*+freepbx не работает перевод звонков.

zzuz · Сообщение **zzuz** » 28 сен 2011, 13:53

Это вообще плохая идея - советовать пересобрать астериск , даже если это всего лишь пакеты со звуковыми файлами. А так еще 100 тем откроются "я пересобрал астериск - всё нахер поломалось, что делать , править конфиги я не хочу , читать не хочу , помогите , что не так" и в посте будет оверквотинг всей темы , где советовали что-то пересобрать или просто выполнить make.

rusya · Сообщение **rusya** » 28 сен 2011, 14:14

*Совет или предложение дается исходя из того, что вопрошающий согласен со следующими утверждениями:
... Я предварительно создал резервную копию того, что я собираюсь менять и\или резервную копию всей системы.
... Я знаю, как вернуть сделанные мной изменения из сделанной резервной копии.
... Я понимаю, что я делаю.
... Я полностью несу ответственность сам за свои действия.
... Мне 18 или более лет

nelgondar · Сообщение **nelgondar** » 29 сен 2011, 12:55

sip set debug peer 1060 при трансфере, почему-то молчит. С кодеками я разобрался.
Вывод во время перевода.

Asterisk 1.8.6.0-1digium1~squeeze, Copyright (C) 1999 - 2011 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 1.8.6.0-1digium1~squeeze currently running on debian (pid = 14916)
debian*CLI> sip set debug peer 1060
SIP Debugging Enabled for IP: x.x.x.99
[Sep 29 12:39:47] NOTICE[14930]: chan_sip.c:12593 sip_reregister: -- Re-registration for 0031754852@sipnet.ru
[Sep 29 12:39:47] NOTICE[14930]: chan_sip.c:20125 handle_response_register: Outbound Registration: Expiry for sipnet.ru is 113 sec (Scheduling reregistration in 98 s)
Reliably Transmitting (NAT) to x.x.x.99:43938:
OPTIONS sip:1060@172.16.32.200:5062 SIP/2.0
Via: SIP/2.0/UDP x.x.x.134:5060;branch=z9hG4bK77c74000;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@x.x.x.134>;tag=as6417b9ba
To: <sip:1060@172.16.32.200:5062>
Contact: <sip:Unknown@x.x.x.134:5060>
Call-ID: 2312ab5365566ede489e1f951037db25@x.x.x.134:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.9.0(1.8.6.0)
Date: Thu, 29 Sep 2011 08:40:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

---

<--- SIP read from UDP:x.x.x.99:43938 --->
SIP/2.0 200 OK
To: <sip:1060@172.16.32.200:5062>;tag=70800a688eaa7064i2
From: "Unknown" <sip:Unknown@x.x.x.134>;tag=as6417b9ba
Call-ID: 2312ab5365566ede489e1f951037db25@x.x.x.134:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP x.x.x.134:5060;branch=z9hG4bK77c74000
Server: Linksys/SPA942-6.1.5(a)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '2312ab5365566ede489e1f951037db25@x.x.x.134:5060' Method: OPTIONS
[Sep 29 12:40:40] ERROR[14983]: res_config_sqlite.c:847 cdr_handler: unable to open database file
Reliably Transmitting (NAT) to x.x.x.99:43938:
OPTIONS sip:1060@172.16.32.200:5062 SIP/2.0
Via: SIP/2.0/UDP x.x.x.134:5060;branch=z9hG4bK33eea0c5;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@x.x.x.134>;tag=as368feea6
To: <sip:1060@172.16.32.200:5062>
Contact: <sip:Unknown@x.x.x.134:5060>
Call-ID: 0b36d3d31aeb496e53a9223122382bb5@x.x.x.134:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.9.0(1.8.6.0)
Date: Thu, 29 Sep 2011 08:41:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

---

<--- SIP read from UDP:x.x.x.99:43938 --->
SIP/2.0 200 OK
To: <sip:1060@172.16.32.200:5062>;tag=70800a688eaa7064i2
From: "Unknown" <sip:Unknown@x.x.x.134>;tag=as368feea6
Call-ID: 0b36d3d31aeb496e53a9223122382bb5@x.x.x.134:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP x.x.x.134:5060;branch=z9hG4bK33eea0c5
Server: Linksys/SPA942-6.1.5(a)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '0b36d3d31aeb496e53a9223122382bb5@x.x.x.134:5060' Method: OPTIONS
debian*CLI>

ded · Сообщение **ded** » 29 сен 2011, 13:01

1) Не надо сюда постить дебаг пакетов OPTIONS (прочитайте что такое OPTIONS и для чего они используются)
2) Есть полезный таг spoiler= когда Вы выкладываете длинные дампы дебага.
3) В платный суппорт?

