 524288 Jul 3 01:26:43 DEBUG[23533]: chan_sip.c:1878 create_addr_from_peer: Setting NAT on VRTP to 524288 Jul 3 01:26:43 DEBUG[23533]: channel.c:1336 ast_hangup: Hanging up channel 'SIP/1211-08225c50' Jul 3 01:26:43 DEBUG[23533]: chan_sip.c:2415 sip_hangup: Hangup call SIP/1211-08225c50, SIP callid 7d1a13860005811440aedb943210c3fa@masterhost.ru) Jul 3 01:26:43 DEBUG[23533]: chan_sip.c:2423 sip_hangup: update_call_counter(1211) - decrement call limit counter Jul 3 01:26:43 DEBUG[23533]: chan_sip.c:2206 update_call_counter: Updating call counter for incoming call Jul 3 01:26:43 DEBUG[23533]: pbx.c:1677 pbx_extension_helper: Launching 'NoOp' -- Executing NoOp("Local/1211@toagent-8456,2", "0") in new stack Jul 3 01:26:43 DEBUG[23510]: chan_sip.c:11671 sip_devicestate: Checking device state for peer 1211 Jul 3 01:26:43 DEBUG[23533]: pbx.c:1589 pbx_substitute_variables_helper_full: Expression result is '0' Jul 3 01:26:43 DEBUG[23533]: pbx.c:1677 pbx_extension_helper: Launching 'GotoIf' -- Executing GotoIf("Local/1211@toagent-8456,2", "0?unavail") in new stack Jul 3 01:26:43 DEBUG[23533]: pbx.c:6178 pbx_builtin_gotoif: Not taking any branch Jul 3 01:26:43 DEBUG[23510]: devicestate.c:187 do_state_change: Changing state for SIP/1211 - state 1 (Not in use) Jul 3 01:26:43 DEBUG[23533]: pbx.c:1589 pbx_substitute_variables_helper_full: Expression result is '0' Jul 3 01:26:43 DEBUG[23533]: pbx.c:1677 pbx_extension_helper: Launching 'GotoIf' -- Executing GotoIf("Local/1211@toagent-8456,2", "0?busy") in new stack Jul 3 01:26:43 DEBUG[23534]: app_queue.c:490 changethread: Device 'SIP/1211' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. Jul 3 01:26:43 DEBUG[23533]: pbx.c:6178 pbx_builtin_gotoif: Not taking any branch Jul 3 01:26:43 DEBUG[23533]: pbx.c:1677 pbx_extension_helper: Launching 'Dial' -- Executing Dial("Local/1211@toagent-8456,2", "SIP/1211|30|tw") in new stack Jul 3 01:26:43 DEBUG[23533]: chan_sip.c:3144 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) Jul 3 01:26:43 DEBUG[23533]: chan_sip.c:1874 create_addr_from_peer: Setting NAT on RTP to 524288 Jul 3 01:26:43 DEBUG[23533]: chan_sip.c:1878 create_addr_from_peer: Setting NAT on VRTP to 524288 Destroying call '7d1a13860005811440aedb943210c3fa@masterhost.ru' Jul 3 01:26:43 DEBUG[23533]: channel.c:2836 ast_channel_inherit_variables: Not copying variable STACK-toagent-1211-6. Jul 3 01:26:43 DEBUG[23533]: channel.c:2836 ast_channel_inherit_variables: Not copying variable STACK-toagent-1211-5. Jul 3 01:26:43 DEBUG[23533]: channel.c:2836 ast_channel_inherit_variables: Not copying variable STACK-toagent-1211-4. Jul 3 01:26:43 DEBUG[23533]: channel.c:2836 ast_channel_inherit_variables: Not copying variable STACK-toagent-1211-3. Jul 3 01:26:43 DEBUG[23533]: channel.c:2836 ast_channel_inherit_variables: Not copying variable AVAILSTATUS. Jul 3 01:26:43 DEBUG[23533]: channel.c:2836 ast_channel_inherit_variables: Not copying variable AVAILORIGCHAN. Jul 3 01:26:43 DEBUG[23533]: channel.c:2836 ast_channel_inherit_variables: Not copying variable AVAILCHAN. Jul 3 01:26:43 DEBUG[23533]: channel.c:2836 ast_channel_inherit_variables: Not copying variable STACK-toagent-1211-2. Jul 3 01:26:43 DEBUG[23533]: channel.c:2836 ast_channel_inherit_variables: Not copying variable chan. Jul 3 01:26:43 DEBUG[23533]: channel.c:2836 ast_channel_inherit_variables: Not copying variable STACK-toagent-1211-1. Jul 3 01:26:43 DEBUG[23533]: channel.c:2836 ast_channel_inherit_variables: Not copying variable DB_RESULT. Jul 3 01:26:43 DEBUG[23533]: chan_sip.c:2068 sip_call: Outgoing Call for 1211 Jul 3 01:26:43 DEBUG[23533]: chan_sip.c:2206 update_call_counter: Updating call counter for outgoing call We're at 83.222.22.242 port 12820 Video is at 83.222.22.242 port 17500 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x80000 (h263) to SDP Adding non-codec 0x1 (telephone-event) to SDP Jul 3 01:26:43 DEBUG[23533]: chan_sip.c:3360 parse_request: Header 0: INVITE sip:1211@10.10.10.7:5060 SIP/2.0 (39) Jul 3 01:26:43 DEBUG[23533]: chan_sip.c:3360 parse_request: Header 1: Via: SIP/2.0/UDP 83.222.22.242:5060;branch=z9hG4bK0e2574d7;rport (64) Jul 3 01:26:43 DEBUG[23533]: chan_sip.c:3360 parse_request: Header 2: From: "9166104314" ;tag=as1249a25f (64) Jul 3 01:26:43 DEBUG[23533]: chan_sip.c:3360 parse_request: Header 3: To: (30) Jul 3 01:26:43 DEBUG[23533]: chan_sip.c:3360 parse_request: Header 4: Contact: (39) Jul 3 01:26:43 DEBUG[23533]: chan_sip.c:3360 parse_request: Header 5: Call-ID: 70bbb3cf70f2429302325f3b63f5e59b@masterhost.ru (55) Jul 3 01:26:43 DEBUG[23533]: chan_sip.c:3360 parse_request: Header 6: CSeq: 102 INVITE (16) Jul 3 01:26:43 DEBUG[23533]: chan_sip.c:3360 parse_request: Header 7: User-Agent: Asterisk PBX (24) Jul 3 01:26:43 DEBUG[23533]: chan_sip.c:3360 parse_request: Header 8: Max-Forwards: 70 (16) Jul 3 01:26:43 DEBUG[23533]: chan_sip.c:3360 parse_request: Header 9: Date: Sun, 02 Jul 2006 21:26:43 GMT (35) Jul 3 01:26:43 DEBUG[23533]: chan_sip.c:3360 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Jul 3 01:26:43 DEBUG[23533]: chan_sip.c:3360 parse_request: Header 11: Content-Type: application/sdp (29) Jul 3 01:26:43 DEBUG[23533]: chan_sip.c:3360 parse_request: Header 12: Content-Length: 315 (19) Jul 3 01:26:43 DEBUG[23533]: chan_sip.c:3360 parse_request: Header 13: (0) Jul 3 01:26:43 DEBUG[23533]: chan_sip.c:3392 parse_request: Line: v=0 (3) Jul 3 01:26:43 DEBUG[23533]: chan_sip.c:3392 parse_request: Line: o=root 23506 23506 IN IP4 83.222.22.242 (39) Jul 3 01:26:43 DEBUG[23533]: chan_sip.c:3392 parse_request: Line: s=session (9) Jul 3 01:26:43 DEBUG[23533]: chan_sip.c:3392 parse_request: Line: c=IN IP4 83.222.22.242 (22) Jul 3 01:26:43 DEBUG[23533]: chan_sip.c:3392 parse_request: Line: t=0 0 (5) Jul 3 01:26:43 DEBUG[23533]: chan_sip.c:3392 parse_request: Line: m=audio 12820 RTP/AVP 0 8 3 101 (31) Jul 3 01:26:43 DEBUG[23533]: chan_sip.c:3392 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) Jul 3 01:26:43 DEBUG[23533]: chan_sip.c:3392 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) Jul 3 01:26:43 DEBUG[23533]: chan_sip.c:3392 parse_request: Line: a=rtpmap:3 GSM/8000 (19) Jul 3 01:26:43 DEBUG[23533]: chan_sip.c:3392 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) Jul 3 01:26:43 DEBUG[23533]: chan_sip.c:3392 parse_request: Line: a=fmtp:101 0-16 (15) Jul 3 01:26:43 DEBUG[23533]: chan_sip.c:3392 parse_request: Line: a=silenceSupp:off - - - - (25) Jul 3 01:26:43 DEBUG[23533]: chan_sip.c:3392 parse_request: Line: m=video 17500 RTP/AVP 34 (24) Jul 3 01:26:43 DEBUG[23533]: chan_sip.c:3392 parse_request: Line: a=rtpmap:34 H263/90000 (22) 13 headers, 14 lines Reliably Transmitting (NAT) to 85.140.10.142:62443: INVITE sip:1211@10.10.10.7:5060 SIP/2.0 Via: SIP/2.0/UDP 83.222.22.242:5060;branch=z9hG4bK0e2574d7;rport From: "9166104314" ;tag=as1249a25f To: Contact: Call-ID: 70bbb3cf70f2429302325f3b63f5e59b@masterhost.ru CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sun, 02 Jul 2006 21:26:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 315 v=0 o=root 23506 23506 IN IP4 83.222.22.242 s=session c=IN IP4 83.222.22.242 t=0 0 m=audio 12820 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - m=video 17500 RTP/AVP 34 a=rtpmap:34 H263/90000 --- Jul 3 01:26:43 DEBUG[23533]: chan_sip.c:1293 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #184 -- Called 1211 Jul 3 01:26:43 DEBUG[23533]: channel.c:2363 set_format: Set channel SIP/1211-0822deb8 to read format slin Jul 3 01:26:43 DEBUG[23533]: channel.c:2363 set_format: Set channel Local/1211@toagent-8456,2 to write format slin Jul 3 01:26:43 DEBUG[23533]: channel.c:2363 set_format: Set channel Local/1211@toagent-8456,2 to read format slin Jul 3 01:26:43 DEBUG[23533]: channel.c:2363 set_format: Set channel SIP/1211-0822deb8 to write format slin Jul 3 01:26:43 DEBUG[23529]: channel.c:1988 ast_read: Generator got voice, switching to phase locked mode Jul 3 01:26:43 DEBUG[23529]: channel.c:1724 ast_settimeout: Scheduling timer at 0 sample intervals Jul 3 01:26:43 DEBUG[23529]: channel.c:2363 set_format: Set channel Zap/32-1 to write format ulaw Jul 3 01:26:43 DEBUG[23529]: res_musiconhold.c:233 ast_moh_files_next: Zap/32-1 Opened file 0 '/var/lib/asterisk/mohmp3/misc/We_Are_The_Champions_Queen' <-- SIP read from 85.140.10.142:62443: SIP/2.0 100 Trying Via: SIP/2.0/UDP 83.222.22.242:5060;rport=5060;received=83.222.22.242;branch=z9hG4bK0e2574d7 Content-Length: 0 Call-ID: 70bbb3cf70f2429302325f3b63f5e59b@masterhost.ru CSeq: 102 INVITE From: "9166104314";tag=as1249a25f Server: SJphone/1.60.289a (SJ Labs) To: "operator name";tag=47972382613179 Jul 3 01:26:44 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 0: SIP/2.0 100 Trying (18) Jul 3 01:26:44 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 1: Via: SIP/2.0/UDP 83.222.22.242:5060;rport=5060;received=83.222.22.242;branch=z9hG4bK0e2574d7 (92) Jul 3 01:26:44 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 2: Content-Length: 0 (17) Jul 3 01:26:44 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 3: Call-ID: 70bbb3cf70f2429302325f3b63f5e59b@masterhost.ru (55) Jul 3 01:26:44 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 4: CSeq: 102 INVITE (16) Jul 3 01:26:44 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 5: From: "9166104314";tag=as1249a25f (63) Jul 3 01:26:44 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 6: Server: SJphone/1.60.289a (SJ Labs) (35) Jul 3 01:26:44 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 7: To: "operator name";tag=47972382613179 (66) Jul 3 01:26:44 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 8: (0) --- (8 headers 0 lines)--- Jul 3 01:26:44 DEBUG[23515]: chan_sip.c:1445 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #184 - INVITE (got response) Jul 3 01:26:44 DEBUG[23515]: chan_sip.c:1454 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '70bbb3cf70f2429302325f3b63f5e59b@masterhost.ru' Request 102: Found Jul 3 01:26:44 DEBUG[23515]: chan_sip.c:9581 handle_response_invite: SIP response 100 to standard invite <-- SIP read from 85.140.10.142:62443: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 83.222.22.242:5060;rport=5060;received=83.222.22.242;branch=z9hG4bK0e2574d7 Content-Length: 0 Contact: Call-ID: 70bbb3cf70f2429302325f3b63f5e59b@masterhost.ru CSeq: 102 INVITE From: "9166104314";tag=as1249a25f Server: SJphone/1.60.289a (SJ Labs) To: "operator name";tag=47972382613179 Jul 3 01:26:44 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 0: SIP/2.0 180 Ringing (19) Jul 3 01:26:44 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 1: Via: SIP/2.0/UDP 83.222.22.242:5060;rport=5060;received=83.222.22.242;branch=z9hG4bK0e2574d7 (92) Jul 3 01:26:44 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 2: Content-Length: 0 (17) Jul 3 01:26:44 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 3: Contact: (35) Jul 3 01:26:44 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 4: Call-ID: 70bbb3cf70f2429302325f3b63f5e59b@masterhost.ru (55) Jul 3 01:26:44 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 5: CSeq: 102 INVITE (16) Jul 3 01:26:44 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 6: From: "9166104314";tag=as1249a25f (63) Jul 3 01:26:44 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 7: Server: SJphone/1.60.289a (SJ Labs) (35) Jul 3 01:26:44 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 8: To: "operator name";tag=47972382613179 (66) Jul 3 01:26:44 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 9: (0) --- (9 headers 0 lines)--- Jul 3 01:26:44 DEBUG[23515]: chan_sip.c:1454 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '70bbb3cf70f2429302325f3b63f5e59b@masterhost.ru' Request 102: Found Jul 3 01:26:44 DEBUG[23515]: chan_sip.c:9581 handle_response_invite: SIP response 180 to standard invite Jul 3 01:26:44 DEBUG[23510]: chan_sip.c:11671 sip_devicestate: Checking device state for peer 1211 -- SIP/1211-0822deb8 is ringing Jul 3 01:26:44 DEBUG[23510]: devicestate.c:187 do_state_change: Changing state for SIP/1211 - state 6 (Ringing) Jul 3 01:26:44 DEBUG[23535]: app_queue.c:490 changethread: Device 'SIP/1211' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. -- Local/1211@toagent-8456,1 is ringing <-- SIP read from 85.140.10.142:62443: SIP/2.0 200 OK Via: SIP/2.0/UDP 83.222.22.242:5060;rport=5060;received=83.222.22.242;branch=z9hG4bK0e2574d7 Content-Length: 214 Contact: Call-ID: 70bbb3cf70f2429302325f3b63f5e59b@masterhost.ru Content-Type: application/sdp CSeq: 102 INVITE From: "9166104314";tag=as1249a25f Server: SJphone/1.60.289a (SJ Labs) To: "operator name";tag=47972382613179 v=0 o=- 3360864403 3360864403 IN IP4 10.10.10.7 s=SJphone c=IN IP4 10.10.10.7 t=0 0 a=direction:active m=audio 49168 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11,16 Jul 3 01:26:47 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 0: SIP/2.0 200 OK (14) Jul 3 01:26:47 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 1: Via: SIP/2.0/UDP 83.222.22.242:5060;rport=5060;received=83.222.22.242;branch=z9hG4bK0e2574d7 (92) Jul 3 01:26:47 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 2: Content-Length: 214 (19) Jul 3 01:26:47 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 3: Contact: (35) Jul 3 01:26:47 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 4: Call-ID: 70bbb3cf70f2429302325f3b63f5e59b@masterhost.ru (55) Jul 3 01:26:47 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 5: Content-Type: application/sdp (29) Jul 3 01:26:47 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 6: CSeq: 102 INVITE (16) Jul 3 01:26:47 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 7: From: "9166104314";tag=as1249a25f (63) Jul 3 01:26:47 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 8: Server: SJphone/1.60.289a (SJ Labs) (35) Jul 3 01:26:47 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 9: To: "operator name";tag=47972382613179 (66) Jul 3 01:26:47 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 10: (0) Jul 3 01:26:47 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: v=0 (3) Jul 3 01:26:47 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: o=- 3360864403 3360864403 IN IP4 10.10.10.7 (43) Jul 3 01:26:47 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: s=SJphone (9) Jul 3 01:26:47 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: c=IN IP4 10.10.10.7 (19) Jul 3 01:26:47 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: t=0 0 (5) Jul 3 01:26:47 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: a=direction:active (18) Jul 3 01:26:47 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: m=audio 49168 RTP/AVP 3 101 (27) Jul 3 01:26:47 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: a=rtpmap:3 GSM/8000 (19) Jul 3 01:26:47 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) Jul 3 01:26:47 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: a=fmtp:101 0-11,16 (18) --- (10 headers 10 lines)--- Jul 3 01:26:47 DEBUG[23515]: chan_sip.c:1379 __sip_ack: Acked pending invite 102 Jul 3 01:26:47 DEBUG[23515]: chan_sip.c:1401 __sip_ack: Stopping retransmission on '70bbb3cf70f2429302325f3b63f5e59b@masterhost.ru' of Request 102: Match Found Jul 3 01:26:47 DEBUG[23515]: chan_sip.c:9581 handle_response_invite: SIP response 200 to standard invite Found RTP audio format 3 Found RTP audio format 101 Peer audio RTP is at port 10.10.10.7:49168 Jul 3 01:26:47 DEBUG[23515]: chan_sip.c:3601 process_sdp: Peer audio RTP is at port 10.10.10.7:49168 Peer video RTP is at port 10.10.10.7:65535 Jul 3 01:26:47 DEBUG[23515]: chan_sip.c:3625 process_sdp: Peer video RTP is at port 10.10.10.7:65535 Found description format GSM Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x2 (gsm)/video=0x0 (nothing), combined - 0x2 (gsm) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Jul 3 01:26:47 DEBUG[23515]: chan_sip.c:3698 process_sdp: Oooh, we need to change our formats since our peer supports only 0x2 (gsm) and not 0x4 (ulaw) Jul 3 01:26:47 DEBUG[23515]: channel.c:2363 set_format: Set channel SIP/1211-0822deb8 to read format slin Jul 3 01:26:47 DEBUG[23515]: channel.c:2363 set_format: Set channel SIP/1211-0822deb8 to write format slin Jul 3 01:26:47 DEBUG[23515]: chan_sip.c:6134 build_route: build_route: Contact hop: list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.10.10.7, port 5060 Transmitting (NAT) to 85.140.10.142:62443: ACK sip:1211@10.10.10.7:5060 SIP/2.0 Via: SIP/2.0/UDP 83.222.22.242:5060;branch=z9hG4bK100dc830;rport From: "9166104314" ;tag=as1249a25f To: ;tag=47972382613179 Contact: Call-ID: 70bbb3cf70f2429302325f3b63f5e59b@masterhost.ru CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/1211-0822deb8 answered Local/1211@toagent-8456,2 Jul 3 01:26:47 DEBUG[23510]: chan_sip.c:11671 sip_devicestate: Checking device state for peer 1211 Jul 3 01:26:47 DEBUG[23510]: devicestate.c:187 do_state_change: Changing state for SIP/1211 - state 2 (In use) Jul 3 01:26:47 DEBUG[23536]: app_queue.c:490 changethread: Device 'SIP/1211' changed to state '2' (In use) but we don't care because they're not a member of any queue. Jul 3 01:26:47 DEBUG[23529]: app_queue.c:1930 wait_for_answer: Dunno what to do with control type -1 Jul 3 01:26:47 DEBUG[23533]: channel.c:2363 set_format: Set channel Local/1211@toagent-8456,2 to read format slin Jul 3 01:26:47 DEBUG[23533]: channel.c:2363 set_format: Set channel SIP/1211-0822deb8 to write format slin Jul 3 01:26:47 DEBUG[23533]: channel.c:2363 set_format: Set channel SIP/1211-0822deb8 to read format slin Jul 3 01:26:47 DEBUG[23533]: channel.c:2363 set_format: Set channel Local/1211@toagent-8456,2 to write format slin Jul 3 01:26:47 DEBUG[23510]: channel.c:777 channel_find_locked: Avoiding initial deadlock for 'Local/1211@toagent-8456,2' -- Local/1211@toagent-8456,1 answered Zap/32-1 Jul 3 01:26:47 DEBUG[23529]: chan_zap.c:2766 zt_setoption: Set option TONE VERIFY, mode: MUTECONF(1) on Zap/32-1 -- Stopped music on hold on Zap/32-1 Jul 3 01:26:47 DEBUG[23529]: channel.c:2363 set_format: Set channel Zap/32-1 to write format alaw Jul 3 01:26:47 DEBUG[23529]: channel.c:1724 ast_settimeout: Scheduling timer at 0 sample intervals Jul 3 01:26:47 DEBUG[23529]: channel.c:2363 set_format: Set channel Zap/32-1 to read format slin Jul 3 01:26:47 DEBUG[23529]: channel.c:2363 set_format: Set channel Local/1211@toagent-8456,1 to write format slin Jul 3 01:26:47 DEBUG[23529]: channel.c:2363 set_format: Set channel Local/1211@toagent-8456,1 to read format slin Jul 3 01:26:47 DEBUG[23529]: channel.c:2363 set_format: Set channel Zap/32-1 to write format slin Jul 3 01:26:47 DEBUG[23510]: devicestate.c:187 do_state_change: Changing state for Local/1211@toagent - state 2 (In use) Jul 3 01:26:47 DEBUG[23537]: app_queue.c:496 changethread: Device 'Local/1211@toagent' changed to state '2' (In use) Jul 3 01:26:47 DEBUG[23510]: devicestate.c:187 do_state_change: Changing state for Local/1211@toagent - state 2 (In use) Jul 3 01:26:47 DEBUG[23538]: app_queue.c:496 changethread: Device 'Local/1211@toagent' changed to state '2' (In use) Jul 3 01:26:47 DEBUG[23533]: rtp.c:479 ast_rtp_read: RTP NAT: Got audio from other end. Now sending to address 85.140.10.142:62558 Jul 3 01:26:47 DEBUG[23533]: channel.c:2770 ast_channel_masquerade: Planning to masquerade channel SIP/1211-0822deb8 into the structure of Local/1211@toagent-8456,1 Jul 3 01:26:47 DEBUG[23533]: channel.c:2783 ast_channel_masquerade: Done planning to masquerade channel SIP/1211-0822deb8 into the structure of Local/1211@toagent-8456,1 Jul 3 01:26:47 DEBUG[23533]: chan_local.c:256 local_write: Not posting to queue since already masked on 'Local/1211@toagent-8456,2' Jul 3 01:26:47 DEBUG[23529]: channel.c:2893 ast_do_masquerade: Actually Masquerading SIP/1211-0822deb8(6) into the structure of Local/1211@toagent-8456,1(6) Jul 3 01:26:47 DEBUG[23529]: channel.c:2904 ast_do_masquerade: Got clone lock for masquerade on 'SIP/1211-0822deb8' at 0x8235db4 Jul 3 01:26:47 DEBUG[23533]: channel.c:3288 ast_generic_bridge: Didn't get a frame from channel: Local/1211@toagent-8456,2 Jul 3 01:26:47 DEBUG[23529]: channel.c:2363 set_format: Set channel SIP/1211-0822deb8 to write format slin Jul 3 01:26:47 DEBUG[23533]: channel.c:3563 ast_channel_bridge: Bridge stops bridging channels Local/1211@toagent-8456,2 and Local/1211@toagent-8456,1 Jul 3 01:26:47 DEBUG[23529]: channel.c:2363 set_format: Set channel SIP/1211-0822deb8 to read format slin Jul 3 01:26:47 DEBUG[23529]: channel.c:3065 ast_do_masquerade: Putting channel SIP/1211-0822deb8 in 64/64 formats Jul 3 01:26:47 DEBUG[23529]: channel.c:3100 ast_do_masquerade: Released clone lock on 'Local/1211@toagent-8456,1' Jul 3 01:26:47 DEBUG[23529]: channel.c:3109 ast_do_masquerade: Done Masquerading SIP/1211-0822deb8 (6) Jul 3 01:26:47 DEBUG[23533]: channel.c:1341 ast_hangup: Hanging up zombie 'Local/1211@toagent-8456,1' Jul 3 01:26:47 DEBUG[23529]: channel.c:2363 set_format: Set channel Zap/32-1 to read format slin Jul 3 01:26:47 DEBUG[23533]: app_dial.c:1628 dial_exec_full: Exiting with DIALSTATUS=ANSWER. Jul 3 01:26:47 DEBUG[23529]: channel.c:2363 set_format: Set channel SIP/1211-0822deb8 to write format slin Jul 3 01:26:47 DEBUG[23510]: devicestate.c:187 do_state_change: Changing state for Local/1211@toagent - state 2 (In use) Jul 3 01:26:47 DEBUG[23533]: pbx.c:2316 __ast_pbx_run: Spawn extension (toagent,1211,6) exited non-zero on 'Local/1211@toagent-8456,2' Jul 3 01:26:47 DEBUG[23529]: channel.c:2363 set_format: Set channel SIP/1211-0822deb8 to read format slin Jul 3 01:26:47 DEBUG[23529]: channel.c:2363 set_format: Set channel Zap/32-1 to write format slin Jul 3 01:26:47 DEBUG[23533]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '9166104314' Jul 3 01:26:47 DEBUG[23539]: app_queue.c:496 changethread: Device 'Local/1211@toagent' changed to state '2' (In use) Jul 3 01:26:47 DEBUG[23533]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '9166104314' Jul 3 01:26:47 DEBUG[23533]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '1211' Jul 3 01:26:47 DEBUG[23533]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'toagent' Jul 3 01:26:47 DEBUG[23533]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'Local/1211@toagent-8456,2' Jul 3 01:26:47 DEBUG[23533]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'SIP/1211-0822deb8' Jul 3 01:26:47 DEBUG[23533]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'Dial' Jul 3 01:26:47 DEBUG[23533]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'SIP/1211|30|tw' Jul 3 01:26:47 DEBUG[23533]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '2006-07-03 01:26:43' Jul 3 01:26:47 DEBUG[23533]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '2006-07-03 01:26:47' Jul 3 01:26:47 DEBUG[23533]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '2006-07-03 01:26:47' Jul 3 01:26:47 DEBUG[23533]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '4' Jul 3 01:26:47 DEBUG[23533]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '0' Jul 3 01:26:47 DEBUG[23533]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'ANSWERED' Jul 3 01:26:47 DEBUG[23533]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'DOCUMENTATION' Jul 3 01:26:47 DEBUG[23533]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '(null)' Jul 3 01:26:47 DEBUG[23533]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '1151875603.2' Jul 3 01:26:47 DEBUG[23533]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '(null)' Jul 3 01:26:47 DEBUG[23533]: channel.c:1336 ast_hangup: Hanging up channel 'Local/1211@toagent-8456,2' Jul 3 01:26:47 DEBUG[23510]: devicestate.c:187 do_state_change: Changing state for Local/1211@toagent - state 0 (Unknown) Jul 3 01:26:47 DEBUG[23540]: app_queue.c:496 changethread: Device 'Local/1211@toagent' changed to state '0' (Unknown) Jul 3 01:26:47 DEBUG[23529]: rtp.c:1352 ast_rtp_write: Ooh, format changed from unknown to gsm <-- SIP read from 85.140.10.142:62443: INVITE sip:9166104314@83.222.22.242 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.7;rport;branch=z9hG4bK0a0a0a070000007c44a83a170000318700000214 Content-Length: 211 Contact: Call-ID: 70bbb3cf70f2429302325f3b63f5e59b@masterhost.ru Content-Type: application/sdp CSeq: 1 INVITE From: "operator name";tag=47972382613179 Max-Forwards: 70 To: ;tag=as1249a25f User-Agent: SJphone/1.60.289a (SJ Labs) v=0 o=- 3360864403 3360864404 IN IP4 10.10.10.7 s=SJphone c=IN IP4 0.0.0.0 t=0 0 a=direction:active m=audio 49168 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11,16 Jul 3 01:26:48 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 0: INVITE sip:9166104314@83.222.22.242 SIP/2.0 (43) Jul 3 01:26:48 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 1: Via: SIP/2.0/UDP 10.10.10.7;rport;branch=z9hG4bK0a0a0a070000007c44a83a170000318700000214 (88) Jul 3 01:26:48 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 2: Content-Length: 211 (19) Jul 3 01:26:48 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 3: Contact: (35) Jul 3 01:26:48 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 4: Call-ID: 70bbb3cf70f2429302325f3b63f5e59b@masterhost.ru (55) Jul 3 01:26:48 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 5: Content-Type: application/sdp (29) Jul 3 01:26:48 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 6: CSeq: 1 INVITE (14) Jul 3 01:26:48 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 7: From: "operator name";tag=47972382613179 (68) Jul 3 01:26:48 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 8: Max-Forwards: 70 (16) Jul 3 01:26:48 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 9: To: ;tag=as1249a25f (49) Jul 3 01:26:48 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 10: User-Agent: SJphone/1.60.289a (SJ Labs) (39) Jul 3 01:26:48 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 11: (0) Jul 3 01:26:48 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: v=0 (3) Jul 3 01:26:48 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: o=- 3360864403 3360864404 IN IP4 10.10.10.7 (43) Jul 3 01:26:48 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: s=SJphone (9) Jul 3 01:26:48 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: c=IN IP4 0.0.0.0 (16) Jul 3 01:26:48 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: t=0 0 (5) Jul 3 01:26:48 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: a=direction:active (18) Jul 3 01:26:48 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: m=audio 49168 RTP/AVP 3 101 (27) Jul 3 01:26:48 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: a=rtpmap:3 GSM/8000 (19) Jul 3 01:26:48 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) Jul 3 01:26:48 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: a=fmtp:101 0-11,16 (18) --- (11 headers 10 lines)--- Jul 3 01:26:48 DEBUG[23515]: chan_sip.c:11140 handle_request: **** Received INVITE (5) - Command in SIP INVITE Using INVITE request as basis request - 70bbb3cf70f2429302325f3b63f5e59b@masterhost.ru Sending to 10.10.10.7 : 5060 (NAT) Found RTP audio format 3 Found RTP audio format 101 Peer audio RTP is at port 0.0.0.0:49168 Jul 3 01:26:48 DEBUG[23515]: chan_sip.c:3601 process_sdp: Peer audio RTP is at port 0.0.0.0:49168 Peer video RTP is at port 0.0.0.0:65535 Jul 3 01:26:48 DEBUG[23515]: chan_sip.c:3625 process_sdp: Peer video RTP is at port 0.0.0.0:65535 Found description format GSM Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x2 (gsm)/video=0x0 (nothing), combined - 0x2 (gsm) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) -- Started music on hold, class 'default', on Zap/32-1 Jul 3 01:26:48 DEBUG[23515]: channel.c:1724 ast_settimeout: Scheduling timer at 160 sample intervals Jul 3 01:26:48 DEBUG[23515]: chan_sip.c:10553 handle_request_invite: Got a SIP re-invite for call 70bbb3cf70f2429302325f3b63f5e59b@masterhost.ru We're at 83.222.22.242 port 12820 Video is at 83.222.22.242 port 17500 Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 85.140.10.142:62443: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.7;branch=z9hG4bK0a0a0a070000007c44a83a170000318700000214;received=85.140.10.142;rport=62443 From: "operator name";tag=47972382613179 To: ;tag=as1249a25f Call-ID: 70bbb3cf70f2429302325f3b63f5e59b@masterhost.ru CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 217 v=0 o=root 23506 23507 IN IP4 83.222.22.242 s=session c=IN IP4 83.222.22.242 t=0 0 m=audio 12820 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Jul 3 01:26:48 DEBUG[23515]: chan_sip.c:1293 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #185 Jul 3 01:26:48 DEBUG[23529]: channel.c:1988 ast_read: Generator got voice, switching to phase locked mode Jul 3 01:26:48 DEBUG[23529]: channel.c:1724 ast_settimeout: Scheduling timer at 0 sample intervals Jul 3 01:26:48 DEBUG[23529]: channel.c:2363 set_format: Set channel Zap/32-1 to write format ulaw Jul 3 01:26:48 DEBUG[23529]: res_musiconhold.c:233 ast_moh_files_next: Zap/32-1 Opened file 0 '/var/lib/asterisk/mohmp3/misc/We_Are_The_Champions_Queen' <-- SIP read from 85.140.10.142:62443: ACK sip:9166104314@83.222.22.242 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.7;rport;branch=z9hG4bK0a0a0a070000007c44a83a170000730100000216 Content-Length: 0 Call-ID: 70bbb3cf70f2429302325f3b63f5e59b@masterhost.ru CSeq: 1 ACK From: "operator name";tag=47972382613179 Max-Forwards: 70 To: ;tag=as1249a25f User-Agent: SJphone/1.60.289a (SJ Labs) Jul 3 01:26:48 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 0: ACK sip:9166104314@83.222.22.242 SIP/2.0 (40) Jul 3 01:26:48 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 1: Via: SIP/2.0/UDP 10.10.10.7;rport;branch=z9hG4bK0a0a0a070000007c44a83a170000730100000216 (88) Jul 3 01:26:48 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 2: Content-Length: 0 (17) Jul 3 01:26:48 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 3: Call-ID: 70bbb3cf70f2429302325f3b63f5e59b@masterhost.ru (55) Jul 3 01:26:48 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 4: CSeq: 1 ACK (11) Jul 3 01:26:48 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 5: From: "operator name";tag=47972382613179 (68) Jul 3 01:26:48 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 6: Max-Forwards: 70 (16) Jul 3 01:26:48 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 7: To: ;tag=as1249a25f (49) Jul 3 01:26:48 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 8: User-Agent: SJphone/1.60.289a (SJ Labs) (39) Jul 3 01:26:48 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 9: (0) --- (9 headers 0 lines)--- Jul 3 01:26:48 DEBUG[23515]: chan_sip.c:11140 handle_request: **** Received ACK (6) - Command in SIP ACK Jul 3 01:26:48 DEBUG[23515]: chan_sip.c:1390 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #185 Jul 3 01:26:48 DEBUG[23515]: chan_sip.c:1401 __sip_ack: Stopping retransmission on '70bbb3cf70f2429302325f3b63f5e59b@masterhost.ru' of Response 1: Match Found <-- SIP read from 10.10.11.200:5060: OPTIONS sip:10.10.11.61 SIP/2.0 Via: SIP/2.0/UDP 10.10.11.200;rport;branch=z9hG4bK0a0a0bc80000001044a83a1a00002b8b000008d9 Content-Length: 0 Call-ID: 4E03B055-DDE1-488D-82FA-B9555A34F797@10.10.11.200 CSeq: 759 OPTIONS From: ;tag=6700214016822 Max-Forwards: 70 To: Jul 3 01:26:49 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 0: OPTIONS sip:10.10.11.61 SIP/2.0 (31) Jul 3 01:26:49 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 1: Via: SIP/2.0/UDP 10.10.11.200;rport;branch=z9hG4bK0a0a0bc80000001044a83a1a00002b8b000008d9 (90) Jul 3 01:26:49 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 2: Content-Length: 0 (17) Jul 3 01:26:49 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 3: Call-ID: 4E03B055-DDE1-488D-82FA-B9555A34F797@10.10.11.200 (58) Jul 3 01:26:49 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 4: CSeq: 759 OPTIONS (17) Jul 3 01:26:49 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 5: From: ;tag=6700214016822 (46) Jul 3 01:26:49 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 6: Max-Forwards: 70 (16) Jul 3 01:26:49 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 7: To: (21) Jul 3 01:26:49 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 8: (0) --- (8 headers 0 lines)--- Jul 3 01:26:49 DEBUG[23515]: chan_sip.c:3144 sip_alloc: Allocating new SIP dialog for 4E03B055-DDE1-488D-82FA-B9555A34F797@10.10.11.200 - OPTIONS (No RTP) Jul 3 01:26:49 DEBUG[23515]: chan_sip.c:11140 handle_request: **** Received OPTIONS (3) - Command in SIP OPTIONS Looking for s in fromoutside (domain 10.10.11.61) Transmitting (NAT) to 10.10.11.200:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.10.11.200;branch=z9hG4bK0a0a0bc80000001044a83a1a00002b8b000008d9;received=10.10.11.200;rport=5060 From: ;tag=6700214016822 To: ;tag=as5da2f672 Call-ID: 4E03B055-DDE1-488D-82FA-B9555A34F797@10.10.11.200 CSeq: 759 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Jul 3 01:26:49 DEBUG[23515]: chan_sip.c:11341 sipsock_read: SIP message could not be handled, bad request: 4E03B055-DDE1-488D-82FA-B9555A34F797@10.10.11.200 Destroying call '4E03B055-DDE1-488D-82FA-B9555A34F797@10.10.11.200' <-- SIP read from 85.140.10.142:62443: INVITE sip:7677@phone2.masterhost.ru SIP/2.0 Via: SIP/2.0/UDP 10.10.10.7;rport;branch=z9hG4bK0a0a0a070000007e44a83a1a0000198b00000217 Content-Length: 285 Contact: Call-ID: 47260494-3736-49E0-B7F9-87B25C7A9B24@10.10.10.7 Content-Type: application/sdp CSeq: 1 INVITE From: "operator name";tag=47973139712706 Max-Forwards: 70 To: User-Agent: SJphone/1.60.289a (SJ Labs) v=0 o=- 3360864410 3360864410 IN IP4 10.10.10.7 s=SJphone c=IN IP4 10.10.10.7 t=0 0 a=direction:active m=audio 49170 RTP/AVP 3 97 98 101 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11,16 Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 0: INVITE sip:7677@phone2.masterhost.ru SIP/2.0 (44) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 1: Via: SIP/2.0/UDP 10.10.10.7;rport;branch=z9hG4bK0a0a0a070000007e44a83a1a0000198b00000217 (88) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 2: Content-Length: 285 (19) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 3: Contact: (35) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 4: Call-ID: 47260494-3736-49E0-B7F9-87B25C7A9B24@10.10.10.7 (56) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 5: Content-Type: application/sdp (29) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 6: CSeq: 1 INVITE (14) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 7: From: "operator name";tag=47973139712706 (73) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 8: Max-Forwards: 70 (16) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 9: To: (35) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 10: User-Agent: SJphone/1.60.289a (SJ Labs) (39) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 11: (0) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: v=0 (3) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: o=- 3360864410 3360864410 IN IP4 10.10.10.7 (43) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: s=SJphone (9) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: c=IN IP4 10.10.10.7 (19) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: t=0 0 (5) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: a=direction:active (18) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: m=audio 49170 RTP/AVP 3 97 98 101 (33) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: a=rtpmap:3 GSM/8000 (19) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: a=rtpmap:97 iLBC/8000 (21) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: a=rtpmap:98 iLBC/8000 (21) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: a=fmtp:98 mode=20 (17) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: a=fmtp:101 0-11,16 (18) --- (11 headers 13 lines)--- Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3144 sip_alloc: Allocating new SIP dialog for 47260494-3736-49E0-B7F9-87B25C7A9B24@10.10.10.7 - INVITE (With RTP) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:11140 handle_request: **** Received INVITE (5) - Command in SIP INVITE Using INVITE request as basis request - 47260494-3736-49E0-B7F9-87B25C7A9B24@10.10.10.7 Sending to 10.10.10.7 : 5060 (NAT) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:7152 check_user_full: Setting NAT on RTP to 524288 Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:7156 check_user_full: Setting NAT on VRTP to 524288 Reliably Transmitting (NAT) to 85.140.10.142:62443: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.10.10.7;branch=z9hG4bK0a0a0a070000007e44a83a1a0000198b00000217;received=85.140.10.142;rport=62443 From: "operator name";tag=47973139712706 To: ;tag=as449fcb54 Call-ID: 47260494-3736-49E0-B7F9-87B25C7A9B24@10.10.10.7 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Proxy-Authenticate: Digest realm="phone1.masterhost.ru", nonce="57377122" Content-Length: 0 --- Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:1293 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #186 Scheduling destruction of call '47260494-3736-49E0-B7F9-87B25C7A9B24@10.10.10.7' in 15000 ms Found user '1211' <-- SIP read from 85.140.10.142:62443: ACK sip:7677@phone2.masterhost.ru SIP/2.0 Via: SIP/2.0/UDP 10.10.10.7;rport;branch=z9hG4bK0a0a0a070000007e44a83a1a0000198b00000217 Content-Length: 0 Call-ID: 47260494-3736-49E0-B7F9-87B25C7A9B24@10.10.10.7 CSeq: 1 ACK From: "operator name";tag=47973139712706 Max-Forwards: 70 To: ;tag=as449fcb54 User-Agent: SJphone/1.60.289a (SJ Labs) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 0: ACK sip:7677@phone2.masterhost.ru SIP/2.0 (41) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 1: Via: SIP/2.0/UDP 10.10.10.7;rport;branch=z9hG4bK0a0a0a070000007e44a83a1a0000198b00000217 (88) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 2: Content-Length: 0 (17) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 3: Call-ID: 47260494-3736-49E0-B7F9-87B25C7A9B24@10.10.10.7 (56) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 4: CSeq: 1 ACK (11) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 5: From: "operator name";tag=47973139712706 (73) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 6: Max-Forwards: 70 (16) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 7: To: ;tag=as449fcb54 (50) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 8: User-Agent: SJphone/1.60.289a (SJ Labs) (39) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 9: (0) --- (9 headers 0 lines)--- Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:11140 handle_request: **** Received ACK (6) - Command in SIP ACK Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:1390 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #186 Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:1401 __sip_ack: Stopping retransmission on '47260494-3736-49E0-B7F9-87B25C7A9B24@10.10.10.7' of Response 1: Match Found <-- SIP read from 85.140.10.142:62443: INVITE sip:7677@phone2.masterhost.ru SIP/2.0 Via: SIP/2.0/UDP 10.10.10.7;rport;branch=z9hG4bK0a0a0a070000007e44a83a1a000006b500000218 Content-Length: 285 Contact: Call-ID: 47260494-3736-49E0-B7F9-87B25C7A9B24@10.10.10.7 Content-Type: application/sdp CSeq: 2 INVITE From: "operator name";tag=47973139712706 Max-Forwards: 70 To: User-Agent: SJphone/1.60.289a (SJ Labs) Proxy-Authorization: Digest username="1211",realm="phone1.masterhost.ru",nonce="57377122",uri="sip:7677@phone2.masterhost.ru",response="***" v=0 o=- 3360864410 3360864410 IN IP4 10.10.10.7 s=SJphone c=IN IP4 10.10.10.7 t=0 0 a=direction:active m=audio 49170 RTP/AVP 3 97 98 101 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11,16 Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 0: INVITE sip:7677@phone2.masterhost.ru SIP/2.0 (44) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 1: Via: SIP/2.0/UDP 10.10.10.7;rport;branch=z9hG4bK0a0a0a070000007e44a83a1a000006b500000218 (88) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 2: Content-Length: 285 (19) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 3: Contact: (35) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 4: Call-ID: 47260494-3736-49E0-B7F9-87B25C7A9B24@10.10.10.7 (56) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 5: Content-Type: application/sdp (29) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 6: CSeq: 2 INVITE (14) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 7: From: "operator name";tag=47973139712706 (73) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 8: Max-Forwards: 70 (16) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 9: To: (35) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 10: User-Agent: SJphone/1.60.289a (SJ Labs) (39) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 11: Proxy-Authorization: Digest username="1211",realm="phone1.masterhost.ru",nonce="57377122",uri="sip:7677@phone2.masterhost.ru",response="***" (169) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 12: (0) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: v=0 (3) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: o=- 3360864410 3360864410 IN IP4 10.10.10.7 (43) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: s=SJphone (9) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: c=IN IP4 10.10.10.7 (19) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: t=0 0 (5) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: a=direction:active (18) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: m=audio 49170 RTP/AVP 3 97 98 101 (33) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: a=rtpmap:3 GSM/8000 (19) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: a=rtpmap:97 iLBC/8000 (21) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: a=rtpmap:98 iLBC/8000 (21) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: a=fmtp:98 mode=20 (17) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: a=fmtp:101 0-11,16 (18) --- (12 headers 13 lines)--- Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:11140 handle_request: **** Received INVITE (5) - Command in SIP INVITE Using INVITE request as basis request - 47260494-3736-49E0-B7F9-87B25C7A9B24@10.10.10.7 Sending to 10.10.10.7 : 5060 (NAT) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:7152 check_user_full: Setting NAT on RTP to 524288 Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:7156 check_user_full: Setting NAT on VRTP to 524288 Found user '1211' Found RTP audio format 3 Found RTP audio format 97 Found RTP audio format 98 Found RTP audio format 101 Peer audio RTP is at port 10.10.10.7:49170 Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3601 process_sdp: Peer audio RTP is at port 10.10.10.7:49170 Peer video RTP is at port 10.10.10.7:65535 Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3625 process_sdp: Peer video RTP is at port 10.10.10.7:65535 Found description format GSM Found description format iLBC Found description format iLBC Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x402 (gsm|ilbc)/video=0x0 (nothing), combined - 0x2 (gsm) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:10499 handle_request_invite: Checking SIP call limits for device 1211 Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:2206 update_call_counter: Updating call counter for incoming call Looking for 7677 in fromoffice (domain phone2.masterhost.ru) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:6134 build_route: build_route: Contact hop: list_route: hop: Transmitting (NAT) to 85.140.10.142:62443: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.10.7;branch=z9hG4bK0a0a0a070000007e44a83a1a000006b500000218;received=85.140.10.142;rport=62443 From: "operator name";tag=47973139712706 To: Call-ID: 47260494-3736-49E0-B7F9-87B25C7A9B24@10.10.10.7 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Jul 3 01:26:51 DEBUG[23510]: chan_sip.c:11671 sip_devicestate: Checking device state for peer 1211 Jul 3 01:26:51 DEBUG[23510]: devicestate.c:187 do_state_change: Changing state for SIP/1211 - state 2 (In use) Jul 3 01:26:51 DEBUG[23542]: app_queue.c:490 changethread: Device 'SIP/1211' changed to state '2' (In use) but we don't care because they're not a member of any queue. Jul 3 01:26:51 DEBUG[23541]: pbx.c:1589 pbx_substitute_variables_helper_full: Expression result is '0' Jul 3 01:26:51 DEBUG[23541]: pbx.c:1677 pbx_extension_helper: Launching 'GotoIf' -- Executing GotoIf("SIP/1211-082267e8", "0?dial") in new stack Jul 3 01:26:51 DEBUG[23541]: pbx.c:6178 pbx_builtin_gotoif: Not taking any branch Jul 3 01:26:51 DEBUG[23541]: pbx.c:1677 pbx_extension_helper: Launching 'Set' -- Executing Set("SIP/1211-082267e8", "DIALOPT=TW") in new stack Jul 3 01:26:51 DEBUG[23541]: pbx.c:1677 pbx_extension_helper: Launching 'Dial' -- Executing Dial("SIP/1211-082267e8", "SIP/677@gatewayTDO|60|twTW") in new stack Jul 3 01:26:51 DEBUG[23541]: chan_sip.c:3144 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) Jul 3 01:26:51 DEBUG[23541]: chan_sip.c:1874 create_addr_from_peer: Setting NAT on RTP to 524288 Jul 3 01:26:51 DEBUG[23541]: chan_sip.c:1878 create_addr_from_peer: Setting NAT on VRTP to 524288 Jul 3 01:26:51 DEBUG[23541]: channel.c:2836 ast_channel_inherit_variables: Not copying variable STACK-fromoffice-7677-3. Jul 3 01:26:51 DEBUG[23541]: channel.c:2836 ast_channel_inherit_variables: Not copying variable DIALOPT. Jul 3 01:26:51 DEBUG[23541]: channel.c:2836 ast_channel_inherit_variables: Not copying variable STACK-fromoffice-7677-2. Jul 3 01:26:51 DEBUG[23541]: channel.c:2836 ast_channel_inherit_variables: Not copying variable STACK-fromoffice-7677-1. Jul 3 01:26:51 DEBUG[23541]: channel.c:2836 ast_channel_inherit_variables: Not copying variable SIPCALLID. Jul 3 01:26:51 DEBUG[23541]: channel.c:2836 ast_channel_inherit_variables: Not copying variable SIPUSERAGENT. Jul 3 01:26:51 DEBUG[23541]: channel.c:2836 