zzuz · Сообщение **zzuz** » 29 сен 2011, 13:03

Дебаг нужен для вас - это во-первых.
Во-вторых - это не дебаг звонка и перевода , а дебаг пакетов , которые шлются из-за опции qualify.

nelgondar · Сообщение **nelgondar** » 29 сен 2011, 15:37

Ребят, я не пытаюсь на вас перекладывать свою работу. Если чесн, я первый раз пишу на форуме по поводу астериска, до этого обходился гуглом. Работаю с астериском недавно, примерно месяц. Дело в том что даже при подробном дебаге вне в логи не выбрасывается хоть какае-то ошибка, поэтому сложно искать решение, когда вроде все работает. Я не спрашиваю что конкретно я должен сделать чтобы работало, я спрашиваю почему может не работать. Посмотрите пожалуйста мой теперь более подробный лог. Я там не нашел ничего криминального.

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

Код: Выделить всё

[Sep 29 15:13:16] DEBUG[15964]: rtp_engine.c:1537 ast_rtp_instance_early_bridge: Setting early bridge SDP of 'SIP/1011-00000000' with that of 'SIP/1777-00000001'
[Sep 29 15:13:16] DEBUG[15964]: chan_sip.c:3323 __sip_xmit: Trying to put 'SIP/2.0 180' onto UDP socket destined for x.x.x.99:1541
[Sep 29 15:13:16] DEBUG[15922]: devicestate.c:340 _ast_device_state: No provider found, checking channel drivers for SIP - 1777
[Sep 29 15:13:16] DEBUG[15922]: chan_sip.c:25615 sip_devicestate: Checking device state for peer 1777
[Sep 29 15:13:16] DEBUG[15922]: devicestate.c:458 do_state_change: Changing state for SIP/1777 - state 6 (Ringing)
[Sep 29 15:13:16] DEBUG[15922]: devicestate.c:438 devstate_event: device 'SIP/1777' state '6'
[Sep 29 15:13:16] DEBUG[15957]: app_queue.c:1491 handle_statechange: Device 'SIP/1777' changed to state '6' (Ringing) but we don't care because they're not a member of any queue.
[Sep 29 15:13:18] DEBUG[15930]: chan_sip.c:7833 find_call: = Looking for  Call ID: 6a33c3f719d1eb1a287e7720226014fc@x.x.x.134:5060 (Checking To) --From tag as0ef8ec24 --To-tag 6cf83e6ce6414de3i1  
[Sep 29 15:13:18] DEBUG[15930]: chan_sip.c:3971 __sip_ack: Acked pending invite 102
[Sep 29 15:13:18] DEBUG[15930]: chan_sip.c:4009 __sip_ack: Stopping retransmission on '6a33c3f719d1eb1a287e7720226014fc@x.x.x.134:5060' of Request 102: Match Found
[Sep 29 15:13:18] DEBUG[15930]: chan_sip.c:19349 handle_response_invite: SIP response 200 to standard invite
[Sep 29 15:13:18] DEBUG[15930]: chan_sip.c:8624 process_sdp: Processing session-level SDP v=0... UNSUPPORTED.
[Sep 29 15:13:18] DEBUG[15930]: chan_sip.c:8624 process_sdp: Processing session-level SDP o=- 2303712 2303712 IN IP4 172.16.32.162... UNSUPPORTED.
[Sep 29 15:13:18] DEBUG[15930]: chan_sip.c:8624 process_sdp: Processing session-level SDP s=-... UNSUPPORTED.
[Sep 29 15:13:18] DEBUG[15930]: netsock2.c:134 ast_sockaddr_split_hostport: Splitting '172.16.32.162' into...
[Sep 29 15:13:18] DEBUG[15930]: netsock2.c:188 ast_sockaddr_split_hostport: ...host '172.16.32.162' and port ''.
[Sep 29 15:13:18] DEBUG[15930]: chan_sip.c:8624 process_sdp: Processing session-level SDP c=IN IP4 172.16.32.162... OK.
[Sep 29 15:13:18] DEBUG[15930]: chan_sip.c:8624 process_sdp: Processing session-level SDP t=0 0... UNSUPPORTED.
[Sep 29 15:13:18] DEBUG[15930]: rtp_engine.c:536 ast_rtp_codecs_payloads_set_m_type: Setting payload 0 based on m type on 0xb560c588
[Sep 29 15:13:18] DEBUG[15930]: rtp_engine.c:536 ast_rtp_codecs_payloads_set_m_type: Setting payload 101 based on m type on 0xb560c588