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. Jul 3 01:26:51 DEBUG[23541]: channel.c:2836 ast_channel_inherit_variables: Not copying variable SIPURI. Jul 3 01:26:51 DEBUG[23541]: chan_sip.c:2068 sip_call: Outgoing Call for 677 Jul 3 01:26:51 DEBUG[23541]: chan_sip.c:2206 update_call_counter: Updating call counter for outgoing call We're at 10.10.10.61 port 10476 Video is at 10.10.10.61 port 11116 Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x80000 (h263) to SDP Adding non-codec 0x1 (telephone-event) to SDP Jul 3 01:26:51 DEBUG[23541]: chan_sip.c:3360 parse_request: Header 0: INVITE sip:677@10.10.10.37 SIP/2.0 (34) Jul 3 01:26:51 DEBUG[23541]: chan_sip.c:3360 parse_request: Header 1: Via: SIP/2.0/UDP 10.10.10.61:5060;branch=z9hG4bK699d533d;rport (62) Jul 3 01:26:51 DEBUG[23541]: chan_sip.c:3360 parse_request: Header 2: From: "test sjphone" ;tag=as45c1e8df (58) Jul 3 01:26:51 DEBUG[23541]: chan_sip.c:3360 parse_request: Header 3: To: (25) Jul 3 01:26:51 DEBUG[23541]: chan_sip.c:3360 parse_request: Header 4: Contact: (31) Jul 3 01:26:51 DEBUG[23541]: chan_sip.c:3360 parse_request: Header 5: Call-ID: 5da17644127ed868555762cd4955e724@10.10.10.61 (53) Jul 3 01:26:51 DEBUG[23541]: chan_sip.c:3360 parse_request: Header 6: CSeq: 102 INVITE (16) Jul 3 01:26:51 DEBUG[23541]: chan_sip.c:3360 parse_request: Header 7: User-Agent: Asterisk PBX (24) Jul 3 01:26:51 DEBUG[23541]: chan_sip.c:3360 parse_request: Header 8: Max-Forwards: 70 (16) Jul 3 01:26:51 DEBUG[23541]: chan_sip.c:3360 parse_request: Header 9: Date: Sun, 02 Jul 2006 21:26:51 GMT (35) Jul 3 01:26:51 DEBUG[23541]: chan_sip.c:3360 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Jul 3 01:26:51 DEBUG[23541]: chan_sip.c:3360 parse_request: Header 11: Content-Type: application/sdp (29) Jul 3 01:26:51 DEBUG[23541]: chan_sip.c:3360 parse_request: Header 12: Content-Length: 311 (19) Jul 3 01:26:51 DEBUG[23541]: chan_sip.c:3360 parse_request: Header 13: (0) Jul 3 01:26:51 DEBUG[23541]: chan_sip.c:3392 parse_request: Line: v=0 (3) Jul 3 01:26:51 DEBUG[23541]: chan_sip.c:3392 parse_request: Line: o=root 23506 23506 IN IP4 10.10.10.61 (37) Jul 3 01:26:51 DEBUG[23541]: chan_sip.c:3392 parse_request: Line: s=session (9) Jul 3 01:26:51 DEBUG[23541]: chan_sip.c:3392 parse_request: Line: c=IN IP4 10.10.10.61 (20) Jul 3 01:26:51 DEBUG[23541]: chan_sip.c:3392 parse_request: Line: t=0 0 (5) Jul 3 01:26:51 DEBUG[23541]: chan_sip.c:3392 parse_request: Line: m=audio 10476 RTP/AVP 3 0 8 101 (31) Jul 3 01:26:51 DEBUG[23541]: chan_sip.c:3392 parse_request: Line: a=rtpmap:3 GSM/8000 (19) Jul 3 01:26:51 DEBUG[23541]: chan_sip.c:3392 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) Jul 3 01:26:51 DEBUG[23541]: chan_sip.c:3392 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) Jul 3 01:26:51 DEBUG[23541]: chan_sip.c:3392 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) Jul 3 01:26:51 DEBUG[23541]: chan_sip.c:3392 parse_request: Line: a=fmtp:101 0-16 (15) Jul 3 01:26:51 DEBUG[23541]: chan_sip.c:3392 parse_request: Line: a=silenceSupp:off - - - - (25) Jul 3 01:26:51 DEBUG[23541]: chan_sip.c:3392 parse_request: Line: m=video 11116 RTP/AVP 34 (24) Jul 3 01:26:51 DEBUG[23541]: chan_sip.c:3392 parse_request: Line: a=rtpmap:34 H263/90000 (22) 13 headers, 14 lines Reliably Transmitting (NAT) to 10.10.10.37:5060: INVITE sip:677@10.10.10.37 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.61:5060;branch=z9hG4bK699d533d;rport From: "test sjphone" ;tag=as45c1e8df To: Contact: Call-ID: 5da17644127ed868555762cd4955e724@10.10.10.61 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sun, 02 Jul 2006 21:26:51 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 311 v=0 o=root 23506 23506 IN IP4 10.10.10.61 s=session c=IN IP4 10.10.10.61 t=0 0 m=audio 10476 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - m=video 11116 RTP/AVP 34 a=rtpmap:34 H263/90000 --- Jul 3 01:26:51 DEBUG[23541]: chan_sip.c:1293 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #188 -- Called 677@gatewayTDO Jul 3 01:26:51 DEBUG[23541]: channel.c:2363 set_format: Set channel SIP/gatewayTDO-08239888 to read format slin Jul 3 01:26:51 DEBUG[23541]: channel.c:2363 set_format: Set channel SIP/1211-082267e8 to write format slin Jul 3 01:26:51 DEBUG[23541]: channel.c:2363 set_format: Set channel SIP/1211-082267e8 to read format slin Jul 3 01:26:51 DEBUG[23541]: channel.c:2363 set_format: Set channel SIP/gatewayTDO-08239888 to write format slin <-- SIP read from 10.10.10.37:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.10.61:5060;branch=z9hG4bK699d533d;rport From: "test sjphone" ;tag=as45c1e8df To: ;tag=1c1943051071 Call-ID: 5da17644127ed868555762cd4955e724@10.10.10.61 CSeq: 102 INVITE Supported: em,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Server: Audiocodes-Sip-Gateway-MP-104 FXO/v.4.40.215.400 Content-Length: 0 Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 0: SIP/2.0 100 Trying (18) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 1: Via: SIP/2.0/UDP 10.10.10.61:5060;branch=z9hG4bK699d533d;rport (62) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 2: From: "test sjphone" ;tag=as45c1e8df (58) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 3: To: ;tag=1c1943051071 (42) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 4: Call-ID: 5da17644127ed868555762cd4955e724@10.10.10.61 (53) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 5: CSeq: 102 INVITE (16) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 6: Supported: em,timer,replaces,path (33) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 7: Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE (86) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 8: Server: Audiocodes-Sip-Gateway-MP-104 FXO/v.4.40.215.400 (56) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 9: Content-Length: 0 (17) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 10: (0) --- (10 headers 0 lines)--- Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:1445 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #188 - INVITE (got response) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:1454 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '5da17644127ed868555762cd4955e724@10.10.10.61' Request 102: Found Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:9581 handle_response_invite: SIP response 100 to standard invite <-- SIP read from 10.10.10.37:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.10.10.61:5060;branch=z9hG4bK699d533d;rport From: "test sjphone" ;tag=as45c1e8df To: ;tag=1c1943051071 Call-ID: 5da17644127ed868555762cd4955e724@10.10.10.61 CSeq: 102 INVITE Contact: Supported: em,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Server: Audiocodes-Sip-Gateway-MP-104 FXO/v.4.40.215.400 Content-Length: 0 Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 0: SIP/2.0 180 Ringing (19) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 1: Via: SIP/2.0/UDP 10.10.10.61:5060;branch=z9hG4bK699d533d;rport (62) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 2: From: "test sjphone" ;tag=as45c1e8df (58) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 3: To: ;tag=1c1943051071 (42) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 4: Call-ID: 5da17644127ed868555762cd4955e724@10.10.10.61 (53) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 5: CSeq: 102 INVITE (16) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 6: Contact: (30) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 7: Supported: em,timer,replaces,path (33) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 8: Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE (86) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 9: Server: Audiocodes-Sip-Gateway-MP-104 FXO/v.4.40.215.400 (56) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 10: Content-Length: 0 (17) Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 11: (0) --- (11 headers 0 lines)--- Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:1454 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '5da17644127ed868555762cd4955e724@10.10.10.61' Request 102: Found Jul 3 01:26:51 DEBUG[23515]: chan_sip.c:9581 handle_response_invite: SIP response 180 to standard invite Jul 3 01:26:51 DEBUG[23510]: chan_sip.c:11671 sip_devicestate: Checking device state for peer gatewayTDO -- SIP/gatewayTDO-08239888 is ringing Jul 3 01:26:51 DEBUG[23510]: devicestate.c:187 do_state_change: Changing state for SIP/gatewayTDO - state 6 (Ringing) Transmitting (NAT) to 85.140.10.142:62443: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.10.10.7;branch=z9hG4bK0a0a0a070000007e44a83a1a000006b500000218;received=85.140.10.142;rport=62443 From: "operator name";tag=47973139712706 To: ;tag=as7bd4aa0e Call-ID: 47260494-3736-49E0-B7F9-87B25C7A9B24@10.10.10.7 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Jul 3 01:26:51 DEBUG[23543]: app_queue.c:490 changethread: Device 'SIP/gatewayTDO' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. <-- SIP read from 195.14.53.142:5060: OPTIONS sip:phone2.masterhost.ru SIP/2.0 Via: SIP/2.0/UDP 192.168.200.218;rport;branch=z9hG4bKc0a8c8da0000001044a83a130000237800001fce Content-Length: 0 Call-ID: 169469BD-4B0D-4AEC-B64B-77E6BB62D403@192.168.200.218 CSeq: 4243 OPTIONS From: ;tag=2246592502778 Max-Forwards: 70 To: Jul 3 01:26:52 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 0: OPTIONS sip:phone2.masterhost.ru SIP/2.0 (40) Jul 3 01:26:52 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.200.218;rport;branch=z9hG4bKc0a8c8da0000001044a83a130000237800001fce (93) Jul 3 01:26:52 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 2: Content-Length: 0 (17) Jul 3 01:26:52 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 3: Call-ID: 169469BD-4B0D-4AEC-B64B-77E6BB62D403@192.168.200.218 (61) Jul 3 01:26:52 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 4: CSeq: 4243 OPTIONS (18) Jul 3 01:26:52 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 5: From: ;tag=2246592502778 (55) Jul 3 01:26:52 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 6: Max-Forwards: 70 (16) Jul 3 01:26:52 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 7: To: (30) Jul 3 01:26:52 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 8: (0) --- (8 headers 0 lines)--- Jul 3 01:26:52 DEBUG[23515]: chan_sip.c:3144 sip_alloc: Allocating new SIP dialog for 169469BD-4B0D-4AEC-B64B-77E6BB62D403@192.168.200.218 - OPTIONS (No RTP) Jul 3 01:26:52 DEBUG[23515]: chan_sip.c:11140 handle_request: **** Received OPTIONS (3) - Command in SIP OPTIONS Looking for s in fromoutside (domain phone2.masterhost.ru) Transmitting (NAT) to 195.14.53.142:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.200.218;branch=z9hG4bKc0a8c8da0000001044a83a130000237800001fce;received=195.14.53.142;rport=5060 From: ;tag=2246592502778 To: ;tag=as368cc219 Call-ID: 169469BD-4B0D-4AEC-B64B-77E6BB62D403@192.168.200.218 CSeq: 4243 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Jul 3 01:26:52 DEBUG[23515]: chan_sip.c:11341 sipsock_read: SIP message could not be handled, bad request: 169469BD-4B0D-4AEC-B64B-77E6BB62D403@192.168.200.218 Destroying call '169469BD-4B0D-4AEC-B64B-77E6BB62D403@192.168.200.218' <-- SIP read from 10.10.10.37:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.61:5060;branch=z9hG4bK699d533d;rport From: "test sjphone" ;tag=as45c1e8df To: ;tag=1c1943051071 Call-ID: 5da17644127ed868555762cd4955e724@10.10.10.61 CSeq: 102 INVITE Contact: Supported: em,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Server: Audiocodes-Sip-Gateway-MP-104 FXO/v.4.40.215.400 Content-Type: application/sdp Content-Length: 253 v=0 o=AudiocodesGW 1943058805 1943058739 IN IP4 10.10.10.37 s=Phone-Call c=IN IP4 10.10.10.37 t=0 0 m=audio 4030 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv m=video 0 RTP/AVP Jul 3 01:26:53 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 0: SIP/2.0 200 OK (14) Jul 3 01:26:53 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 1: Via: SIP/2.0/UDP 10.10.10.61:5060;branch=z9hG4bK699d533d;rport (62) Jul 3 01:26:53 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 2: From: "test sjphone" ;tag=as45c1e8df (58) Jul 3 01:26:53 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 3: To: ;tag=1c1943051071 (42) Jul 3 01:26:53 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 4: Call-ID: 5da17644127ed868555762cd4955e724@10.10.10.61 (53) Jul 3 01:26:53 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 5: CSeq: 102 INVITE (16) Jul 3 01:26:53 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 6: Contact: (30) Jul 3 01:26:53 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 7: Supported: em,timer,replaces,path (33) Jul 3 01:26:53 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 8: Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE (86) Jul 3 01:26:53 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 9: Server: Audiocodes-Sip-Gateway-MP-104 FXO/v.4.40.215.400 (56) Jul 3 01:26:53 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 10: Content-Type: application/sdp (29) Jul 3 01:26:53 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 11: Content-Length: 253 (19) Jul 3 01:26:53 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 12: (0) Jul 3 01:26:53 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: v=0 (3) Jul 3 01:26:53 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: o=AudiocodesGW 1943058805 1943058739 IN IP4 10.10.10.37 (55) Jul 3 01:26:53 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: s=Phone-Call (12) Jul 3 01:26:53 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: c=IN IP4 10.10.10.37 (20) Jul 3 01:26:53 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: t=0 0 (5) Jul 3 01:26:53 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: m=audio 4030 RTP/AVP 0 101 (26) Jul 3 01:26:53 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: a=rtpmap:0 pcmu/8000 (20) Jul 3 01:26:53 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) Jul 3 01:26:53 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: a=fmtp:101 0-15 (15) Jul 3 01:26:53 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: a=ptime:20 (10) Jul 3 01:26:53 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: a=sendrecv (10) Jul 3 01:26:53 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: m=video 0 RTP/AVP (18) Jul 3 01:26:53 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: (0) --- (12 headers 13 lines)--- Jul 3 01:26:53 DEBUG[23515]: chan_sip.c:1379 __sip_ack: Acked pending invite 102 Jul 3 01:26:53 DEBUG[23515]: chan_sip.c:1401 __sip_ack: Stopping retransmission on '5da17644127ed868555762cd4955e724@10.10.10.61' of Request 102: Match Found Jul 3 01:26:53 DEBUG[23515]: chan_sip.c:9581 handle_response_invite: SIP response 200 to standard invite Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 10.10.10.37:4030 Jul 3 01:26:53 DEBUG[23515]: chan_sip.c:3601 process_sdp: Peer audio RTP is at port 10.10.10.37:4030 Found description format pcmu Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Jul 3 01:26:53 DEBUG[23515]: chan_sip.c:6134 build_route: build_route: Contact hop: list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.10.10.37, port 5060 Transmitting (NAT) to 10.10.10.37:5060: ACK sip:491@10.10.10.37 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.61:5060;branch=z9hG4bK09973fb2;rport From: "test sjphone" ;tag=as45c1e8df To: ;tag=1c1943051071 Contact: Call-ID: 5da17644127ed868555762cd4955e724@10.10.10.61 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Jul 3 01:26:53 DEBUG[23510]: chan_sip.c:11671 sip_devicestate: Checking device state for peer gatewayTDO -- SIP/gatewayTDO-08239888 answered SIP/1211-082267e8 Jul 3 01:26:53 DEBUG[23510]: channel.c:777 channel_find_locked: Avoiding initial deadlock for 'SIP/gatewayTDO-08239888' Jul 3 01:26:53 DEBUG[23541]: channel.c:2363 set_format: Set channel SIP/1211-082267e8 to read format slin Jul 3 01:26:53 DEBUG[23541]: channel.c:2363 set_format: Set channel SIP/gatewayTDO-08239888 to write format slin Jul 3 01:26:53 DEBUG[23541]: channel.c:2363 set_format: Set channel SIP/gatewayTDO-08239888 to read format slin Jul 3 01:26:53 DEBUG[23541]: channel.c:2363 set_format: Set channel SIP/1211-082267e8 to write format slin Jul 3 01:26:53 DEBUG[23541]: chan_sip.c:2537 sip_answer: sip_answer(SIP/1211-082267e8) We're at 83.222.22.242 port 19190 Video is at 83.222.22.242 port 12922 Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 85.140.10.142:62443: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.7;branch=z9hG4bK0a0a0a070000007e44a83a1a000006b500000218;received=85.140.10.142;rport=62443 From: "operator name";tag=47973139712706 To: ;tag=as7bd4aa0e Call-ID: 47260494-3736-49E0-B7F9-87B25C7A9B24@10.10.10.7 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 217 v=0 o=root 23506 23506 IN IP4 83.222.22.242 s=session c=IN IP4 83.222.22.242 t=0 0 m=audio 19190 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Jul 3 01:26:53 DEBUG[23541]: chan_sip.c:1293 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #189 -- Attempting native bridge of SIP/1211-082267e8 and SIP/gatewayTDO-08239888 Jul 3 01:26:53 DEBUG[23510]: devicestate.c:187 do_state_change: Changing state for SIP/gatewayTDO - state 2 (In use) Jul 3 01:26:53 DEBUG[23510]: chan_sip.c:11671 sip_devicestate: Checking device state for peer 1211 Jul 3 01:26:53 DEBUG[23544]: app_queue.c:490 changethread: Device 'SIP/gatewayTDO' changed to state '2' (In use) but we don't care because they're not a member of any queue. Jul 3 01:26:53 DEBUG[23510]: devicestate.c:187 do_state_change: Changing state for SIP/1211 - state 2 (In use) Jul 3 01:26:53 DEBUG[23545]: app_queue.c:490 changethread: Device 'SIP/1211' changed to state '2' (In use) but we don't care because they're not a member of any queue. Jul 3 01:26:53 DEBUG[23541]: rtp.c:1352 ast_rtp_write: Ooh, format changed from unknown to gsm Jul 3 01:26:53 DEBUG[23541]: rtp.c:411 ast_rtcp_read: Got RTCP report of 44 bytes <-- SIP read from 85.140.10.142:62443: ACK sip:7677@83.222.22.242 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.7;rport;branch=z9hG4bK0a0a0a070000007e44a83a1c00003c330000021c Content-Length: 0 Call-ID: 47260494-3736-49E0-B7F9-87B25C7A9B24@10.10.10.7 CSeq: 2 ACK From: "operator name";tag=47973139712706 Max-Forwards: 70 To: ;tag=as7bd4aa0e User-Agent: SJphone/1.60.289a (SJ Labs) Jul 3 01:26:53 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 0: ACK sip:7677@83.222.22.242 SIP/2.0 (34) Jul 3 01:26:53 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 1: Via: SIP/2.0/UDP 10.10.10.7;rport;branch=z9hG4bK0a0a0a070000007e44a83a1c00003c330000021c (88) Jul 3 01:26:53 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 2: Content-Length: 0 (17) Jul 3 01:26:53 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 3: Call-ID: 47260494-3736-49E0-B7F9-87B25C7A9B24@10.10.10.7 (56) Jul 3 01:26:53 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 4: CSeq: 2 ACK (11) Jul 3 01:26:53 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 5: From: "operator name";tag=47973139712706 (73) Jul 3 01:26:53 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 6: Max-Forwards: 70 (16) Jul 3 01:26:53 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 7: To: ;tag=as7bd4aa0e (50) Jul 3 01:26:53 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 8: User-Agent: SJphone/1.60.289a (SJ Labs) (39) Jul 3 01:26:53 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 9: (0) --- (9 headers 0 lines)--- Jul 3 01:26:53 DEBUG[23515]: chan_sip.c:11140 handle_request: **** Received ACK (6) - Command in SIP ACK Jul 3 01:26:53 DEBUG[23515]: chan_sip.c:1390 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #189 Jul 3 01:26:53 DEBUG[23515]: chan_sip.c:1401 __sip_ack: Stopping retransmission on '47260494-3736-49E0-B7F9-87B25C7A9B24@10.10.10.7' of Response 2: Match Found Jul 3 01:26:53 DEBUG[23541]: rtp.c:479 ast_rtp_read: RTP NAT: Got audio from other end. Now sending to address 85.140.10.142:62562 Jul 3 01:26:53 DEBUG[23541]: rtp.c:1352 ast_rtp_write: Ooh, format changed from unknown to ulaw <-- SIP read from 10.10.10.37:5060: BYE sip:1211@10.10.10.61 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.37;branch=z9hG4bKac1945963029 Max-Forwards: 70 From: ;tag=1c1943051071 To: "test sjphone" ;tag=as45c1e8df Call-ID: 5da17644127ed868555762cd4955e724@10.10.10.61 CSeq: 1 BYE Contact: Supported: em,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-MP-104 FXO/v.4.40.215.400 Content-Length: 0 Jul 3 01:26:54 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 0: BYE sip:1211@10.10.10.61 SIP/2.0 (32) Jul 3 01:26:54 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 1: Via: SIP/2.0/UDP 10.10.10.37;branch=z9hG4bKac1945963029 (55) Jul 3 01:26:54 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 2: Max-Forwards: 70 (16) Jul 3 01:26:54 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 3: From: ;tag=1c1943051071 (44) Jul 3 01:26:54 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 4: To: "test sjphone" ;tag=as45c1e8df (56) Jul 3 01:26:54 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 5: Call-ID: 5da17644127ed868555762cd4955e724@10.10.10.61 (53) Jul 3 01:26:54 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 6: CSeq: 1 BYE (11) Jul 3 01:26:54 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 7: Contact: (30) Jul 3 01:26:54 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 8: Supported: em,timer,replaces,path (33) Jul 3 01:26:54 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 9: Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE (86) Jul 3 01:26:54 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 10: User-Agent: Audiocodes-Sip-Gateway-MP-104 FXO/v.4.40.215.400 (60) Jul 3 01:26:54 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 11: Content-Length: 0 (17) Jul 3 01:26:54 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 12: (0) --- (12 headers 0 lines)--- Jul 3 01:26:54 DEBUG[23515]: chan_sip.c:11140 handle_request: **** Received BYE (8) - Command in SIP BYE Sending to 10.10.10.37 : 5060 (NAT) Transmitting (NAT) to 10.10.10.37:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.37;branch=z9hG4bKac1945963029;received=10.10.10.37 From: ;tag=1c1943051071 To: "test sjphone" ;tag=as45c1e8df Call-ID: 5da17644127ed868555762cd4955e724@10.10.10.61 CSeq: 1 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing --- Jul 3 01:26:54 DEBUG[23541]: channel.c:3288 ast_generic_bridge: Didn't get a frame from channel: SIP/gatewayTDO-08239888 Jul 3 01:26:54 DEBUG[23541]: channel.c:3563 ast_channel_bridge: Bridge stops bridging channels SIP/1211-082267e8 and SIP/gatewayTDO-08239888 Jul 3 01:26:54 DEBUG[23541]: channel.c:1336 ast_hangup: Hanging up channel 'SIP/gatewayTDO-08239888' Jul 3 01:26:54 DEBUG[23541]: chan_sip.c:2415 sip_hangup: Hangup call SIP/gatewayTDO-08239888, SIP callid 5da17644127ed868555762cd4955e724@10.10.10.61) Jul 3 01:26:54 DEBUG[23541]: chan_sip.c:2423 sip_hangup: update_call_counter(677) - decrement call limit counter Jul 3 01:26:54 DEBUG[23541]: chan_sip.c:2206 update_call_counter: Updating call counter for outgoing call Jul 3 01:26:54 DEBUG[23541]: app_dial.c:1628 dial_exec_full: Exiting with DIALSTATUS=ANSWER. Jul 3 01:26:54 DEBUG[23541]: pbx.c:2316 __ast_pbx_run: Spawn extension (fromoffice,7677,3) exited non-zero on 'SIP/1211-082267e8' Jul 3 01:26:54 DEBUG[23510]: chan_sip.c:11671 sip_devicestate: Checking device state for peer gatewayTDO Jul 3 01:26:54 DEBUG[23541]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '"test sjphone" <1211>' Jul 3 01:26:54 DEBUG[23541]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '1211' Jul 3 01:26:54 DEBUG[23541]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '7677' Jul 3 01:26:54 DEBUG[23510]: devicestate.c:187 do_state_change: Changing state for SIP/gatewayTDO - state 1 (Not in use) Jul 3 01:26:54 DEBUG[23541]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'fromoffice' Jul 3 01:26:54 DEBUG[23546]: app_queue.c:490 changethread: Device 'SIP/gatewayTDO' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. Jul 3 01:26:54 DEBUG[23541]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'SIP/1211-082267e8' Jul 3 01:26:54 DEBUG[23541]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'SIP/gatewayTDO-08239888' Jul 3 01:26:54 DEBUG[23541]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'Dial' Jul 3 01:26:54 DEBUG[23541]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'SIP/677@gatewayTDO|60|twTW' Jul 3 01:26:54 DEBUG[23541]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '2006-07-03 01:26:51' Jul 3 01:26:54 DEBUG[23541]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '2006-07-03 01:26:53' Jul 3 01:26:54 DEBUG[23541]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '2006-07-03 01:26:54' Jul 3 01:26:54 DEBUG[23541]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '3' Jul 3 01:26:54 DEBUG[23541]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '1' Jul 3 01:26:54 DEBUG[23541]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'ANSWERED' Jul 3 01:26:54 DEBUG[23541]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'DOCUMENTATION' Jul 3 01:26:54 DEBUG[23541]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '(null)' Jul 3 01:26:54 DEBUG[23541]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '1151875611.5' Jul 3 01:26:54 DEBUG[23541]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '(null)' Jul 3 01:26:54 DEBUG[23541]: channel.c:1336 ast_hangup: Hanging up channel 'SIP/1211-082267e8' Jul 3 01:26:54 DEBUG[23541]: chan_sip.c:2415 sip_hangup: Hangup call SIP/1211-082267e8, SIP callid 47260494-3736-49E0-B7F9-87B25C7A9B24@10.10.10.7) Jul 3 01:26:54 DEBUG[23541]: chan_sip.c:2423 sip_hangup: update_call_counter(1211) - decrement call limit counter Jul 3 01:26:54 DEBUG[23541]: chan_sip.c:2206 update_call_counter: Updating call counter for incoming call set_destination: Parsing for address/port to send to set_destination: set destination to 10.10.10.7, port 5060 Reliably Transmitting (NAT) to 85.140.10.142:62443: BYE sip:1211@10.10.10.7:5060 SIP/2.0 Via: SIP/2.0/UDP 83.222.22.242:5060;branch=z9hG4bK523f3789;rport From: ;tag=as7bd4aa0e To: "operator name";tag=47973139712706 Contact: Call-ID: 47260494-3736-49E0-B7F9-87B25C7A9B24@10.10.10.7 CSeq: 102 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Jul 3 01:26:54 DEBUG[23541]: chan_sip.c:1293 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #190 Jul 3 01:26:54 DEBUG[23510]: chan_sip.c:11671 sip_devicestate: Checking device state for peer 1211 Jul 3 01:26:54 DEBUG[23510]: devicestate.c:187 do_state_change: Changing state for SIP/1211 - state 2 (In use) Jul 3 01:26:54 DEBUG[23547]: app_queue.c:490 changethread: Device 'SIP/1211' changed to state '2' (In use) but we don't care because they're not a member of any queue. <-- SIP read from 85.140.10.142:62443: SIP/2.0 200 OK Via: SIP/2.0/UDP 83.222.22.242:5060;rport=5060;received=83.222.22.242;branch=z9hG4bK523f3789 Content-Length: 0 Call-ID: 47260494-3736-49E0-B7F9-87B25C7A9B24@10.10.10.7 CSeq: 102 BYE From: ;tag=as7bd4aa0e Server: SJphone/1.60.289a (SJ Labs) To: "operator name";tag=47973139712706 Jul 3 01:26:54 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 0: SIP/2.0 200 OK (14) Jul 3 01:26:54 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 1: Via: SIP/2.0/UDP 83.222.22.242:5060;rport=5060;received=83.222.22.242;branch=z9hG4bK523f3789 (92) Jul 3 01:26:54 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 2: Content-Length: 0 (17) Jul 3 01:26:54 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 3: Call-ID: 47260494-3736-49E0-B7F9-87B25C7A9B24@10.10.10.7 (56) Jul 3 01:26:54 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 4: CSeq: 102 BYE (13) Jul 3 01:26:54 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 5: From: ;tag=as7bd4aa0e (52) Jul 3 01:26:54 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 6: Server: SJphone/1.60.289a (SJ Labs) (35) Jul 3 01:26:54 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 7: To: "operator name";tag=47973139712706 (71) Jul 3 01:26:54 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 8: (0) --- (8 headers 0 lines)--- Jul 3 01:26:54 DEBUG[23515]: chan_sip.c:1390 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #190 Jul 3 01:26:54 DEBUG[23515]: chan_sip.c:1401 __sip_ack: Stopping retransmission on '47260494-3736-49E0-B7F9-87B25C7A9B24@10.10.10.7' of Request 102: Match Found Destroying call '5da17644127ed868555762cd4955e724@10.10.10.61' Destroying call '47260494-3736-49E0-B7F9-87B25C7A9B24@10.10.10.7' <-- SIP read from 85.140.10.142:62443: INVITE sip:9166104314@83.222.22.242 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.7;rport;branch=z9hG4bK0a0a0a070000007c44a83a1e000021ba0000021e Content-Length: 214 Contact: Call-ID: 70bbb3cf70f2429302325f3b63f5e59b@masterhost.ru Content-Type: application/sdp CSeq: 2 INVITE From: "operator name";tag=47972382613179 Max-Forwards: 70 To: ;tag=as1249a25f User-Agent: SJphone/1.60.289a (SJ Labs) v=0 o=- 3360864403 3360864405 IN IP4 10.10.10.7 s=SJphone c=IN IP4 10.10.10.7 t=0 0 a=direction:active m=audio 49168 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11,16 Jul 3 01:26:54 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 0: INVITE sip:9166104314@83.222.22.242 SIP/2.0 (43) Jul 3 01:26:54 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 1: Via: SIP/2.0/UDP 10.10.10.7;rport;branch=z9hG4bK0a0a0a070000007c44a83a1e000021ba0000021e (88) Jul 3 01:26:54 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 2: Content-Length: 214 (19) Jul 3 01:26:54 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 3: Contact: (35) Jul 3 01:26:54 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 4: Call-ID: 70bbb3cf70f2429302325f3b63f5e59b@masterhost.ru (55) Jul 3 01:26:54 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 5: Content-Type: application/sdp (29) Jul 3 01:26:54 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 6: CSeq: 2 INVITE (14) Jul 3 01:26:54 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 7: From: "operator name";tag=47972382613179 (68) Jul 3 01:26:54 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 8: Max-Forwards: 70 (16) Jul 3 01:26:54 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 9: To: ;tag=as1249a25f (49) Jul 3 01:26:54 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 10: User-Agent: SJphone/1.60.289a (SJ Labs) (39) Jul 3 01:26:54 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 11: (0) Jul 3 01:26:54 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: v=0 (3) Jul 3 01:26:54 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: o=- 3360864403 3360864405 IN IP4 10.10.10.7 (43) Jul 3 01:26:54 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: s=SJphone (9) Jul 3 01:26:54 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: c=IN IP4 10.10.10.7 (19) Jul 3 01:26:54 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: t=0 0 (5) Jul 3 01:26:54 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: a=direction:active (18) Jul 3 01:26:54 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: m=audio 49168 RTP/AVP 3 101 (27) Jul 3 01:26:54 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: a=rtpmap:3 GSM/8000 (19) Jul 3 01:26:54 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) Jul 3 01:26:54 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: a=fmtp:101 0-11,16 (18) --- (11 headers 10 lines)--- Jul 3 01:26:54 DEBUG[23515]: chan_sip.c:11140 handle_request: **** Received INVITE (5) - Command in SIP INVITE Using INVITE request as basis request - 70bbb3cf70f2429302325f3b63f5e59b@masterhost.ru Sending to 10.10.10.7 : 5060 (NAT) Found RTP audio format 3 Found RTP audio format 101 Peer audio RTP is at port 10.10.10.7:49168 Jul 3 01:26:54 DEBUG[23515]: chan_sip.c:3601 process_sdp: Peer audio RTP is at port 10.10.10.7:49168 Peer video RTP is at port 10.10.10.7:65535 Jul 3 01:26:54 DEBUG[23515]: chan_sip.c:3625 process_sdp: Peer video RTP is at port 10.10.10.7:65535 Found description format GSM Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x2 (gsm)/video=0x0 (nothing), combined - 0x2 (gsm) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) -- Stopped music on hold on Zap/32-1 Jul 3 01:26:54 DEBUG[23515]: channel.c:2363 set_format: Set channel Zap/32-1 to write format slin Jul 3 01:26:54 DEBUG[23515]: channel.c:1724 ast_settimeout: Scheduling timer at 0 sample intervals Jul 3 01:26:54 DEBUG[23515]: chan_sip.c:10553 handle_request_invite: Got a SIP re-invite for call 70bbb3cf70f2429302325f3b63f5e59b@masterhost.ru We're at 83.222.22.242 port 12820 Video is at 83.222.22.242 port 17500 Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 85.140.10.142:62443: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.7;branch=z9hG4bK0a0a0a070000007c44a83a1e000021ba0000021e;received=85.140.10.142;rport=62443 From: "operator name";tag=47972382613179 To: ;tag=as1249a25f Call-ID: 70bbb3cf70f2429302325f3b63f5e59b@masterhost.ru CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 217 v=0 o=root 23506 23508 IN IP4 83.222.22.242 s=session c=IN IP4 83.222.22.242 t=0 0 m=audio 12820 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Jul 3 01:26:54 DEBUG[23515]: chan_sip.c:1293 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #191 Jul 3 01:26:54 DEBUG[23529]: rtp.c:479 ast_rtp_read: RTP NAT: Got audio from other end. Now sending to address 85.140.10.142:62558 Jul 3 01:26:54 DEBUG[23529]: rtp.c:1260 ast_rtp_raw_write: Difference is 50720, ms is 6360 <-- SIP read from 85.140.10.142:62443: ACK sip:9166104314@83.222.22.242 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.7;rport;branch=z9hG4bK0a0a0a070000007c44a83a1e0000285100000220 Content-Length: 0 Call-ID: 70bbb3cf70f2429302325f3b63f5e59b@masterhost.ru CSeq: 2 ACK From: "operator name";tag=47972382613179 Max-Forwards: 70 To: ;tag=as1249a25f User-Agent: SJphone/1.60.289a (SJ Labs) Jul 3 01:26:55 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 0: ACK sip:9166104314@83.222.22.242 SIP/2.0 (40) Jul 3 01:26:55 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 1: Via: SIP/2.0/UDP 10.10.10.7;rport;branch=z9hG4bK0a0a0a070000007c44a83a1e0000285100000220 (88) Jul 3 01:26:55 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 2: Content-Length: 0 (17) Jul 3 01:26:55 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 3: Call-ID: 70bbb3cf70f2429302325f3b63f5e59b@masterhost.ru (55) Jul 3 01:26:55 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 4: CSeq: 2 ACK (11) Jul 3 01:26:55 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 5: From: "operator name";tag=47972382613179 (68) Jul 3 01:26:55 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 6: Max-Forwards: 70 (16) Jul 3 01:26:55 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 7: To: ;tag=as1249a25f (49) Jul 3 01:26:55 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 8: User-Agent: SJphone/1.60.289a (SJ Labs) (39) Jul 3 01:26:55 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 9: (0) --- (9 headers 0 lines)--- Jul 3 01:26:55 DEBUG[23515]: chan_sip.c:11140 handle_request: **** Received ACK (6) - Command in SIP ACK Jul 3 01:26:55 DEBUG[23515]: chan_sip.c:1390 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #191 Jul 3 01:26:55 DEBUG[23515]: chan_sip.c:1401 __sip_ack: Stopping retransmission on '70bbb3cf70f2429302325f3b63f5e59b@masterhost.ru' of Response 2: Match Found <-- SIP read from 10.10.11.134:5060: OPTIONS sip:10.10.11.61 SIP/2.0 Via: SIP/2.0/UDP 10.10.11.134;rport;branch=z9hG4bK0a0a0b860000001044a83a1f000055f800010969 Content-Length: 0 Call-ID: 23D02ACD-0BD6-43A6-B6CB-DF5066A7D3AA@10.10.11.134 CSeq: 28473 OPTIONS From: ;tag=80351531221779 Max-Forwards: 70 To: Jul 3 01:26:55 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 0: OPTIONS sip:10.10.11.61 SIP/2.0 (31) Jul 3 01:26:55 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 1: Via: SIP/2.0/UDP 10.10.11.134;rport;branch=z9hG4bK0a0a0b860000001044a83a1f000055f800010969 (90) Jul 3 01:26:55 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 2: Content-Length: 0 (17) Jul 3 01:26:55 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 3: Call-ID: 23D02ACD-0BD6-43A6-B6CB-DF5066A7D3AA@10.10.11.134 (58) Jul 3 01:26:55 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 4: CSeq: 28473 OPTIONS (19) Jul 3 01:26:55 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 5: From: ;tag=80351531221779 (47) Jul 3 01:26:55 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 6: Max-Forwards: 70 (16) Jul 3 01:26:55 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 7: To: (21) Jul 3 01:26:55 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 8: (0) --- (8 headers 0 lines)--- Jul 3 01:26:55 DEBUG[23515]: chan_sip.c:3144 sip_alloc: Allocating new SIP dialog for 23D02ACD-0BD6-43A6-B6CB-DF5066A7D3AA@10.10.11.134 - OPTIONS (No RTP) Jul 3 01:26:55 DEBUG[23515]: chan_sip.c:11140 handle_request: **** Received OPTIONS (3) - Command in SIP OPTIONS Looking for s in fromoutside (domain 10.10.11.61) Transmitting (NAT) to 10.10.11.134:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.10.11.134;branch=z9hG4bK0a0a0b860000001044a83a1f000055f800010969;received=10.10.11.134;rport=5060 From: ;tag=80351531221779 To: ;tag=as6d311e4c Call-ID: 23D02ACD-0BD6-43A6-B6CB-DF5066A7D3AA@10.10.11.134 CSeq: 28473 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Jul 3 01:26:55 DEBUG[23515]: chan_sip.c:11341 sipsock_read: SIP message could not be handled, bad request: 23D02ACD-0BD6-43A6-B6CB-DF5066A7D3AA@10.10.11.134 Destroying call '23D02ACD-0BD6-43A6-B6CB-DF5066A7D3AA@10.10.11.134' <-- SIP read from 10.10.11.145:5060: OPTIONS sip:10.10.11.61 SIP/2.0 Via: SIP/2.0/UDP 10.10.11.145;rport;branch=z9hG4bK0a0a0b910000001044a83a20000001aa00002ad1 Content-Length: 0 Call-ID: A03FC774-DC9D-472A-BCBE-388C15D77FF5@10.10.11.145 CSeq: 4299 OPTIONS From: ;tag=1292482189461 Max-Forwards: 70 To: Jul 3 01:26:56 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 0: OPTIONS sip:10.10.11.61 SIP/2.0 (31) Jul 3 01:26:56 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 1: Via: SIP/2.0/UDP 10.10.11.145;rport;branch=z9hG4bK0a0a0b910000001044a83a20000001aa00002ad1 (90) Jul 3 01:26:56 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 2: Content-Length: 0 (17) Jul 3 01:26:56 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 3: Call-ID: A03FC774-DC9D-472A-BCBE-388C15D77FF5@10.10.11.145 (58) Jul 3 01:26:56 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 4: CSeq: 4299 OPTIONS (18) Jul 3 01:26:56 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 5: From: ;tag=1292482189461 (46) Jul 3 01:26:56 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 6: Max-Forwards: 70 (16) Jul 3 01:26:56 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 7: To: (21) Jul 3 01:26:56 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 8: (0) --- (8 headers 0 lines)--- Jul 3 01:26:56 DEBUG[23515]: chan_sip.c:3144 sip_alloc: Allocating new SIP dialog for A03FC774-DC9D-472A-BCBE-388C15D77FF5@10.10.11.145 - OPTIONS (No RTP) Jul 3 01:26:56 DEBUG[23515]: chan_sip.c:11140 handle_request: **** Received OPTIONS (3) - Command in SIP OPTIONS Looking for s in fromoutside (domain 10.10.11.61) Transmitting (NAT) to 10.10.11.145:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.10.11.145;branch=z9hG4bK0a0a0b910000001044a83a20000001aa00002ad1;received=10.10.11.145;rport=5060 From: ;tag=1292482189461 To: ;tag=as0e9990d0 Call-ID: A03FC774-DC9D-472A-BCBE-388C15D77FF5@10.10.11.145 CSeq: 4299 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Jul 3 01:26:56 DEBUG[23515]: chan_sip.c:11341 sipsock_read: SIP message could not be handled, bad request: A03FC774-DC9D-472A-BCBE-388C15D77FF5@10.10.11.145 Destroying call 'A03FC774-DC9D-472A-BCBE-388C15D77FF5@10.10.11.145' <-- SIP read from 85.140.10.142:62443: OPTIONS sip:phone2.masterhost.ru SIP/2.0 Via: SIP/2.0/UDP 10.10.10.7;rport;branch=z9hG4bK0a0a0a070000000b44a83a1f0000281100000221 Content-Length: 0 Call-ID: 1F1AB8A8-761A-4E01-9221-D9AB61AC8B76@10.10.10.7 CSeq: 191 OPTIONS From: ;tag=47973654421362 Max-Forwards: 70 To: Jul 3 01:26:56 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 0: OPTIONS sip:phone2.masterhost.ru SIP/2.0 (40) Jul 3 01:26:56 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 1: Via: SIP/2.0/UDP 10.10.10.7;rport;branch=z9hG4bK0a0a0a070000000b44a83a1f0000281100000221 (88) Jul 3 01:26:56 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 2: Content-Length: 0 (17) Jul 3 01:26:56 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 3: Call-ID: 1F1AB8A8-761A-4E01-9221-D9AB61AC8B76@10.10.10.7 (56) Jul 3 01:26:56 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 4: CSeq: 191 OPTIONS (17) Jul 3 01:26:56 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 5: From: ;tag=47973654421362 (56) Jul 3 01:26:56 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 6: Max-Forwards: 70 (16) Jul 3 01:26:56 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 7: To: (30) Jul 3 01:26:56 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 8: (0) --- (8 headers 0 lines)--- Jul 3 01:26:56 DEBUG[23515]: chan_sip.c:3144 sip_alloc: Allocating new SIP dialog for 1F1AB8A8-761A-4E01-9221-D9AB61AC8B76@10.10.10.7 - OPTIONS (No RTP) Jul 3 01:26:56 DEBUG[23515]: chan_sip.c:11140 handle_request: **** Received OPTIONS (3) - Command in SIP OPTIONS Looking for s in fromoutside (domain phone2.masterhost.ru) Transmitting (NAT) to 85.140.10.142:62443: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.10.10.7;branch=z9hG4bK0a0a0a070000000b44a83a1f0000281100000221;received=85.140.10.142;rport=62443 From: ;tag=47973654421362 To: ;tag=as0ca4459f Call-ID: 1F1AB8A8-761A-4E01-9221-D9AB61AC8B76@10.10.10.7 CSeq: 191 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Jul 3 01:26:56 DEBUG[23515]: chan_sip.c:11341 sipsock_read: SIP message could not be handled, bad request: 1F1AB8A8-761A-4E01-9221-D9AB61AC8B76@10.10.10.7 Destroying call '1F1AB8A8-761A-4E01-9221-D9AB61AC8B76@10.10.10.7' <-- SIP read from 85.140.10.142:62443: INVITE sip:9166104314@83.222.22.242 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.7;rport;branch=z9hG4bK0a0a0a070000007c44a83a210000302000000223 Content-Length: 211 Contact: Call-ID: 70bbb3cf70f2429302325f3b63f5e59b@masterhost.ru Content-Type: application/sdp CSeq: 3 INVITE From: "operator name";tag=47972382613179 Max-Forwards: 70 To: ;tag=as1249a25f User-Agent: SJphone/1.60.289a (SJ Labs) v=0 o=- 3360864403 3360864406 IN IP4 10.10.10.7 s=SJphone c=IN IP4 0.0.0.0 t=0 0 a=direction:active m=audio 49168 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11,16 Jul 3 01:26:57 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 0: INVITE sip:9166104314@83.222.22.242 SIP/2.0 (43) Jul 3 01:26:57 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 1: Via: SIP/2.0/UDP 10.10.10.7;rport;branch=z9hG4bK0a0a0a070000007c44a83a210000302000000223 (88) Jul 3 01:26:57 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 2: Content-Length: 211 (19) Jul 3 01:26:57 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 3: Contact: (35) Jul 3 01:26:57 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 4: Call-ID: 70bbb3cf70f2429302325f3b63f5e59b@masterhost.ru (55) Jul 3 01:26:57 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 5: Content-Type: application/sdp (29) Jul 3 01:26:57 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 6: CSeq: 3 INVITE (14) Jul 3 01:26:57 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 7: From: "operator name";tag=47972382613179 (68) Jul 3 01:26:57 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 8: Max-Forwards: 70 (16) Jul 3 01:26:57 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 9: To: ;tag=as1249a25f (49) Jul 3 01:26:57 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 10: User-Agent: SJphone/1.60.289a (SJ Labs) (39) Jul 3 01:26:57 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 11: (0) Jul 3 01:26:57 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: v=0 (3) Jul 3 01:26:57 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: o=- 3360864403 3360864406 IN IP4 10.10.10.7 (43) Jul 3 01:26:57 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: s=SJphone (9) Jul 3 01:26:57 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: c=IN IP4 0.0.0.0 (16) Jul 3 01:26:57 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: t=0 0 (5) Jul 3 01:26:57 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: a=direction:active (18) Jul 3 01:26:57 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: m=audio 49168 RTP/AVP 3 101 (27) Jul 3 01:26:57 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: a=rtpmap:3 GSM/8000 (19) Jul 3 01:26:57 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) Jul 3 01:26:57 DEBUG[23515]: chan_sip.c:3392 parse_request: Line: a=fmtp:101 0-11,16 (18) --- (11 headers 10 lines)--- Jul 3 01:26:57 DEBUG[23515]: chan_sip.c:11140 handle_request: **** Received INVITE (5) - Command in SIP INVITE Using INVITE request as basis request - 70bbb3cf70f2429302325f3b63f5e59b@masterhost.ru Sending to 10.10.10.7 : 5060 (NAT) Found RTP audio format 3 Found RTP audio format 101 Peer audio RTP is at port 0.0.0.0:49168 Jul 3 01:26:57 DEBUG[23515]: chan_sip.c:3601 process_sdp: Peer audio RTP is at port 0.0.0.0:49168 Peer video RTP is at port 0.0.0.0:65535 Jul 3 01:26:57 DEBUG[23515]: chan_sip.c:3625 process_sdp: Peer video RTP is at port 0.0.0.0:65535 Found description format GSM Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x2 (gsm)/video=0x0 (nothing), combined - 0x2 (gsm) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) -- Started music on hold, class 'default', on Zap/32-1 Jul 3 01:26:57 DEBUG[23515]: channel.c:1724 ast_settimeout: Scheduling timer at 160 