[Sep 29 15:13:18] DEBUG[15930]: chan_sip.c:8811 process_sdp: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK.
[Sep 29 15:13:18] DEBUG[15930]: chan_sip.c:8811 process_sdp: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK.
[Sep 29 15:13:18] DEBUG[15930]: chan_sip.c:8811 process_sdp: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED.
[Sep 29 15:13:18] DEBUG[15930]: chan_sip.c:8811 process_sdp: Processing media-level (audio) SDP a=ptime:30... OK.
[Sep 29 15:13:18] DEBUG[15930]: chan_sip.c:8811 process_sdp: Processing media-level (audio) SDP a=sendrecv... OK.
[Sep 29 15:13:18] DEBUG[15930]: rtp_engine.c:639 ast_rtp_codecs_payload_formats: Incorporating payload 0 on 0xb560c588
[Sep 29 15:13:18] DEBUG[15930]: rtp_engine.c:639 ast_rtp_codecs_payload_formats: Incorporating payload 101 on 0xb560c588
[Sep 29 15:13:18] DEBUG[15930]: res_rtp_asterisk.c:2415 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0xa7e19a8'
[Sep 29 15:13:18] DEBUG[15930]: rtp_engine.c:517 ast_rtp_codecs_payloads_copy: Copying payload 0 from 0xb560c588 to 0xa7e1b54
[Sep 29 15:13:18] DEBUG[15930]: rtp_engine.c:517 ast_rtp_codecs_payloads_copy: Copying payload 101 from 0xb560c588 to 0xa7e1b54
[Sep 29 15:13:18] DEBUG[15930]: chan_sip.c:9043 process_sdp: We're settling with these formats: 0x4 (ulaw)
[Sep 29 15:13:18] DEBUG[15930]: chan_sip.c:9048 process_sdp: We have an owner, now see if we need to change this call
[Sep 29 15:13:18] DEBUG[15930]: chan_sip.c:5748 update_call_counter: Updating call counter for outgoing call
[Sep 29 15:13:18] DEBUG[15930]: chan_sip.c:13741 build_route: build_route: Contact hop: "mytel" <sip:1777@172.16.32.162:5060>
[Sep 29 15:13:18] DEBUG[15930]: netsock2.c:134 ast_sockaddr_split_hostport: Splitting '172.16.32.162:5060' into...
[Sep 29 15:13:18] DEBUG[15930]: netsock2.c:188 ast_sockaddr_split_hostport: ...host '172.16.32.162' and port '5060'.
[Sep 29 15:13:18] DEBUG[15930]: chan_sip.c:3323 __sip_xmit: Trying to put 'ACK sip:177' onto UDP socket destined for x.x.x.99:62863
    -- Connected line update to SIP/1011-00000000 prevented.
    -- SIP/1777-00000001 answered SIP/1011-00000000
[Sep 29 15:13:18] DEBUG[15964]: chan_sip.c:6322 sip_answer: SIP answering channel: SIP/1011-00000000
[Sep 29 15:13:18] DEBUG[15964]: res_rtp_asterisk.c:749 ast_rtp_update_source: Setting the marker bit due to a source update
[Sep 29 15:13:18] DEBUG[15964]: chan_sip.c:11358 transmit_response_with_sdp: Setting framing from config on incoming call
[Sep 29 15:13:18] DEBUG[15964]: chan_sip.c:11004 add_sdp: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True
[Sep 29 15:13:18] DEBUG[15964]: chan_sip.c:11005 add_sdp: ** Our prefcodec: 0x0 (nothing) 
[Sep 29 15:13:18] DEBUG[15964]: chan_sip.c:11114 add_sdp: -- Done with adding codecs to SDP
[Sep 29 15:13:18] DEBUG[15964]: chan_sip.c:11253 add_sdp: Done building SDP. Settling with this capability: 0xc (ulaw|alaw)
[Sep 29 15:13:18] DEBUG[15964]: chan_sip.c:3323 __sip_xmit: Trying to put 'SIP/2.0 200' onto UDP socket destined for x.x.x.99:1541
[Sep 29 15:13:18] DEBUG[15964]: features.c:3551 ast_bridge_call: bridge answer set, chan answer set
[Sep 29 15:13:18] DEBUG[15964]: features.c:3399 clear_dialed_interfaces: Removing dialed interfaces datastore on SIP/1777-00000001 since we're bridging
[Sep 29 15:13:18] DEBUG[15964]: res_rtp_asterisk.c:749 ast_rtp_update_source: Setting the marker bit due to a source update