sample intervals Jul 3 01:26:57 DEBUG[23515]: chan_sip.c:10553 handle_request_invite: Got a SIP re-invite for call 70bbb3cf70f2429302325f3b63f5e59b@masterhost.ru We're at 83.222.22.242 port 12820 Video is at 83.222.22.242 port 17500 Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 85.140.10.142:62443: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.7;branch=z9hG4bK0a0a0a070000007c44a83a210000302000000223;received=85.140.10.142;rport=62443 From: "operator name";tag=47972382613179 To: ;tag=as1249a25f Call-ID: 70bbb3cf70f2429302325f3b63f5e59b@masterhost.ru CSeq: 3 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 217 v=0 o=root 23506 23509 IN IP4 83.222.22.242 s=session c=IN IP4 83.222.22.242 t=0 0 m=audio 12820 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Jul 3 01:26:57 DEBUG[23515]: chan_sip.c:1293 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #192 Jul 3 01:26:57 DEBUG[23529]: channel.c:1988 ast_read: Generator got voice, switching to phase locked mode Jul 3 01:26:57 DEBUG[23529]: channel.c:1724 ast_settimeout: Scheduling timer at 0 sample intervals Jul 3 01:26:57 DEBUG[23529]: channel.c:2363 set_format: Set channel Zap/32-1 to write format ulaw Jul 3 01:26:57 DEBUG[23529]: res_musiconhold.c:233 ast_moh_files_next: Zap/32-1 Opened file 0 '/var/lib/asterisk/mohmp3/misc/We_Are_The_Champions_Queen' <-- SIP read from 10.15.4.244:5060: REGISTER sip:10.15.1.2 SIP/2.0 Via: SIP/2.0/UDP 10.15.4.244:5060;branch=z9hG4bK73a7988d5ae295b From: ;tag=4132175538 To: Call-ID: 2996840658@10.15.4.244 CSeq: 847 REGISTER Contact: ;expires=3600;+sip.instance="" User-Agent: Cisco-CP7912/8.0.0-060111A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces, 100rel Content-Length: 0 Jul 3 01:26:57 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 0: REGISTER sip:10.15.1.2 SIP/2.0 (30) Jul 3 01:26:57 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 1: Via: SIP/2.0/UDP 10.15.4.244:5060;branch=z9hG4bK73a7988d5ae295b (63) Jul 3 01:26:57 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 2: From: ;tag=4132175538 (52) Jul 3 01:26:57 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 3: To: (35) Jul 3 01:26:57 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 4: Call-ID: 2996840658@10.15.4.244 (31) Jul 3 01:26:57 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 5: CSeq: 847 REGISTER (18) Jul 3 01:26:57 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 6: Contact: ;expires=3600;+sip.instance="" (138) Jul 3 01:26:57 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 7: User-Agent: Cisco-CP7912/8.0.0-060111A (38) Jul 3 01:26:57 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 8: Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE (80) Jul 3 01:26:57 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 9: Supported: replaces, 100rel (27) Jul 3 01:26:57 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 10: Content-Length: 0 (17) Jul 3 01:26:57 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 11: (0) --- (11 headers 0 lines)--- Jul 3 01:26:57 DEBUG[23515]: chan_sip.c:3144 sip_alloc: Allocating new SIP dialog for 2996840658@10.15.4.244 - REGISTER (No RTP) Jul 3 01:26:57 DEBUG[23515]: chan_sip.c:11140 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER Using latest REGISTER request as basis request Sending to 10.15.4.244 : 5060 (NAT) Transmitting (NAT) to 10.15.4.244:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.15.4.244:5060;branch=z9hG4bK73a7988d5ae295b;received=10.15.4.244 From: ;tag=4132175538 To: Call-ID: 2996840658@10.15.4.244 CSeq: 847 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Transmitting (NAT) to 10.15.4.244:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.15.4.244:5060;branch=z9hG4bK73a7988d5ae295b;received=10.15.4.244 From: ;tag=4132175538 To: ;tag=as1d68a09f Call-ID: 2996840658@10.15.4.244 CSeq: 847 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: WWW-Authenticate: Digest realm="phone1.masterhost.ru", nonce="1d6e21b4" Content-Length: 0 --- Scheduling destruction of call '2996840658@10.15.4.244' in 15000 ms <-- SIP read from 10.15.4.244:5060: REGISTER sip:10.15.1.2 SIP/2.0 Via: SIP/2.0/UDP 10.15.4.244:5060;branch=z9hG4bKef8767f964fbf5b0 From: ;tag=4132175538 To: Call-ID: 2996840658@10.15.4.244 CSeq: 848 REGISTER Contact: ;expires=3600;+sip.instance="" User-Agent: Cisco-CP7912/8.0.0-060111A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces, 100rel Authorization: Digest username="1647",realm="phone1.masterhost.ru",nonce="1d6e21b4",uri="sip:10.15.1.2",response="***" Content-Length: 0 Jul 3 01:26:58 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 0: REGISTER sip:10.15.1.2 SIP/2.0 (30) Jul 3 01:26:58 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 1: Via: SIP/2.0/UDP 10.15.4.244:5060;branch=z9hG4bKef8767f964fbf5b0 (64) Jul 3 01:26:58 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 2: From: ;tag=4132175538 (52) Jul 3 01:26:58 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 3: To: (35) Jul 3 01:26:58 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 4: Call-ID: 2996840658@10.15.4.244 (31) Jul 3 01:26:58 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 5: CSeq: 848 REGISTER (18) Jul 3 01:26:58 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 6: Contact: ;expires=3600;+sip.instance="" (138) Jul 3 01:26:58 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 7: User-Agent: Cisco-CP7912/8.0.0-060111A (38) Jul 3 01:26:58 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 8: Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE (80) Jul 3 01:26:58 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 9: Supported: replaces, 100rel (27) Jul 3 01:26:58 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 10: Authorization: Digest username="1647",realm="phone1.masterhost.ru",nonce="1d6e21b4",uri="sip:10.15.1.2",response="***" (147) Jul 3 01:26:58 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 11: Content-Length: 0 (17) Jul 3 01:26:58 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 12: (0) --- (12 headers 0 lines)--- Jul 3 01:26:58 DEBUG[23515]: chan_sip.c:11140 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER Using latest REGISTER request as basis request Sending to 10.15.4.244 : 5060 (NAT) Transmitting (NAT) to 10.15.4.244:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.15.4.244:5060;branch=z9hG4bKef8767f964fbf5b0;received=10.15.4.244 From: ;tag=4132175538 To: Call-ID: 2996840658@10.15.4.244 CSeq: 848 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- -- Saved useragent "Cisco-CP7912/8.0.0-060111A" for peer 1647 Transmitting (NAT) to 10.15.4.244:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.15.4.244:5060;branch=z9hG4bKef8767f964fbf5b0;received=10.15.4.244 From: ;tag=4132175538 To: ;tag=as1d68a09f Call-ID: 2996840658@10.15.4.244 CSeq: 848 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Expires: 3600 Contact: ;expires=3600 Date: Sun, 02 Jul 2006 21:26:58 GMT Content-Length: 0 --- Scheduling destruction of call '2996840658@10.15.4.244' in 15000 ms Jul 3 01:26:58 DEBUG[23510]: chan_sip.c:11671 sip_devicestate: Checking device state for peer 1647 Jul 3 01:26:58 DEBUG[23510]: devicestate.c:187 do_state_change: Changing state for SIP/1647 - state 1 (Not in use) Jul 3 01:26:58 DEBUG[23548]: app_queue.c:490 changethread: Device 'SIP/1647' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. <-- SIP read from 85.140.10.142:62443: ACK sip:9166104314@83.222.22.242 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.7;rport;branch=z9hG4bK0a0a0a070000007c44a83a2100002e5e00000225 Content-Length: 0 Call-ID: 70bbb3cf70f2429302325f3b63f5e59b@masterhost.ru CSeq: 3 ACK From: "operator name";tag=47972382613179 Max-Forwards: 70 To: ;tag=as1249a25f User-Agent: SJphone/1.60.289a (SJ Labs) Jul 3 01:26:58 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 0: ACK sip:9166104314@83.222.22.242 SIP/2.0 (40) Jul 3 01:26:58 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 1: Via: SIP/2.0/UDP 10.10.10.7;rport;branch=z9hG4bK0a0a0a070000007c44a83a2100002e5e00000225 (88) Jul 3 01:26:58 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 2: Content-Length: 0 (17) Jul 3 01:26:58 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 3: Call-ID: 70bbb3cf70f2429302325f3b63f5e59b@masterhost.ru (55) Jul 3 01:26:58 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 4: CSeq: 3 ACK (11) Jul 3 01:26:58 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 5: From: "operator name";tag=47972382613179 (68) Jul 3 01:26:58 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 6: Max-Forwards: 70 (16) Jul 3 01:26:58 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 7: To: ;tag=as1249a25f (49) Jul 3 01:26:58 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 8: User-Agent: SJphone/1.60.289a (SJ Labs) (39) Jul 3 01:26:58 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 9: (0) --- (9 headers 0 lines)--- Jul 3 01:26:58 DEBUG[23515]: chan_sip.c:11140 handle_request: **** Received ACK (6) - Command in SIP ACK Jul 3 01:26:58 DEBUG[23515]: chan_sip.c:1390 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #192 Jul 3 01:26:58 DEBUG[23515]: chan_sip.c:1401 __sip_ack: Stopping retransmission on '70bbb3cf70f2429302325f3b63f5e59b@masterhost.ru' of Response 3: Match Found <-- SIP read from 85.140.10.142:62443: REFER sip:9166104314@83.222.22.242 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.7;rport;branch=z9hG4bK0a0a0a070000007c44a83a21000037d300000226 Content-Length: 0 Contact: Call-ID: 70bbb3cf70f2429302325f3b63f5e59b@masterhost.ru CSeq: 4 REFER From: "operator name";tag=47972382613179 Max-Forwards: 70 Referred-By: Refer-To: To: ;tag=as1249a25f User-Agent: SJphone/1.60.289a (SJ Labs) Jul 3 01:26:58 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 0: REFER sip:9166104314@83.222.22.242 SIP/2.0 (42) Jul 3 01:26:58 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 1: Via: SIP/2.0/UDP 10.10.10.7;rport;branch=z9hG4bK0a0a0a070000007c44a83a21000037d300000226 (88) Jul 3 01:26:58 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 2: Content-Length: 0 (17) Jul 3 01:26:58 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 3: Contact: (35) Jul 3 01:26:58 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 4: Call-ID: 70bbb3cf70f2429302325f3b63f5e59b@masterhost.ru (55) Jul 3 01:26:58 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 5: CSeq: 4 REFER (13) Jul 3 01:26:58 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 6: From: "operator name";tag=47972382613179 (68) Jul 3 01:26:58 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 7: Max-Forwards: 70 (16) Jul 3 01:26:58 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 8: Referred-By: (39) Jul 3 01:26:58 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 9: Refer-To: (148) Jul 3 01:26:58 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 10: To: ;tag=as1249a25f (49) Jul 3 01:26:58 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 11: User-Agent: SJphone/1.60.289a (SJ Labs) (39) Jul 3 01:26:58 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 12: (0) --- (12 headers 0 lines)--- Jul 3 01:26:58 DEBUG[23515]: chan_sip.c:11140 handle_request: **** Received REFER (9) - Command in SIP REFER Jul 3 01:26:58 DEBUG[23515]: chan_sip.c:10665 handle_request_refer: SIP call transfer received for call 70bbb3cf70f2429302325f3b63f5e59b@masterhost.ru (REFER)! Transfer to 9166104314 in fromoffice Transfer from 1211 in fromoffice Jul 3 01:26:58 DEBUG[23515]: chan_sip.c:6854 get_refer_info: Assigning Replace-Call-ID Info 70bbb3cf70f2429302325f3b63f5e59b@masterhost.ru to REPLACE_CALL_ID Jul 3 01:26:58 NOTICE[23515]: chan_sip.c:6863 get_refer_info: Supervised transfer attempted to transfer into same call id (70bbb3cf70f2429302325f3b63f5e59b@masterhost.ru == 70bbb3cf70f2429302325f3b63f5e59b@masterhost.ru)! Transmitting (NAT) to 85.140.10.142:62443: SIP/2.0 603 Declined Via: SIP/2.0/UDP 10.10.10.7;branch=z9hG4bK0a0a0a070000007c44a83a21000037d300000226;received=85.140.10.142;rport=62443 From: "operator name";tag=47972382613179 To: ;tag=as1249a25f Call-ID: 70bbb3cf70f2429302325f3b63f5e59b@masterhost.ru CSeq: 4 REFER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Jul 3 01:26:58 DEBUG[23515]: chan_sip.c:11341 sipsock_read: SIP message could not be handled, bad request: 70bbb3cf70f2429302325f3b63f5e59b@masterhost.ru <-- SIP read from 10.10.11.201:5060: OPTIONS sip:10.10.11.61 SIP/2.0 Via: SIP/2.0/UDP 10.10.11.201;rport;branch=z9hG4bK0a0a0bc90000001044a83a2500006de400011809 Content-Length: 0 Call-ID: 75817351-940B-4522-A96B-ABF0B97EDC36@10.10.11.201 CSeq: 23476 OPTIONS From: ;tag=80417003522685 Max-Forwards: 70 To: Jul 3 01:27:01 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 0: OPTIONS sip:10.10.11.61 SIP/2.0 (31) Jul 3 01:27:01 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 1: Via: SIP/2.0/UDP 10.10.11.201;rport;branch=z9hG4bK0a0a0bc90000001044a83a2500006de400011809 (90) Jul 3 01:27:01 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 2: Content-Length: 0 (17) Jul 3 01:27:01 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 3: Call-ID: 75817351-940B-4522-A96B-ABF0B97EDC36@10.10.11.201 (58) Jul 3 01:27:01 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 4: CSeq: 23476 OPTIONS (19) Jul 3 01:27:01 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 5: From: ;tag=80417003522685 (47) Jul 3 01:27:01 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 6: Max-Forwards: 70 (16) Jul 3 01:27:01 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 7: To: (21) Jul 3 01:27:01 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 8: (0) --- (8 headers 0 lines)--- Jul 3 01:27:01 DEBUG[23515]: chan_sip.c:3144 sip_alloc: Allocating new SIP dialog for 75817351-940B-4522-A96B-ABF0B97EDC36@10.10.11.201 - OPTIONS (No RTP) Jul 3 01:27:01 DEBUG[23515]: chan_sip.c:11140 handle_request: **** Received OPTIONS (3) - Command in SIP OPTIONS Looking for s in fromoutside (domain 10.10.11.61) Transmitting (NAT) to 10.10.11.201:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.10.11.201;branch=z9hG4bK0a0a0bc90000001044a83a2500006de400011809;received=10.10.11.201;rport=5060 From: ;tag=80417003522685 To: ;tag=as102dc142 Call-ID: 75817351-940B-4522-A96B-ABF0B97EDC36@10.10.11.201 CSeq: 23476 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Jul 3 01:27:01 DEBUG[23515]: chan_sip.c:11341 sipsock_read: SIP message could not be handled, bad request: 75817351-940B-4522-A96B-ABF0B97EDC36@10.10.11.201 Destroying call '75817351-940B-4522-A96B-ABF0B97EDC36@10.10.11.201' phone2*CLI> <-- SIP read from 10.10.11.200:5060: OPTIONS sip:10.10.11.61 SIP/2.0 Via: SIP/2.0/UDP 10.10.11.200;rport;branch=z9hG4bK0a0a0bc80000001044a83a2e00007b2e000008db Content-Length: 0 Call-ID: 4E03B055-DDE1-488D-82FA-B9555A34F797@10.10.11.200 CSeq: 760 OPTIONS From: ;tag=6702214026548 Max-Forwards: 70 To: Jul 3 01:27:09 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 0: OPTIONS sip:10.10.11.61 SIP/2.0 (31) Jul 3 01:27:09 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 1: Via: SIP/2.0/UDP 10.10.11.200;rport;branch=z9hG4bK0a0a0bc80000001044a83a2e00007b2e000008db (90) Jul 3 01:27:09 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 2: Content-Length: 0 (17) Jul 3 01:27:09 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 3: Call-ID: 4E03B055-DDE1-488D-82FA-B9555A34F797@10.10.11.200 (58) Jul 3 01:27:09 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 4: CSeq: 760 OPTIONS (17) Jul 3 01:27:09 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 5: From: ;tag=6702214026548 (46) Jul 3 01:27:09 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 6: Max-Forwards: 70 (16) Jul 3 01:27:09 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 7: To: (21) Jul 3 01:27:09 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 8: (0) --- (8 headers 0 lines)--- Jul 3 01:27:09 DEBUG[23515]: chan_sip.c:3144 sip_alloc: Allocating new SIP dialog for 4E03B055-DDE1-488D-82FA-B9555A34F797@10.10.11.200 - OPTIONS (No RTP) Jul 3 01:27:09 DEBUG[23515]: chan_sip.c:11140 handle_request: **** Received OPTIONS (3) - Command in SIP OPTIONS Looking for s in fromoutside (domain 10.10.11.61) Transmitting (NAT) to 10.10.11.200:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.10.11.200;branch=z9hG4bK0a0a0bc80000001044a83a2e00007b2e000008db;received=10.10.11.200;rport=5060 From: ;tag=6702214026548 To: ;tag=as6546ed4c Call-ID: 4E03B055-DDE1-488D-82FA-B9555A34F797@10.10.11.200 CSeq: 760 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Jul 3 01:27:09 DEBUG[23515]: chan_sip.c:11341 sipsock_read: SIP message could not be handled, bad request: 4E03B055-DDE1-488D-82FA-B9555A34F797@10.10.11.200 Destroying call '4E03B055-DDE1-488D-82FA-B9555A34F797@10.10.11.200' <-- SIP read from 195.14.53.142:5060: OPTIONS sip:phone2.masterhost.ru SIP/2.0 Via: SIP/2.0/UDP 192.168.200.218;rport;branch=z9hG4bKc0a8c8da0000001044a83a27000054b400001fd0 Content-Length: 0 Call-ID: 169469BD-4B0D-4AEC-B64B-77E6BB62D403@192.168.200.218 CSeq: 4244 OPTIONS From: ;tag=22467925014080 Max-Forwards: 70 To: Jul 3 01:27:12 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 0: OPTIONS sip:phone2.masterhost.ru SIP/2.0 (40) Jul 3 01:27:12 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.200.218;rport;branch=z9hG4bKc0a8c8da0000001044a83a27000054b400001fd0 (93) Jul 3 01:27:12 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 2: Content-Length: 0 (17) Jul 3 01:27:12 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 3: Call-ID: 169469BD-4B0D-4AEC-B64B-77E6BB62D403@192.168.200.218 (61) Jul 3 01:27:12 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 4: CSeq: 4244 OPTIONS (18) Jul 3 01:27:12 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 5: From: ;tag=22467925014080 (56) Jul 3 01:27:12 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 6: Max-Forwards: 70 (16) Jul 3 01:27:12 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 7: To: (30) Jul 3 01:27:12 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 8: (0) --- (8 headers 0 lines)--- Jul 3 01:27:12 DEBUG[23515]: chan_sip.c:3144 sip_alloc: Allocating new SIP dialog for 169469BD-4B0D-4AEC-B64B-77E6BB62D403@192.168.200.218 - OPTIONS (No RTP) Jul 3 01:27:12 DEBUG[23515]: chan_sip.c:11140 handle_request: **** Received OPTIONS (3) - Command in SIP OPTIONS Looking for s in fromoutside (domain phone2.masterhost.ru) Transmitting (NAT) to 195.14.53.142:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.200.218;branch=z9hG4bKc0a8c8da0000001044a83a27000054b400001fd0;received=195.14.53.142;rport=5060 From: ;tag=22467925014080 To: ;tag=as393f7070 Call-ID: 169469BD-4B0D-4AEC-B64B-77E6BB62D403@192.168.200.218 CSeq: 4244 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Jul 3 01:27:12 DEBUG[23515]: chan_sip.c:11341 sipsock_read: SIP message could not be handled, bad request: 169469BD-4B0D-4AEC-B64B-77E6BB62D403@192.168.200.218 Destroying call '169469BD-4B0D-4AEC-B64B-77E6BB62D403@192.168.200.218' Jul 3 01:27:13 DEBUG[23515]: chan_sip.c:1323 __sip_autodestruct: Auto destroying call '2996840658@10.15.4.244' Destroying call '2996840658@10.15.4.244' <-- SIP read from 10.10.11.134:5060: OPTIONS sip:10.10.11.61 SIP/2.0 Via: SIP/2.0/UDP 10.10.11.134;rport;branch=z9hG4bK0a0a0b860000001044a83a330000165c0001096b Content-Length: 0 Call-ID: 23D02ACD-0BD6-43A6-B6CB-DF5066A7D3AA@10.10.11.134 CSeq: 28474 OPTIONS From: ;tag=8035353123807 Max-Forwards: 70 To: Jul 3 01:27:15 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 0: OPTIONS sip:10.10.11.61 SIP/2.0 (31) Jul 3 01:27:15 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 1: Via: SIP/2.0/UDP 10.10.11.134;rport;branch=z9hG4bK0a0a0b860000001044a83a330000165c0001096b (90) Jul 3 01:27:15 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 2: Content-Length: 0 (17) Jul 3 01:27:15 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 3: Call-ID: 23D02ACD-0BD6-43A6-B6CB-DF5066A7D3AA@10.10.11.134 (58) Jul 3 01:27:15 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 4: CSeq: 28474 OPTIONS (19) Jul 3 01:27:15 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 5: From: ;tag=8035353123807 (46) Jul 3 01:27:15 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 6: Max-Forwards: 70 (16) Jul 3 01:27:15 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 7: To: (21) Jul 3 01:27:15 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 8: (0) --- (8 headers 0 lines)--- Jul 3 01:27:15 DEBUG[23515]: chan_sip.c:3144 sip_alloc: Allocating new SIP dialog for 23D02ACD-0BD6-43A6-B6CB-DF5066A7D3AA@10.10.11.134 - OPTIONS (No RTP) Jul 3 01:27:15 DEBUG[23515]: chan_sip.c:11140 handle_request: **** Received OPTIONS (3) - Command in SIP OPTIONS Looking for s in fromoutside (domain 10.10.11.61) Transmitting (NAT) to 10.10.11.134:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.10.11.134;branch=z9hG4bK0a0a0b860000001044a83a330000165c0001096b;received=10.10.11.134;rport=5060 From: ;tag=8035353123807 To: ;tag=as176a4966 Call-ID: 23D02ACD-0BD6-43A6-B6CB-DF5066A7D3AA@10.10.11.134 CSeq: 28474 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Jul 3 01:27:15 DEBUG[23515]: chan_sip.c:11341 sipsock_read: SIP message could not be handled, bad request: 23D02ACD-0BD6-43A6-B6CB-DF5066A7D3AA@10.10.11.134 Destroying call '23D02ACD-0BD6-43A6-B6CB-DF5066A7D3AA@10.10.11.134' <-- SIP read from 85.140.10.142:62443: OPTIONS sip:phone2.masterhost.ru SIP/2.0 Via: SIP/2.0/UDP 10.10.10.7;rport;branch=z9hG4bK0a0a0a070000000b44a83a33000021cb00000228 Content-Length: 0 Call-ID: 1F1AB8A8-761A-4E01-9221-D9AB61AC8B76@10.10.10.7 CSeq: 192 OPTIONS From: ;tag=47975655330873 Max-Forwards: 70 To: Jul 3 01:27:16 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 0: OPTIONS sip:phone2.masterhost.ru SIP/2.0 (40) Jul 3 01:27:16 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 1: Via: SIP/2.0/UDP 10.10.10.7;rport;branch=z9hG4bK0a0a0a070000000b44a83a33000021cb00000228 (88) Jul 3 01:27:16 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 2: Content-Length: 0 (17) Jul 3 01:27:16 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 3: Call-ID: 1F1AB8A8-761A-4E01-9221-D9AB61AC8B76@10.10.10.7 (56) Jul 3 01:27:16 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 4: CSeq: 192 OPTIONS (17) Jul 3 01:27:16 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 5: From: ;tag=47975655330873 (56) Jul 3 01:27:16 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 6: Max-Forwards: 70 (16) Jul 3 01:27:16 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 7: To: (30) Jul 3 01:27:16 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 8: (0) --- (8 headers 0 lines)--- Jul 3 01:27:16 DEBUG[23515]: chan_sip.c:3144 sip_alloc: Allocating new SIP dialog for 1F1AB8A8-761A-4E01-9221-D9AB61AC8B76@10.10.10.7 - OPTIONS (No RTP) Jul 3 01:27:16 DEBUG[23515]: chan_sip.c:11140 handle_request: **** Received OPTIONS (3) - Command in SIP OPTIONS Looking for s in fromoutside (domain phone2.masterhost.ru) Transmitting (NAT) to 85.140.10.142:62443: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.10.10.7;branch=z9hG4bK0a0a0a070000000b44a83a33000021cb00000228;received=85.140.10.142;rport=62443 From: ;tag=47975655330873 To: ;tag=as4a2a4a1a Call-ID: 1F1AB8A8-761A-4E01-9221-D9AB61AC8B76@10.10.10.7 CSeq: 192 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Jul 3 01:27:16 DEBUG[23515]: chan_sip.c:11341 sipsock_read: SIP message could not be handled, bad request: 1F1AB8A8-761A-4E01-9221-D9AB61AC8B76@10.10.10.7 Destroying call '1F1AB8A8-761A-4E01-9221-D9AB61AC8B76@10.10.10.7' <-- SIP read from 10.10.11.201:5060: OPTIONS sip:10.10.11.61 SIP/2.0 Via: SIP/2.0/UDP 10.10.11.201;rport;branch=z9hG4bK0a0a0bc90000001044a83a39000069460001180b Content-Length: 0 Call-ID: 75817351-940B-4522-A96B-ABF0B97EDC36@10.10.11.201 CSeq: 23477 OPTIONS From: ;tag=80419004411304 Max-Forwards: 70 To: Jul 3 01:27:21 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 0: OPTIONS sip:10.10.11.61 SIP/2.0 (31) Jul 3 01:27:21 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 1: Via: SIP/2.0/UDP 10.10.11.201;rport;branch=z9hG4bK0a0a0bc90000001044a83a39000069460001180b (90) Jul 3 01:27:21 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 2: Content-Length: 0 (17) Jul 3 01:27:21 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 3: Call-ID: 75817351-940B-4522-A96B-ABF0B97EDC36@10.10.11.201 (58) Jul 3 01:27:21 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 4: CSeq: 23477 OPTIONS (19) Jul 3 01:27:21 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 5: From: ;tag=80419004411304 (47) Jul 3 01:27:21 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 6: Max-Forwards: 70 (16) Jul 3 01:27:21 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 7: To: (21) Jul 3 01:27:21 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 8: (0) --- (8 headers 0 lines)--- Jul 3 01:27:21 DEBUG[23515]: chan_sip.c:3144 sip_alloc: Allocating new SIP dialog for 75817351-940B-4522-A96B-ABF0B97EDC36@10.10.11.201 - OPTIONS (No RTP) Jul 3 01:27:21 DEBUG[23515]: chan_sip.c:11140 handle_request: **** Received OPTIONS (3) - Command in SIP OPTIONS Looking for s in fromoutside (domain 10.10.11.61) Transmitting (NAT) to 10.10.11.201:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.10.11.201;branch=z9hG4bK0a0a0bc90000001044a83a39000069460001180b;received=10.10.11.201;rport=5060 From: ;tag=80419004411304 To: ;tag=as7c79fde8 Call-ID: 75817351-940B-4522-A96B-ABF0B97EDC36@10.10.11.201 CSeq: 23477 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Jul 3 01:27:21 DEBUG[23515]: chan_sip.c:11341 sipsock_read: SIP message could not be handled, bad request: 75817351-940B-4522-A96B-ABF0B97EDC36@10.10.11.201 Destroying call '75817351-940B-4522-A96B-ABF0B97EDC36@10.10.11.201' <-- SIP read from 10.10.11.145:5060: OPTIONS sip:10.10.11.61 SIP/2.0 Via: SIP/2.0/UDP 10.10.11.145;rport;branch=z9hG4bK0a0a0b910000001044a83a3e0000247b00002ad3 Content-Length: 0 Call-ID: A03FC774-DC9D-472A-BCBE-388C15D77FF5@10.10.11.145 CSeq: 4300 OPTIONS From: ;tag=1292782188619 Max-Forwards: 70 To: Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 0: OPTIONS sip:10.10.11.61 SIP/2.0 (31) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 1: Via: SIP/2.0/UDP 10.10.11.145;rport;branch=z9hG4bK0a0a0b910000001044a83a3e0000247b00002ad3 (90) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 2: Content-Length: 0 (17) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 3: Call-ID: A03FC774-DC9D-472A-BCBE-388C15D77FF5@10.10.11.145 (58) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 4: CSeq: 4300 OPTIONS (18) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 5: From: ;tag=1292782188619 (46) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 6: Max-Forwards: 70 (16) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 7: To: (21) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 8: (0) --- (8 headers 0 lines)--- Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3144 sip_alloc: Allocating new SIP dialog for A03FC774-DC9D-472A-BCBE-388C15D77FF5@10.10.11.145 - OPTIONS (No RTP) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:11140 handle_request: **** Received OPTIONS (3) - Command in SIP OPTIONS Looking for s in fromoutside (domain 10.10.11.61) Transmitting (NAT) to 10.10.11.145:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.10.11.145;branch=z9hG4bK0a0a0b910000001044a83a3e0000247b00002ad3;received=10.10.11.145;rport=5060 From: ;tag=1292782188619 To: ;tag=as369dd5b3 Call-ID: A03FC774-DC9D-472A-BCBE-388C15D77FF5@10.10.11.145 CSeq: 4300 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:11341 sipsock_read: SIP message could not be handled, bad request: A03FC774-DC9D-472A-BCBE-388C15D77FF5@10.10.11.145 Destroying call 'A03FC774-DC9D-472A-BCBE-388C15D77FF5@10.10.11.145' <-- SIP read from 10.10.11.200:5060: OPTIONS sip:10.10.11.61 SIP/2.0 Via: SIP/2.0/UDP 10.10.11.200;rport;branch=z9hG4bK0a0a0bc80000001044a83a4200005dfb000008dd Content-Length: 0 Call-ID: 4E03B055-DDE1-488D-82FA-B9555A34F797@10.10.11.200 CSeq: 761 OPTIONS From: ;tag=670421402573 Max-Forwards: 70 To: Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 0: OPTIONS sip:10.10.11.61 SIP/2.0 (31) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 1: Via: SIP/2.0/UDP 10.10.11.200;rport;branch=z9hG4bK0a0a0bc80000001044a83a4200005dfb000008dd (90) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 2: Content-Length: 0 (17) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 3: Call-ID: 4E03B055-DDE1-488D-82FA-B9555A34F797@10.10.11.200 (58) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 4: CSeq: 761 OPTIONS (17) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 5: From: ;tag=670421402573 (45) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 6: Max-Forwards: 70 (16) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 7: To: (21) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 8: (0) --- (8 headers 0 lines)--- Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3144 sip_alloc: Allocating new SIP dialog for 4E03B055-DDE1-488D-82FA-B9555A34F797@10.10.11.200 - OPTIONS (No RTP) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:11140 handle_request: **** Received OPTIONS (3) - Command in SIP OPTIONS Looking for s in fromoutside (domain 10.10.11.61) Transmitting (NAT) to 10.10.11.200:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.10.11.200;branch=z9hG4bK0a0a0bc80000001044a83a4200005dfb000008dd;received=10.10.11.200;rport=5060 From: ;tag=670421402573 To: ;tag=as184bcc15 Call-ID: 4E03B055-DDE1-488D-82FA-B9555A34F797@10.10.11.200 CSeq: 761 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:11341 sipsock_read: SIP message could not be handled, bad request: 4E03B055-DDE1-488D-82FA-B9555A34F797@10.10.11.200 Destroying call '4E03B055-DDE1-488D-82FA-B9555A34F797@10.10.11.200' <-- SIP read from 195.14.53.142:5060: OPTIONS sip:phone2.masterhost.ru SIP/2.0 Via: SIP/2.0/UDP 192.168.200.218;rport;branch=z9hG4bKc0a8c8da0000001044a83a3b00007de000001fd2 Content-Length: 0 Call-ID: 169469BD-4B0D-4AEC-B64B-77E6BB62D403@192.168.200.218 CSeq: 4245 OPTIONS From: ;tag=22469925024932 Max-Forwards: 70 To: Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 0: OPTIONS sip:phone2.masterhost.ru SIP/2.0 (40) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.200.218;rport;branch=z9hG4bKc0a8c8da0000001044a83a3b00007de000001fd2 (93) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 2: Content-Length: 0 (17) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 3: Call-ID: 169469BD-4B0D-4AEC-B64B-77E6BB62D403@192.168.200.218 (61) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 4: CSeq: 4245 OPTIONS (18) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 5: From: ;tag=22469925024932 (56) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 6: Max-Forwards: 70 (16) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 7: To: (30) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 8: (0) --- (8 headers 0 lines)--- Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3144 sip_alloc: Allocating new SIP dialog for 169469BD-4B0D-4AEC-B64B-77E6BB62D403@192.168.200.218 - OPTIONS (No RTP) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:11140 handle_request: **** Received OPTIONS (3) - Command in SIP OPTIONS Looking for s in fromoutside (domain phone2.masterhost.ru) Transmitting (NAT) to 195.14.53.142:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.200.218;branch=z9hG4bKc0a8c8da0000001044a83a3b00007de000001fd2;received=195.14.53.142;rport=5060 From: ;tag=22469925024932 To: ;tag=as7e7410ec Call-ID: 169469BD-4B0D-4AEC-B64B-77E6BB62D403@192.168.200.218 CSeq: 4245 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:11341 sipsock_read: SIP message could not be handled, bad request: 169469BD-4B0D-4AEC-B64B-77E6BB62D403@192.168.200.218 Destroying call '169469BD-4B0D-4AEC-B64B-77E6BB62D403@192.168.200.218' <-- SIP read from 10.10.11.134:5060: OPTIONS sip:10.10.11.61 SIP/2.0 Via: SIP/2.0/UDP 10.10.11.134;rport;branch=z9hG4bK0a0a0b860000001044a83a4700006ac50001096d Content-Length: 0 Call-ID: 23D02ACD-0BD6-43A6-B6CB-DF5066A7D3AA@10.10.11.134 CSeq: 28475 OPTIONS From: ;tag=80355531231355 Max-Forwards: 70 To: Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 0: OPTIONS sip:10.10.11.61 SIP/2.0 (31) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 1: Via: SIP/2.0/UDP 10.10.11.134;rport;branch=z9hG4bK0a0a0b860000001044a83a4700006ac50001096d (90) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 2: Content-Length: 0 (17) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 3: Call-ID: 23D02ACD-0BD6-43A6-B6CB-DF5066A7D3AA@10.10.11.134 (58) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 4: CSeq: 28475 OPTIONS (19) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 5: From: ;tag=80355531231355 (47) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 6: Max-Forwards: 70 (16) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 7: To: (21) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 8: (0) --- (8 headers 0 lines)--- Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3144 sip_alloc: Allocating new SIP dialog for 23D02ACD-0BD6-43A6-B6CB-DF5066A7D3AA@10.10.11.134 - OPTIONS (No RTP) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:11140 handle_request: **** Received OPTIONS (3) - Command in SIP OPTIONS Looking for s in fromoutside (domain 10.10.11.61) Transmitting (NAT) to 10.10.11.134:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.10.11.134;branch=z9hG4bK0a0a0b860000001044a83a4700006ac50001096d;received=10.10.11.134;rport=5060 From: ;tag=80355531231355 To: ;tag=as4e5f35f8 Call-ID: 23D02ACD-0BD6-43A6-B6CB-DF5066A7D3AA@10.10.11.134 CSeq: 28475 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:11341 sipsock_read: SIP message could not be handled, bad request: 23D02ACD-0BD6-43A6-B6CB-DF5066A7D3AA@10.10.11.134 Destroying call '23D02ACD-0BD6-43A6-B6CB-DF5066A7D3AA@10.10.11.134' <-- SIP read from 85.140.10.142:62443: OPTIONS sip:phone2.masterhost.ru SIP/2.0 Via: SIP/2.0/UDP 10.10.10.7;rport;branch=z9hG4bK0a0a0a070000000b44a83a4700006c320000022a Content-Length: 0 Call-ID: 1F1AB8A8-761A-4E01-9221-D9AB61AC8B76@10.10.10.7 CSeq: 193 OPTIONS From: ;tag=4797765626873 Max-Forwards: 70 To: Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 0: OPTIONS sip:phone2.masterhost.ru SIP/2.0 (40) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 1: Via: SIP/2.0/UDP 10.10.10.7;rport;branch=z9hG4bK0a0a0a070000000b44a83a4700006c320000022a (88) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 2: Content-Length: 0 (17) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 3: Call-ID: 1F1AB8A8-761A-4E01-9221-D9AB61AC8B76@10.10.10.7 (56) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 4: CSeq: 193 OPTIONS (17) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 5: From: ;tag=4797765626873 (55) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 6: Max-Forwards: 70 (16) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 7: To: (30) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 8: (0) --- (8 headers 0 lines)--- Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3144 sip_alloc: Allocating new SIP dialog for 1F1AB8A8-761A-4E01-9221-D9AB61AC8B76@10.10.10.7 - OPTIONS (No RTP) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:11140 handle_request: **** Received OPTIONS (3) - Command in SIP OPTIONS Looking for s in fromoutside (domain phone2.masterhost.ru) Transmitting (NAT) to 85.140.10.142:62443: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.10.10.7;branch=z9hG4bK0a0a0a070000000b44a83a4700006c320000022a;received=85.140.10.142;rport=62443 From: ;tag=4797765626873 To: ;tag=as075a7f90 Call-ID: 1F1AB8A8-761A-4E01-9221-D9AB61AC8B76@10.10.10.7 CSeq: 193 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:11341 sipsock_read: SIP message could not be handled, bad request: 1F1AB8A8-761A-4E01-9221-D9AB61AC8B76@10.10.10.7 Destroying call '1F1AB8A8-761A-4E01-9221-D9AB61AC8B76@10.10.10.7' <-- SIP read from 10.10.11.201:5060: OPTIONS sip:10.10.11.61 SIP/2.0 Via: SIP/2.0/UDP 10.10.11.201;rport;branch=z9hG4bK0a0a0bc90000001044a83a4d0000534a0001180d Content-Length: 0 Call-ID: 75817351-940B-4522-A96B-ABF0B97EDC36@10.10.11.201 CSeq: 23478 OPTIONS From: ;tag=80421005213209 Max-Forwards: 70 To: Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 0: OPTIONS sip:10.10.11.61 SIP/2.0 (31) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 1: Via: SIP/2.0/UDP 10.10.11.201;rport;branch=z9hG4bK0a0a0bc90000001044a83a4d0000534a0001180d (90) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 2: Content-Length: 0 (17) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 3: Call-ID: 75817351-940B-4522-A96B-ABF0B97EDC36@10.10.11.201 (58) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 4: CSeq: 23478 OPTIONS (19) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 5: From: ;tag=80421005213209 (47) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 6: Max-Forwards: 70 (16) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 7: To: (21) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 8: (0) --- (8 headers 0 lines)--- Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3144 sip_alloc: Allocating new SIP dialog for 75817351-940B-4522-A96B-ABF0B97EDC36@10.10.11.201 - OPTIONS (No RTP) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:11140 handle_request: **** Received OPTIONS (3) - Command in SIP OPTIONS Looking for s in fromoutside (domain 10.10.11.61) Transmitting (NAT) to 10.10.11.201:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.10.11.201;branch=z9hG4bK0a0a0bc90000001044a83a4d0000534a0001180d;received=10.10.11.201;rport=5060 From: ;tag=80421005213209 To: ;tag=as020ce5d4 Call-ID: 75817351-940B-4522-A96B-ABF0B97EDC36@10.10.11.201 CSeq: 23478 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:11341 sipsock_read: SIP message could not be handled, bad request: 75817351-940B-4522-A96B-ABF0B97EDC36@10.10.11.201 Destroying call '75817351-940B-4522-A96B-ABF0B97EDC36@10.10.11.201' <-- SIP read from 10.15.6.236:5060: REGISTER sip:10.15.1.2 SIP/2.0 Via: SIP/2.0/UDP 10.15.6.236:5060;branch=z9hG4bK7e4c7ab3cf518b70 From: ;tag=442590969 To: Call-ID: 363386910@10.15.6.236 CSeq: 2555 REGISTER Contact: ;expires=3600;+sip.instance="" User-Agent: Cisco-CP7912/8.0.0-060111A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces, 100rel Content-Length: 0 Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 0: REGISTER sip:10.15.1.2 SIP/2.0 (30) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 1: Via: SIP/2.0/UDP 10.15.6.236:5060;branch=z9hG4bK7e4c7ab3cf518b70 (64) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 2: From: ;tag=442590969 (51) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 3: To: (35) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 4: Call-ID: 363386910@10.15.6.236 (30) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 5: CSeq: 2555 REGISTER (19) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 6: Contact: ;expires=3600;+sip.instance="" (138) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 7: User-Agent: Cisco-CP7912/8.0.0-060111A (38) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 8: Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE (80) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 9: Supported: replaces, 100rel (27) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 10: Content-Length: 0 (17) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 11: (0) --- (11 headers 0 lines)--- Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3144 sip_alloc: Allocating new SIP dialog for 363386910@10.15.6.236 - REGISTER (No RTP) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:11140 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER Using latest REGISTER request as basis request Sending to 10.15.6.236 : 5060 (NAT) Transmitting (NAT) to 10.15.6.236:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.15.6.236:5060;branch=z9hG4bK7e4c7ab3cf518b70;received=10.15.6.236 From: ;tag=442590969 To: Call-ID: 363386910@10.15.6.236 CSeq: 2555 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Transmitting (NAT) to 10.15.6.236:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.15.6.236:5060;branch=z9hG4bK7e4c7ab3cf518b70;received=10.15.6.236 From: ;tag=442590969 To: ;tag=as734bd22a Call-ID: 363386910@10.15.6.236 CSeq: 2555 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: WWW-Authenticate: Digest realm="phone1.masterhost.ru", nonce="5b09e875" Content-Length: 0 --- Scheduling destruction of call '363386910@10.15.6.236' in 15000 ms <-- SIP read from 10.15.6.236:5060: REGISTER sip:10.15.1.2 SIP/2.0 Via: SIP/2.0/UDP 10.15.6.236:5060;branch=z9hG4bK7e4c7ab3cf518b70 From: ;tag=442590969 To: Call-ID: 363386910@10.15.6.236 CSeq: 2555 REGISTER Contact: ;expires=3600;+sip.instance="" User-Agent: Cisco-CP7912/8.0.0-060111A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces, 100rel Content-Length: 0 Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 0: REGISTER sip:10.15.1.2 SIP/2.0 (30) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 1: Via: SIP/2.0/UDP 10.15.6.236:5060;branch=z9hG4bK7e4c7ab3cf518b70 (64) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 2: From: ;tag=442590969 (51) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 3: To: (35) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 4: Call-ID: 363386910@10.15.6.236 (30) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 5: CSeq: 2555 REGISTER (19) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 6: Contact: ;expires=3600;+sip.instance="" (138) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 7: User-Agent: Cisco-CP7912/8.0.0-060111A (38) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 8: Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE (80) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 9: Supported: replaces, 100rel (27) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 10: Content-Length: 0 (17) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 11: (0) --- (11 headers 0 lines)--- Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:11140 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:11154 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 2555, ours 2555) Using latest REGISTER request as basis request Sending to 10.15.6.236 : 5060 (NAT) Transmitting (NAT) to 10.15.6.236:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.15.6.236:5060;branch=z9hG4bK7e4c7ab3cf518b70;received=10.15.6.236 From: ;tag=442590969 To: Call-ID: 363386910@10.15.6.236 CSeq: 2555 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Transmitting (NAT) to 10.15.6.236:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.15.6.236:5060;branch=z9hG4bK7e4c7ab3cf518b70;received=10.15.6.236 From: ;tag=442590969 To: ;tag=as734bd22a Call-ID: 363386910@10.15.6.236 CSeq: 2555 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: WWW-Authenticate: Digest realm="phone1.masterhost.ru", nonce="5b09e875" Content-Length: 0 --- Scheduling destruction of call '363386910@10.15.6.236' in 15000 ms <-- SIP read from 10.15.6.236:5060: REGISTER sip:10.15.1.2 SIP/2.0 Via: SIP/2.0/UDP 10.15.6.236:5060;branch=z9hG4bK7e4c7ab3cf518b70 From: ;tag=442590969 To: Call-ID: 363386910@10.15.6.236 CSeq: 2555 REGISTER Contact: ;expires=3600;+sip.instance="" User-Agent: Cisco-CP7912/8.0.0-060111A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces, 100rel Content-Length: 0 Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 0: REGISTER sip:10.15.1.2 SIP/2.0 (30) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 1: Via: SIP/2.0/UDP 10.15.6.236:5060;branch=z9hG4bK7e4c7ab3cf518b70 (64) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 2: From: ;tag=442590969 (51) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 3: To: (35) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 4: Call-ID: 363386910@10.15.6.236 (30) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 5: CSeq: 2555 REGISTER (19) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 6: Contact: ;expires=3600;+sip.instance="" (138) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 7: User-Agent: Cisco-CP7912/8.0.0-060111A (38) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 8: Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE (80) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 9: Supported: replaces, 100rel (27) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 10: Content-Length: 0 (17) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 11: (0) --- (11 headers 0 lines)--- Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:11140 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:11154 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 2555, ours 2555) Using latest REGISTER request as basis request Sending to 10.15.6.236 : 5060 (NAT) Transmitting (NAT) to 10.15.6.236:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.15.6.236:5060;branch=z9hG4bK7e4c7ab3cf518b70;received=10.15.6.236 From: ;tag=442590969 To: Call-ID: 363386910@10.15.6.236 CSeq: 2555 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Transmitting (NAT) to 10.15.6.236:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.15.6.236:5060;branch=z9hG4bK7e4c7ab3cf518b70;received=10.15.6.236 From: ;tag=442590969 To: ;tag=as734bd22a Call-ID: 363386910@10.15.6.236 CSeq: 2555 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: WWW-Authenticate: Digest realm="phone1.masterhost.ru", nonce="5b09e875" Content-Length: 0 --- Scheduling destruction of call '363386910@10.15.6.236' in 15000 ms <-- SIP read from 10.15.6.236:5060: REGISTER sip:10.15.1.2 SIP/2.0 Via: SIP/2.0/UDP 10.15.6.236:5060;branch=z9hG4bK7e4c7ab3cf518b70 From: ;tag=442590969 To: Call-ID: 363386910@10.15.6.236 CSeq: 2555 REGISTER Contact: ;expires=3600;+sip.instance="" User-Agent: Cisco-CP7912/8.0.0-060111A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces, 100rel Content-Length: 0 Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 0: REGISTER sip:10.15.1.2 SIP/2.0 (30) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 1: Via: SIP/2.0/UDP 10.15.6.236:5060;branch=z9hG4bK7e4c7ab3cf518b70 (64) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 2: From: ;tag=442590969 (51) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 3: To: (35) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 4: Call-ID: 363386910@10.15.6.236 (30) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 5: CSeq: 2555 REGISTER (19) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 6: Contact: ;expires=3600;+sip.instance="" (138) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 7: User-Agent: Cisco-CP7912/8.0.0-060111A (38) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 8: Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE (80) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 9: Supported: replaces, 100rel (27) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 10: Content-Length: 0 (17) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 11: (0) --- (11 headers 0 lines)--- Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:11140 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:11154 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 2555, ours 2555) Using latest REGISTER request as basis request Sending to 10.15.6.236 : 5060 (NAT) Transmitting (NAT) to 10.15.6.236:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.15.6.236:5060;branch=z9hG4bK7e4c7ab3cf518b70;received=10.15.6.236 From: ;tag=442590969 To: Call-ID: 363386910@10.15.6.236 CSeq: 2555 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Transmitting (NAT) to 10.15.6.236:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.15.6.236:5060;branch=z9hG4bK7e4c7ab3cf518b70;received=10.15.6.236 From: ;tag=442590969 To: ;tag=as734bd22a Call-ID: 363386910@10.15.6.236 CSeq: 2555 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: WWW-Authenticate: Digest realm="phone1.masterhost.ru", nonce="5b09e875" Content-Length: 0 --- Scheduling destruction of call '363386910@10.15.6.236' in 15000 ms <-- SIP read from 10.10.11.200:5060: OPTIONS sip:10.10.11.61 SIP/2.0 Via: SIP/2.0/UDP 10.10.11.200;rport;branch=z9hG4bK0a0a0bc80000001044a83a5600004834000008de Content-Length: 0 Call-ID: 4E03B055-DDE1-488D-82FA-B9555A34F797@10.10.11.200 CSeq: 762 OPTIONS From: ;tag=670621402880 Max-Forwards: 70 To: Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 0: OPTIONS sip:10.10.11.61 SIP/2.0 (31) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 1: Via: SIP/2.0/UDP 10.10.11.200;rport;branch=z9hG4bK0a0a0bc80000001044a83a5600004834000008de (90) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 2: Content-Length: 0 (17) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 3: Call-ID: 4E03B055-DDE1-488D-82FA-B9555A34F797@10.10.11.200 (58) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 4: CSeq: 762 OPTIONS (17) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 5: From: ;tag=670621402880 (45) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 6: Max-Forwards: 70 (16) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 7: To: (21) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 8: (0) --- (8 headers 0 lines)--- Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3144 sip_alloc: Allocating new SIP dialog for 4E03B055-DDE1-488D-82FA-B9555A34F797@10.10.11.200 - OPTIONS (No RTP) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:11140 handle_request: **** Received OPTIONS (3) - Command in SIP OPTIONS Looking for s in fromoutside (domain 10.10.11.61) Transmitting (NAT) to 10.10.11.200:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.10.11.200;branch=z9hG4bK0a0a0bc80000001044a83a5600004834000008de;received=10.10.11.200;rport=5060 From: ;tag=670621402880 To: ;tag=as0659b183 Call-ID: 4E03B055-DDE1-488D-82FA-B9555A34F797@10.10.11.200 CSeq: 762 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:11341 sipsock_read: SIP message could not be handled, bad request: 4E03B055-DDE1-488D-82FA-B9555A34F797@10.10.11.200 Destroying call '4E03B055-DDE1-488D-82FA-B9555A34F797@10.10.11.200' <-- SIP read from 10.15.6.236:5060: REGISTER sip:10.15.1.2 SIP/2.0 Via: SIP/2.0/UDP 10.15.6.236:5060;branch=z9hG4bK7e4c7ab3cf518b70 From: ;tag=442590969 To: Call-ID: 363386910@10.15.6.236 CSeq: 2555 REGISTER Contact: ;expires=3600;+sip.instance="" User-Agent: Cisco-CP7912/8.0.0-060111A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces, 100rel Content-Length: 0 Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 0: REGISTER sip:10.15.1.2 SIP/2.0 (30) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 1: Via: SIP/2.0/UDP 10.15.6.236:5060;branch=z9hG4bK7e4c7ab3cf518b70 (64) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 2: From: ;tag=442590969 (51) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 3: To: (35) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 4: Call-ID: 363386910@10.15.6.236 (30) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 5: CSeq: 2555 REGISTER (19) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 6: Contact: ;expires=3600;+sip.instance="" (138) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 7: User-Agent: Cisco-CP7912/8.0.0-060111A (38) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 8: Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE (80) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 9: Supported: replaces, 100rel (27) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 10: Content-Length: 0 (17) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 11: (0) --- (11 headers 0 lines)--- Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:11140 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:11154 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 2555, ours 2555) Using latest REGISTER request as basis request Sending to 10.15.6.236 : 5060 (NAT) Transmitting (NAT) to 10.15.6.236:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.15.6.236:5060;branch=z9hG4bK7e4c7ab3cf518b70;received=10.15.6.236 From: ;tag=442590969 To: Call-ID: 363386910@10.15.6.236 CSeq: 2555 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Transmitting (NAT) to 10.15.6.236:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.15.6.236:5060;branch=z9hG4bK7e4c7ab3cf518b70;received=10.15.6.236 From: ;tag=442590969 To: ;tag=as734bd22a Call-ID: 363386910@10.15.6.236 CSeq: 2555 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: WWW-Authenticate: Digest realm="phone1.masterhost.ru", nonce="5b09e875" Content-Length: 0 --- Scheduling destruction of call '363386910@10.15.6.236' in 15000 ms <-- SIP read from 195.14.53.142:5060: OPTIONS sip:phone2.masterhost.ru SIP/2.0 Via: SIP/2.0/UDP 192.168.200.218;rport;branch=z9hG4bKc0a8c8da0000001044a83a4f00002d7200001fd3 Content-Length: 0 Call-ID: 169469BD-4B0D-4AEC-B64B-77E6BB62D403@192.168.200.218 CSeq: 4246 OPTIONS From: ;tag=22471925017657 Max-Forwards: 70 To: Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 0: OPTIONS sip:phone2.masterhost.ru SIP/2.0 (40) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.200.218;rport;branch=z9hG4bKc0a8c8da0000001044a83a4f00002d7200001fd3 (93) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 2: Content-Length: 0 (17) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 3: Call-ID: 169469BD-4B0D-4AEC-B64B-77E6BB62D403@192.168.200.218 (61) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 4: CSeq: 4246 OPTIONS (18) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 5: From: ;tag=22471925017657 (56) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 6: Max-Forwards: 70 (16) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 7: To: (30) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 10: Content-Length: 0 (17) == Primary D-Channel on span 2 up Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 11: (0) --- (11 headers 0 lines)--- Jul 3 01:28:14 DEBUG[23510]: chan_iax2.c:9340 iax2_devicestate: Checking device state for device iaxmodem02 == Primary D-Channel on span 2 up Jul 3 01:28:14 DEBUG[23510]: chan_iax2.c:9348 iax2_devicestate: iax2_devicestate: Found peer. What's device state of iaxmodem02? addr=0, defaddr=0 maxms=0, lastms=0 == Primary D-Channel on span 2 up Jul 3 01:28:14 DEBUG[23510]: devicestate.c:187 do_state_change: Changing state for IAX2/iaxmodem02 - state 5 (Unavailable) Jul 3 01:28:14 DEBUG[23589]: app_queue.c:490 changethread: Device 'IAX2/iaxmodem02' changed to state '5' (Unavailable) but we don't care because they're not a member of any queue. Jul 3 01:28:14 DEBUG[23517]: chan_iax2.c:6581 socket_read: Packet arrived out of order (expecting 1, got 0) (frametype = 6, subclass = 13) Jul 3 01:28:14 DEBUG[23517]: chan_iax2.c:6588 socket_read: Acking anyway Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:11140 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:11154 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 2555, ours 2555) Using latest REGISTER request as basis request Sending to 10.15.6.236 : 5060 (NAT) Jul 3 01:28:14 DEBUG[23517]: chan_iax2.c:6581 socket_read: Packet arrived out of order (expecting 1, got 0) (frametype = 6, subclass = 13) Transmitting (NAT) to 10.15.6.236:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.15.6.236:5060;branch=z9hG4bK7e4c7ab3cf518b70;received=10.15.6.236 From: ;tag=442590969 To: Call-ID: 363386910@10.15.6.236 CSeq: 2555 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Jul 3 01:28:14 DEBUG[23517]: chan_iax2.c:6588 socket_read: Acking anyway Transmitting (NAT) to 10.15.6.236:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.15.6.236:5060;branch=z9hG4bK7e4c7ab3cf518b70;received=10.15.6.236 From: ;tag=442590969 To: ;tag=as734bd22a Call-ID: 363386910@10.15.6.236 CSeq: 2555 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: WWW-Authenticate: Digest realm="phone1.masterhost.ru", nonce="5b09e875" Content-Length: 0 --- Scheduling destruction of call '363386910@10.15.6.236' in 15000 ms Jul 3 01:28:14 DEBUG[23517]: chan_iax2.c:6581 socket_read: Packet arrived out of order (expecting 1, got 0) (frametype = 6, subclass = 13) Jul 3 01:28:14 DEBUG[23517]: chan_iax2.c:6588 socket_read: Acking anyway <-- SIP read from 10.10.11.134:5060: OPTIONS sip:10.10.11.61 SIP/2.0 Via: SIP/2.0/UDP 10.10.11.134;rport;branch=z9hG4bK0a0a0b860000001044a83a5b000024c30001096e Content-Length: 0 Call-ID: 23D02ACD-0BD6-43A6-B6CB-DF5066A7D3AA@10.10.11.134 CSeq: 28476 OPTIONS From: ;tag=80357531224228 Max-Forwards: 70 To: Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 0: OPTIONS sip:10.10.11.61 SIP/2.0 (31) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 1: Via: SIP/2.0/UDP 10.10.11.134;rport;branch=z9hG4bK0a0a0b860000001044a83a5b000024c30001096e (90) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 2: Content-Length: 0 (17) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 3: Call-ID: 23D02ACD-0BD6-43A6-B6CB-DF5066A7D3AA@10.10.11.134 (58) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 4: CSeq: 28476 OPTIONS (19) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 5: From: ;tag=80357531224228 (47) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 6: Max-Forwards: 70 (16) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 7: To: (21) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 8: (0) --- (8 headers 0 lines)--- Jul 3 01:28:14 DEBUG[23517]: chan_iax2.c:6581 socket_read: Packet arrived out of order (expecting 1, got 0) (frametype = 6, subclass = 13) Jul 3 01:28:14 DEBUG[23517]: chan_iax2.c:6588 socket_read: Acking anyway Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3144 sip_alloc: Allocating new SIP dialog for 23D02ACD-0BD6-43A6-B6CB-DF5066A7D3AA@10.10.11.134 - OPTIONS (No RTP) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:11140 handle_request: **** Received OPTIONS (3) - Command in SIP OPTIONS Looking for s in fromoutside (domain 10.10.11.61) Jul 3 01:28:14 DEBUG[23517]: chan_iax2.c:6581 socket_read: Packet arrived out of order (expecting 1, got 0) (frametype = 6, subclass = 13) Jul 3 01:28:14 DEBUG[23517]: chan_iax2.c:6588 socket_read: Acking anyway Transmitting (NAT) to 10.10.11.134:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.10.11.134;branch=z9hG4bK0a0a0b860000001044a83a5b000024c30001096e;received=10.10.11.134;rport=5060 From: ;tag=80357531224228 To: ;tag=as5092cf33 Call-ID: 23D02ACD-0BD6-43A6-B6CB-DF5066A7D3AA@10.10.11.134 CSeq: 28476 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:11341 sipsock_read: SIP message could not be handled, bad request: 23D02ACD-0BD6-43A6-B6CB-DF5066A7D3AA@10.10.11.134 Jul 3 01:28:14 DEBUG[23517]: chan_iax2.c:6581 socket_read: Destroying call '23D02ACD-0BD6-43A6-B6CB-DF5066A7D3AA@10.10.11.134' Packet arrived out of order (expecting 1, got 0) (frametype = 6, subclass = 13) Jul 3 01:28:14 DEBUG[23517]: chan_iax2.c:6588 socket_read: <-- SIP read from 10.10.11.145:5060: OPTIONS sip:10.10.11.61 SIP/2.0 Via: SIP/2.0/UDP 10.10.11.145;rport;branch=z9hG4bK0a0a0b910000001044a83a5c00003c2000002ad4 Content-Length: 0 Call-ID: A03FC774-DC9D-472A-BCBE-388C15D77FF5@10.10.11.145 CSeq: 4301 OPTIONS From: ;tag=1293082189222 Max-Forwards: 70 To: Acking anyway Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 0: OPTIONS sip:10.10.11.61 SIP/2.0 (31) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 1: Via: SIP/2.0/UDP 10.10.11.145;rport;branch=z9hG4bK0a0a0b910000001044a83a5c00003c2000002ad4 (90) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 2: Content-Length: 0 (17) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 3: Call-ID: A03FC774-DC9D-472A-BCBE-388C15D77FF5@10.10.11.145 (58) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 4: CSeq: 4301 OPTIONS (18) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 5: From: ;tag=1293082189222 (46) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 6: Max-Forwards: 70 (16) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 7: To: (21) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 8: (0) --- (8 headers 0 lines)--- Jul 3 01:28:14 DEBUG[23517]: chan_iax2.c:6581 socket_read: Packet arrived out of order (expecting 1, got 0) (frametype = 6, subclass = 13) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3144 sip_alloc: Allocating new SIP dialog for A03FC774-DC9D-472A-BCBE-388C15D77FF5@10.10.11.145 - OPTIONS (No RTP) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:11140 handle_request: **** Received OPTIONS (3) - Command in SIP OPTIONS Looking for s in fromoutside (domain 10.10.11.61) Jul 3 01:28:14 DEBUG[23517]: chan_iax2.c:6588 socket_read: Acking anyway Transmitting (NAT) to 10.10.11.145:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.10.11.145;branch=z9hG4bK0a0a0b910000001044a83a5c00003c2000002ad4;received=10.10.11.145;rport=5060 From: ;tag=1293082189222 To: ;tag=as5cc45b41 Call-ID: A03FC774-DC9D-472A-BCBE-388C15D77FF5@10.10.11.145 CSeq: 4301 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:11341 sipsock_read: SIP message could not be handled, bad request: A03FC774-DC9D-472A-BCBE-388C15D77FF5@10.10.11.145 Destroying call 'A03FC774-DC9D-472A-BCBE-388C15D77FF5@10.10.11.145' <-- SIP read from 85.140.10.142:62443: OPTIONS sip:phone2.masterhost.ru SIP/2.0 Via: SIP/2.0/UDP 10.10.10.7;rport;branch=z9hG4bK0a0a0a070000000b44a83a5c00001a870000022b Content-Length: 0 Call-ID: 1F1AB8A8-761A-4E01-9221-D9AB61AC8B76@10.10.10.7 CSeq: 194 OPTIONS From: ;tag=47979657013131 Max-Forwards: 70 To: Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 0: OPTIONS sip:phone2.masterhost.ru SIP/2.0 (40) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 1: Via: SIP/2.0/UDP 10.10.10.7;rport;branch=z9hG4bK0a0a0a070000000b44a83a5c00001a870000022b (88) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 2: Content-Length: 0 (17) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 3: Call-ID: 1F1AB8A8-761A-4E01-9221-D9AB61AC8B76@10.10.10.7 (56) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 4: CSeq: 194 OPTIONS (17) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 5: From: ;tag=47979657013131 (56) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 6: Max-Forwards: 70 (16) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 7: To: (30) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 8: (0) --- (8 headers 0 lines)--- Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3144 sip_alloc: Allocating new SIP dialog for 1F1AB8A8-761A-4E01-9221-D9AB61AC8B76@10.10.10.7 - OPTIONS (No RTP) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:11140 handle_request: **** Received OPTIONS (3) - Command in SIP OPTIONS Looking for s in fromoutside (domain phone2.masterhost.ru) Jul 3 01:28:14 DEBUG[23517]: chan_iax2.c:6581 socket_read: Packet arrived out of order (expecting 1, got 0) (frametype = 6, subclass = 13) Transmitting (NAT) to 85.140.10.142:62443: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.10.10.7;branch=z9hG4bK0a0a0a070000000b44a83a5c00001a870000022b;received=85.140.10.142;rport=62443 From: ;tag=47979657013131 To: ;tag=as5ece47dd Call-ID: 1F1AB8A8-761A-4E01-9221-D9AB61AC8B76@10.10.10.7 CSeq: 194 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Jul 3 01:28:14 DEBUG[23517]: chan_iax2.c:6588 socket_read: Acking anyway Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:11341 sipsock_read: SIP message could not be handled, bad request: 1F1AB8A8-761A-4E01-9221-D9AB61AC8B76@10.10.10.7 Destroying call '1F1AB8A8-761A-4E01-9221-D9AB61AC8B76@10.10.10.7' Jul 3 01:28:14 DEBUG[23517]: chan_iax2.c:6581 socket_read: Packet arrived out of order (expecting 1, got 0) (frametype = 6, subclass = 13) Jul 3 01:28:14 DEBUG[23517]: chan_iax2.c:6588 socket_read: <-- SIP read from 10.15.6.236:5060: REGISTER sip:10.15.1.2 SIP/2.0 Via: SIP/2.0/UDP 10.15.6.236:5060;branch=z9hG4bK7e4c7ab3cf518b70 From: ;tag=442590969 To: Call-ID: 363386910@10.15.6.236 CSeq: 2555 REGISTER Contact: ;expires=3600;+sip.instance="" User-Agent: Cisco-CP7912/8.0.0-060111A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces, 100rel Content-Length: 0 Acking anyway Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 0: REGISTER sip:10.15.1.2 SIP/2.0 (30) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 1: Via: SIP/2.0/UDP 10.15.6.236:5060;branch=z9hG4bK7e4c7ab3cf518b70 (64) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 2: From: ;tag=442590969 (51) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 3: To: (35) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 4: Call-ID: 363386910@10.15.6.236 (30) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 5: CSeq: 2555 REGISTER (19) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 6: Contact: ;expires=3600;+sip.instance="" (138) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 7: User-Agent: Cisco-CP7912/8.0.0-060111A (38) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 8: Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE (80) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 9: Supported: replaces, 100rel (27) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 10: Content-Length: 0 (17) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 11: (0) --- (11 headers 0 lines)--- Jul 3 01:28:14 DEBUG[23517]: chan_iax2.c:6581 socket_read: Packet arrived out of order (expecting 1, got 0) (frametype = 6, subclass = 13) Jul 3 01:28:14 DEBUG[23517]: chan_iax2.c:6588 socket_read: Acking anyway Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:11140 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:11154 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 2555, ours 2555) Using latest REGISTER request as basis request Sending to 10.15.6.236 : 5060 (NAT) Transmitting (NAT) to 10.15.6.236:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.15.6.236:5060;branch=z9hG4bK7e4c7ab3cf518b70;received=10.15.6.236 From: ;tag=442590969 To: Call-ID: 363386910@10.15.6.236 CSeq: 2555 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Transmitting (NAT) to 10.15.6.236:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.15.6.236:5060;branch=z9hG4bK7e4c7ab3cf518b70;received=10.15.6.236 From: ;tag=442590969 To: ;tag=as734bd22a Call-ID: 363386910@10.15.6.236 CSeq: 2555 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: WWW-Authenticate: Digest realm="phone1.masterhost.ru", nonce="5b09e875" Content-Length: 0 --- Scheduling destruction of call '363386910@10.15.6.236' in 15000 ms <-- SIP read from 10.10.11.201:5060: OPTIONS sip:10.10.11.61 SIP/2.0 Via: SIP/2.0/UDP 10.10.11.201;rport;branch=z9hG4bK0a0a0bc90000001044a83a61000015d50001180e Content-Length: 0 Call-ID: 75817351-940B-4522-A96B-ABF0B97EDC36@10.10.11.201 CSeq: 23479 OPTIONS From: ;tag=80423006111563 Max-Forwards: 70 To: Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 0: OPTIONS sip:10.10.11.61 SIP/2.0 (31) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 1: Via: SIP/2.0/UDP 10.10.11.201;rport;branch=z9hG4bK0a0a0bc90000001044a83a61000015d50001180e (90) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 2: Content-Length: 0 (17) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 3: Call-ID: 75817351-940B-4522-A96B-ABF0B97EDC36@10.10.11.201 (58) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 4: CSeq: 23479 OPTIONS (19) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 5: From: ;tag=80423006111563 (47) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 6: Max-Forwards: 70 (16) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 7: To: (21) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 8: (0) --- (8 headers 0 lines)--- Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3144 sip_alloc: Allocating new SIP dialog for 75817351-940B-4522-A96B-ABF0B97EDC36@10.10.11.201 - OPTIONS (No RTP) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:11140 handle_request: **** Received OPTIONS (3) - Command in SIP OPTIONS Looking for s in fromoutside (domain 10.10.11.61) Transmitting (NAT) to 10.10.11.201:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.10.11.201;branch=z9hG4bK0a0a0bc90000001044a83a61000015d50001180e;received=10.10.11.201;rport=5060 From: ;tag=80423006111563 To: ;tag=as4f2773f0 Call-ID: 75817351-940B-4522-A96B-ABF0B97EDC36@10.10.11.201 CSeq: 23479 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:11341 sipsock_read: SIP message could not be handled, bad request: 75817351-940B-4522-A96B-ABF0B97EDC36@10.10.11.201 Destroying call '75817351-940B-4522-A96B-ABF0B97EDC36@10.10.11.201' <-- SIP read from 10.15.6.236:5060: REGISTER sip:10.15.1.2 SIP/2.0 Via: SIP/2.0/UDP 10.15.6.236:5060;branch=z9hG4bK7e4c7ab3cf518b70 From: ;tag=442590969 To: Call-ID: 363386910@10.15.6.236 CSeq: 2555 REGISTER Contact: ;expires=3600;+sip.instance="" User-Agent: Cisco-CP7912/8.0.0-060111A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces, 100rel Content-Length: 0 Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 0: REGISTER sip:10.15.1.2 SIP/2.0 (30) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 1: Via: SIP/2.0/UDP 10.15.6.236:5060;branch=z9hG4bK7e4c7ab3cf518b70 (64) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 2: From: ;tag=442590969 (51) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 3: To: (35) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 4: Call-ID: 363386910@10.15.6.236 (30) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 5: CSeq: 2555 REGISTER (19) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 6: Contact: ;expires=3600;+sip.instance="" (138) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 7: User-Agent: Cisco-CP7912/8.0.0-060111A (38) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 8: Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE (80) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 9: Supported: replaces, 100rel (27) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 10: Content-Length: 0 (17) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 11: (0) --- (11 headers 0 lines)--- Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:11140 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:11154 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 2555, ours 2555) Using latest REGISTER request as basis request Sending to 10.15.6.236 : 5060 (NAT) Transmitting (NAT) to 10.15.6.236:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.15.6.236:5060;branch=z9hG4bK7e4c7ab3cf518b70;received=10.15.6.236 From: ;tag=442590969 To: Call-ID: 363386910@10.15.6.236 CSeq: 2555 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Transmitting (NAT) to 10.15.6.236:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.15.6.236:5060;branch=z9hG4bK7e4c7ab3cf518b70;received=10.15.6.236 From: ;tag=442590969 To: ;tag=as734bd22a Call-ID: 363386910@10.15.6.236 CSeq: 2555 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: WWW-Authenticate: Digest realm="phone1.masterhost.ru", nonce="5b09e875" Content-Length: 0 --- Scheduling destruction of call '363386910@10.15.6.236' in 15000 ms <-- SIP read from 10.15.6.236:5060: REGISTER sip:10.15.1.2 SIP/2.0 Via: SIP/2.0/UDP 10.15.6.236:5060;branch=z9hG4bK7e4c7ab3cf518b70 From: ;tag=442590969 To: Call-ID: 363386910@10.15.6.236 CSeq: 2555 REGISTER Contact: ;expires=3600;+sip.instance="" User-Agent: Cisco-CP7912/8.0.0-060111A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces, 100rel Content-Length: 0 Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 0: REGISTER sip:10.15.1.2 SIP/2.0 (30) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 1: Via: SIP/2.0/UDP 10.15.6.236:5060;branch=z9hG4bK7e4c7ab3cf518b70 (64) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 2: From: ;tag=442590969 (51) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 3: To: (35) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 4: Call-ID: 363386910@10.15.6.236 (30) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 5: CSeq: 2555 REGISTER (19) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 6: Contact: ;expires=3600;+sip.instance="" (138) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 7: User-Agent: Cisco-CP7912/8.0.0-060111A (38) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 8: Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE (80) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 9: Supported: replaces, 100rel (27) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 10: Content-Length: 0 (17) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 11: (0) --- (11 headers 0 lines)--- Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:11140 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:11154 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 2555, ours 2555) Using latest REGISTER request as basis request Sending to 10.15.6.236 : 5060 (NAT) Transmitting (NAT) to 10.15.6.236:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.15.6.236:5060;branch=z9hG4bK7e4c7ab3cf518b70;received=10.15.6.236 From: ;tag=442590969 To: Call-ID: 363386910@10.15.6.236 CSeq: 2555 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Transmitting (NAT) to 10.15.6.236:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.15.6.236:5060;branch=z9hG4bK7e4c7ab3cf518b70;received=10.15.6.236 From: ;tag=442590969 To: ;tag=as734bd22a Call-ID: 363386910@10.15.6.236 CSeq: 2555 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: WWW-Authenticate: Digest realm="phone1.masterhost.ru", nonce="5b09e875" Content-Length: 0 --- Scheduling destruction of call '363386910@10.15.6.236' in 15000 ms <-- SIP read from 10.10.11.200:5060: OPTIONS sip:10.10.11.61 SIP/2.0 Via: SIP/2.0/UDP 10.10.11.200;rport;branch=z9hG4bK0a0a0bc80000001044a83a6a00007bcd000008df Content-Length: 0 Call-ID: 4E03B055-DDE1-488D-82FA-B9555A34F797@10.10.11.200 CSeq: 763 OPTIONS From: ;tag=6708214017217 Max-Forwards: 70 To: Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 0: OPTIONS sip:10.10.11.61 SIP/2.0 (31) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 1: Via: SIP/2.0/UDP 10.10.11.200;rport;branch=z9hG4bK0a0a0bc80000001044a83a6a00007bcd000008df (90) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 2: Content-Length: 0 (17) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 3: Call-ID: 4E03B055-DDE1-488D-82FA-B9555A34F797@10.10.11.200 (58) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 4: CSeq: 763 OPTIONS (17) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 5: From: ;tag=6708214017217 (46) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 6: Max-Forwards: 70 (16) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 7: To: (21) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 8: (0) --- (8 headers 0 lines)--- Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3144 sip_alloc: Allocating new SIP dialog for 4E03B055-DDE1-488D-82FA-B9555A34F797@10.10.11.200 - OPTIONS (No RTP) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:11140 handle_request: **** Received OPTIONS (3) - Command in SIP OPTIONS Looking for s in fromoutside (domain 10.10.11.61) Transmitting (NAT) to 10.10.11.200:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.10.11.200;branch=z9hG4bK0a0a0bc80000001044a83a6a00007bcd000008df;received=10.10.11.200;rport=5060 From: ;tag=6708214017217 To: ;tag=as26b221e4 Call-ID: 4E03B055-DDE1-488D-82FA-B9555A34F797@10.10.11.200 CSeq: 763 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:11341 sipsock_read: SIP message could not be handled, bad request: 4E03B055-DDE1-488D-82FA-B9555A34F797@10.10.11.200 Destroying call '4E03B055-DDE1-488D-82FA-B9555A34F797@10.10.11.200' <-- SIP read from 10.15.6.236:5060: REGISTER sip:10.15.1.2 SIP/2.0 Via: SIP/2.0/UDP 10.15.6.236:5060;branch=z9hG4bK7e4c7ab3cf518b70 From: ;tag=442590969 To: Call-ID: 363386910@10.15.6.236 CSeq: 2555 REGISTER Contact: ;expires=3600;+sip.instance="" User-Agent: Cisco-CP7912/8.0.0-060111A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces, 100rel Content-Length: 0 Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 0: REGISTER sip:10.15.1.2 SIP/2.0 (30) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 1: Via: SIP/2.0/UDP 10.15.6.236:5060;branch=z9hG4bK7e4c7ab3cf518b70 (64) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 2: From: ;tag=442590969 (51) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 3: To: (35) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 4: Call-ID: 363386910@10.15.6.236 (30) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 5: CSeq: 2555 REGISTER (19) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 6: Contact: ;expires=3600;+sip.instance="" (138) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 7: User-Agent: Cisco-CP7912/8.0.0-060111A (38) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 8: Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE (80) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 9: Supported: replaces, 100rel (27) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 10: Content-Length: 0 (17) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 11: (0) --- (11 headers 0 lines)--- Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:11140 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:11154 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 2555, ours 2555) Using latest REGISTER request as basis request Sending to 10.15.6.236 : 5060 (NAT) Transmitting (NAT) to 10.15.6.236:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.15.6.236:5060;branch=z9hG4bK7e4c7ab3cf518b70;received=10.15.6.236 From: ;tag=442590969 To: Call-ID: 363386910@10.15.6.236 CSeq: 2555 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Jul 3 01:28:14 DEBUG[23517]: chan_iax2.c:6581 socket_read: Packet arrived out of order (expecting 1, got 0) (frametype = 6, subclass = 13) Jul 3 01:28:14 DEBUG[23517]: chan_iax2.c:6588 socket_read: Transmitting (NAT) to 10.15.6.236:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.15.6.236:5060;branch=z9hG4bK7e4c7ab3cf518b70;received=10.15.6.236 From: ;tag=442590969 To: ;tag=as734bd22a Call-ID: 363386910@10.15.6.236 CSeq: 2555 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: WWW-Authenticate: Digest realm="phone1.masterhost.ru", nonce="5b09e875" Content-Length: 0 --- Acking anyway Scheduling destruction of call '363386910@10.15.6.236' in 15000 ms <-- SIP read from 195.14.53.142:5060: OPTIONS sip:phone2.masterhost.ru SIP/2.0 Via: SIP/2.0/UDP 192.168.200.218;rport;branch=z9hG4bKc0a8c8da0000001044a83a630000515f00001fd4 Content-Length: 0 Call-ID: 169469BD-4B0D-4AEC-B64B-77E6BB62D403@192.168.200.218 CSeq: 4247 OPTIONS From: ;tag=22473925016486 Max-Forwards: 70 To: Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 0: OPTIONS sip:phone2.masterhost.ru SIP/2.0 (40) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.200.218;rport;branch=z9hG4bKc0a8c8da0000001044a83a630000515f00001fd4 (93) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 2: Content-Length: 0 (17) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 3: Call-ID: 169469BD-4B0D-4AEC-B64B-77E6BB62D403@192.168.200.218 (61) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 4: CSeq: 4247 OPTIONS (18) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 5: From: ;tag=22473925016486 (56) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 6: Max-Forwards: 70 (16) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 7: To: (30) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 8: (0) --- (8 headers 0 lines)--- Jul 3 01:28:14 DEBUG[23517]: chan_iax2.c:6581 socket_read: Packet arrived out of order (expecting 1, got 0) (frametype = 6, subclass = 13) Jul 3 01:28:14 DEBUG[23517]: chan_iax2.c:6588 socket_read: Acking anyway Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3144 sip_alloc: Allocating new SIP dialog for 169469BD-4B0D-4AEC-B64B-77E6BB62D403@192.168.200.218 - OPTIONS (No RTP) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:11140 handle_request: **** Received OPTIONS (3) - Command in SIP OPTIONS Looking for s in fromoutside (domain phone2.masterhost.ru) Transmitting (NAT) to 195.14.53.142:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.200.218;branch=z9hG4bKc0a8c8da0000001044a83a630000515f00001fd4;received=195.14.53.142;rport=5060 From: ;tag=22473925016486 To: ;tag=as54efa156 Call-ID: 169469BD-4B0D-4AEC-B64B-77E6BB62D403@192.168.200.218 CSeq: 4247 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:11341 sipsock_read: SIP message could not be handled, bad request: 169469BD-4B0D-4AEC-B64B-77E6BB62D403@192.168.200.218 Destroying call '169469BD-4B0D-4AEC-B64B-77E6BB62D403@192.168.200.218' Jul 3 01:28:14 DEBUG[23517]: chan_iax2.c:6581 socket_read: Packet arrived out of order (expecting 1, got 0) (frametype = 6, subclass = 13) Jul 3 01:28:14 DEBUG[23517]: chan_iax2.c:6588 socket_read: Acking anyway Jul 3 01:28:14 DEBUG[23517]: chan_iax2.c:6581 socket_read: Packet arrived out of order (expecting 1, got 0) (frametype = 6, subclass = 13) Jul 3 01:28:14 DEBUG[23517]: chan_iax2.c:6588 socket_read: Acking anyway Jul 3 01:28:14 DEBUG[23510]: chan_iax2.c:9340 iax2_devicestate: Checking device state for device iaxmodem02 Jul 3 01:28:14 DEBUG[23510]: chan_iax2.c:9348 iax2_devicestate: iax2_devicestate: Found peer. What's device state of iaxmodem02? addr=16777343, defaddr=0 maxms=0, lastms=0 Jul 3 01:28:14 DEBUG[23510]: devicestate.c:187 do_state_change: Changing state for IAX2/iaxmodem02 - state 1 (Not in use) Jul 3 01:28:14 DEBUG[23590]: app_queue.c:490 changethread: Device 'IAX2/iaxmodem02' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. -- Registered IAX2 'iaxmodem02' (AUTHENTICATED) at 127.0.0.1:4572 Jul 3 01:28:14 DEBUG[23510]: chan_iax2.c:9340 iax2_devicestate: Checking device state for device iaxmodem02 Jul 3 01:28:14 DEBUG[23510]: chan_iax2.c:9348 iax2_devicestate: iax2_devicestate: Found peer. What's device state of iaxmodem02? addr=16777343, defaddr=0 maxms=0, lastms=0 Jul 3 01:28:14 DEBUG[23510]: devicestate.c:187 do_state_change: Changing state for IAX2/iaxmodem02 - state 1 (Not in use) Jul 3 01:28:14 DEBUG[23591]: app_queue.c:490 changethread: Device 'IAX2/iaxmodem02' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. Jul 3 01:28:14 DEBUG[23510]: chan_iax2.c:9340 iax2_devicestate: Checking device state for device iaxmodem01 Jul 3 01:28:14 DEBUG[23510]: chan_iax2.c:9348 iax2_devicestate: iax2_devicestate: Found peer. What's device state of iaxmodem01? addr=16777343, defaddr=0 maxms=0, lastms=0 Jul 3 01:28:14 DEBUG[23510]: devicestate.c:187 do_state_change: Changing state for IAX2/iaxmodem01 - state 1 (Not in use) Jul 3 01:28:14 DEBUG[23592]: app_queue.c:490 changethread: Device 'IAX2/iaxmodem01' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. -- Registered IAX2 'iaxmodem01' (AUTHENTICATED) at 127.0.0.1:4571 Jul 3 01:28:14 DEBUG[23510]: chan_iax2.c:9340 iax2_devicestate: Checking device state for device iaxmodem01 Jul 3 01:28:14 DEBUG[23510]: chan_iax2.c:9348 iax2_devicestate: iax2_devicestate: Found peer. What's device state of iaxmodem01? addr=16777343, defaddr=0 maxms=0, lastms=0 Jul 3 01:28:14 DEBUG[23510]: devicestate.c:187 do_state_change: Changing state for IAX2/iaxmodem01 - state 1 (Not in use) Jul 3 01:28:14 DEBUG[23593]: app_queue.c:490 changethread: Device 'IAX2/iaxmodem01' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. <-- SIP read from 10.15.6.236:5060: REGISTER sip:10.15.1.2 SIP/2.0 Via: SIP/2.0/UDP 10.15.6.236:5060;branch=z9hG4bKeefab19bd97c0528 From: ;tag=442590969 To: Call-ID: 363386910@10.15.6.236 CSeq: 2556 REGISTER Contact: ;expires=3600;+sip.instance="" User-Agent: Cisco-CP7912/8.0.0-060111A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces, 100rel Authorization: Digest username="1346",realm="phone1.masterhost.ru",nonce="5b09e875",uri="sip:10.15.1.2",response="***" Content-Length: 0 Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 0: REGISTER sip:10.15.1.2 SIP/2.0 (30) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 1: Via: SIP/2.0/UDP 10.15.6.236:5060;branch=z9hG4bKeefab19bd97c0528 (64) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 2: From: ;tag=442590969 (51) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 3: To: (35) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 4: Call-ID: 363386910@10.15.6.236 (30) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 5: CSeq: 2556 REGISTER (19) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 6: Contact: ;expires=3600;+sip.instance="" (138) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 7: User-Agent: Cisco-CP7912/8.0.0-060111A (38) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 8: Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE (80) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 9: Supported: replaces, 100rel (27) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 10: Authorization: Digest username="1346",realm="phone1.masterhost.ru",nonce="5b09e875",uri="sip:10.15.1.2",response="***" (147) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 11: Content-Length: 0 (17) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 12: (0) --- (12 headers 0 lines)--- Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:11140 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER Using latest REGISTER request as basis request Sending to 10.15.6.236 : 5060 (NAT) Transmitting (NAT) to 10.15.6.236:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.15.6.236:5060;branch=z9hG4bKeefab19bd97c0528;received=10.15.6.236 From: ;tag=442590969 To: Call-ID: 363386910@10.15.6.236 CSeq: 2556 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- -- Saved useragent "Cisco-CP7912/8.0.0-060111A" for peer 1346 Transmitting (NAT) to 10.15.6.236:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.15.6.236:5060;branch=z9hG4bKeefab19bd97c0528;received=10.15.6.236 From: ;tag=442590969 To: ;tag=as734bd22a Call-ID: 363386910@10.15.6.236 CSeq: 2556 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Expires: 3600 Contact: ;expires=3600 Date: Sun, 02 Jul 2006 21:28:14 GMT Content-Length: 0 --- Scheduling destruction of call '363386910@10.15.6.236' in 15000 ms Jul 3 01:28:14 DEBUG[23510]: chan_sip.c:11671 sip_devicestate: Checking device state for peer 1346 Jul 3 01:28:14 DEBUG[23510]: devicestate.c:187 do_state_change: Changing state for SIP/1346 - state 1 (Not in use) Jul 3 01:28:14 DEBUG[23594]: app_queue.c:490 changethread: Device 'SIP/1346' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. <-- SIP read from 10.15.6.236:5060: REGISTER sip:10.15.1.2 SIP/2.0 Via: SIP/2.0/UDP 10.15.6.236:5060;branch=z9hG4bKeefab19bd97c0528 From: ;tag=442590969 To: Call-ID: 363386910@10.15.6.236 CSeq: 2556 REGISTER Contact: ;expires=3600;+sip.instance="" User-Agent: Cisco-CP7912/8.0.0-060111A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces, 100rel Authorization: Digest username="1346",realm="phone1.masterhost.ru",nonce="5b09e875",uri="sip:10.15.1.2",response="***" Content-Length: 0 Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 0: REGISTER sip:10.15.1.2 SIP/2.0 (30) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 1: Via: SIP/2.0/UDP 10.15.6.236:5060;branch=z9hG4bKeefab19bd97c0528 (64) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 2: From: ;tag=442590969 (51) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 3: To: (35) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 4: Call-ID: 363386910@10.15.6.236 (30) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 5: CSeq: 2556 REGISTER (19) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 6: Contact: ;expires=3600;+sip.instance="" (138) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 7: User-Agent: Cisco-CP7912/8.0.0-060111A (38) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 8: Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE (80) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 9: Supported: replaces, 100rel (27) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 10: Authorization: Digest username="1346",realm="phone1.masterhost.ru",nonce="5b09e875",uri="sip:10.15.1.2",response="***" (147) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 11: Content-Length: 0 (17) Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 12: (0) --- (12 headers 0 lines)--- Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:11140 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER Jul 3 01:28:14 DEBUG[23515]: chan_sip.c:11154 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 2556, ours 2556) Using latest REGISTER request as basis request Sending to 10.15.6.236 : 5060 (NAT) Transmitting (NAT) to 10.15.6.236:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.15.6.236:5060;branch=z9hG4bKeefab19bd97c0528;received=10.15.6.236 From: ;tag=442590969 To: Call-ID: 363386910@10.15.6.236 CSeq: 2556 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Transmitting (NAT) to 10.15.6.236:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.15.6.236:5060;branch=z9hG4bKeefab19bd97c0528;received=10.15.6.236 From: ;tag=442590969 To: ;tag=as734bd22a Call-ID: 363386910@10.15.6.236 CSeq: 2556 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Expires: 3600 Contact: ;expires=3600 Date: Sun, 02 Jul 2006 21:28:14 GMT Content-Length: 0 --- Scheduling destruction of call '363386910@10.15.6.236' in 15000 ms Jul 3 01:28:14 DEBUG[23510]: chan_sip.c:11671 sip_devicestate: Checking device state for peer 1346 Jul 3 01:28:14 DEBUG[23510]: devicestate.c:187 do_state_change: Changing state for SIP/1346 - state 1 (Not in use) Jul 3 01:28:14 DEBUG[23595]: app_queue.c:490 changethread: Device 'SIP/1346' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. st <-- SIP read from 10.10.11.134:5060: OPTIONS sip:10.10.11.61 SIP/2.0 Via: SIP/2.0/UDP 10.10.11.134;rport;branch=z9hG4bK0a0a0b860000001044a83a6f0000022100010971 Content-Length: 0 Call-ID: 23D02ACD-0BD6-43A6-B6CB-DF5066A7D3AA@10.10.11.134 CSeq: 28477 OPTIONS From: ;tag=80359531223900 Max-Forwards: 70 To: Jul 3 01:28:15 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 0: OPTIONS sip:10.10.11.61 SIP/2.0 (31) Jul 3 01:28:15 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 1: Via: SIP/2.0/UDP 10.10.11.134;rport;branch=z9hG4bK0a0a0b860000001044a83a6f0000022100010971 (90) Jul 3 01:28:15 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 2: Content-Length: 0 (17) Jul 3 01:28:15 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 3: Call-ID: 23D02ACD-0BD6-43A6-B6CB-DF5066A7D3AA@10.10.11.134 (58) Jul 3 01:28:15 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 4: CSeq: 28477 OPTIONS (19) Jul 3 01:28:15 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 5: From: ;tag=80359531223900 (47) Jul 3 01:28:15 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 6: Max-Forwards: 70 (16) Jul 3 01:28:15 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 7: To: (21) Jul 3 01:28:15 DEBUG[23515]: chan_sip.c:3360 parse_request: Header 8: (0) --- (8 headers 0 lines)--- Jul 3 01:28:15 DEBUG[23515]: chan_sip.c:3144 sip_alloc: Allocating new SIP dialog for 23D02ACD-0BD6-43A6-B6CB-DF5066A7D3AA@10.10.11.134 - OPTIONS (No RTP) Jul 3 01:28:15 DEBUG[23515]: chan_sip.c:11140 handle_request: **** Received OPTIONS (3) - Command in SIP OPTIONS Looking for s in fromoutside (domain 10.10.11.61) Transmitting (NAT) to 10.10.11.134:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.10.11.134;branch=z9hG4bK0a0a0b860000001044a83a6f0000022100010971;received=10.10.11.134;rport=5060 From: ;tag=80359531223900 To: ;tag=as450ff003 Call-ID: 23D02ACD-0BD6-43A6-B6CB-DF5066A7D3AA@10.10.11.134 CSeq: 28477 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Jul 3 01:28:15 DEBUG[23515]: chan_sip.c:11341 sipsock_read: SIP message could not be handled, bad request: 23D02ACD-0BD6-43A6-B6CB-DF5066A7D3AA@10.10.11.134 Destroying call '23D02ACD-0BD6-43A6-B6CB-DF5066A7D3AA@10.10.11.134' st == Primary D-Channel on span 2 up stop now phone2*CLI> stop now phone2*CLI>