[Sep 29 15:13:18] DEBUG[15964]: res_rtp_asterisk.c:749 ast_rtp_update_source: Setting the marker bit due to a source update
[Sep 29 15:13:18] DEBUG[15922]: devicestate.c:340 _ast_device_state: No provider found, checking channel drivers for SIP - 1777
[Sep 29 15:13:18] DEBUG[15922]: chan_sip.c:25615 sip_devicestate: Checking device state for peer 1777
[Sep 29 15:13:18] DEBUG[15922]: devicestate.c:458 do_state_change: Changing state for SIP/1777 - state 2 (In use)
[Sep 29 15:13:18] DEBUG[15922]: devicestate.c:438 devstate_event: device 'SIP/1777' state '2'
[Sep 29 15:13:18] DEBUG[15922]: devicestate.c:340 _ast_device_state: No provider found, checking channel drivers for SIP - 1777
[Sep 29 15:13:18] DEBUG[15922]: chan_sip.c:25615 sip_devicestate: Checking device state for peer 1777
[Sep 29 15:13:18] DEBUG[15922]: devicestate.c:458 do_state_change: Changing state for SIP/1777 - state 2 (In use)
[Sep 29 15:13:18] DEBUG[15922]: devicestate.c:438 devstate_event: device 'SIP/1777' state '2'
[Sep 29 15:13:18] DEBUG[15922]: devicestate.c:340 _ast_device_state: No provider found, checking channel drivers for SIP - 1011
[Sep 29 15:13:18] DEBUG[15922]: chan_sip.c:25615 sip_devicestate: Checking device state for peer 1011
[Sep 29 15:13:18] DEBUG[15922]: devicestate.c:458 do_state_change: Changing state for SIP/1011 - state 2 (In use)
[Sep 29 15:13:18] DEBUG[15922]: devicestate.c:438 devstate_event: device 'SIP/1011' state '2'
[Sep 29 15:13:18] DEBUG[15923]: app_queue.c:1586 extension_state_cb: Extension '1777@ext-local' changed to state '2' (In use) but we don't care because they're not a member of any queue.
[Sep 29 15:13:18] DEBUG[15957]: app_queue.c:1491 handle_statechange: Device 'SIP/1777' changed to state '2' (In use) but we don't care because they're not a member of any queue.
[Sep 29 15:13:18] DEBUG[15957]: app_queue.c:1491 handle_statechange: Device 'SIP/1777' changed to state '2' (In use) but we don't care because they're not a member of any queue.
[Sep 29 15:13:18] DEBUG[15957]: app_queue.c:1491 handle_statechange: Device 'SIP/1011' changed to state '2' (In use) but we don't care because they're not a member of any queue.
[Sep 29 15:13:18] DEBUG[15930]: chan_sip.c:7833 find_call: = Looking for  Call ID: 38834b38-5791-4aeb-8b94-52115680fe9f (Checking From) --From tag d82313b7-d564-4a42-a21b-fdc3a9e2afc9 --To-tag as3f5b34a7  
[Sep 29 15:13:18] DEBUG[15930]: chan_sip.c:24391 handle_incoming: **** Received ACK (6) - Command in SIP ACK
[Sep 29 15:13:18] DEBUG[15930]: chan_sip.c:4009 __sip_ack: Stopping retransmission on '38834b38-5791-4aeb-8b94-52115680fe9f' of Response 11739: Match Found
[Sep 29 15:13:18] DEBUG[15964]: res_rtp_asterisk.c:1263 ast_rtp_write: Ooh, format changed from unknown to ulaw
[Sep 29 15:13:18] DEBUG[15964]: res_rtp_asterisk.c:1294 ast_rtp_write: Created smoother: format: ulaw ms: 20 len: 160
[Sep 29 15:13:18] DEBUG[15964]: res_rtp_asterisk.c:1164 ast_rtp_raw_write: Starting RTCP transmission on RTP instance '0xa7e19a8'
[Sep 29 15:13:18] DEBUG[15964]: res_rtp_asterisk.c:2415 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0xa7e19a8'
[Sep 29 15:13:18] DEBUG[15964]: res_rtp_asterisk.c:2097 ast_rtp_read: RTP NAT: Got audio from other end. Now sending to address x.x.x.99:48206
[Sep 29 15:13:18] DEBUG[15964]: res_rtp_asterisk.c:1263 ast_rtp_write: Ooh, format changed from unknown to ulaw
[Sep 29 15:13:18] DEBUG[15964]: res_rtp_asterisk.c:1294 ast_rtp_write: Created smoother: format: ulaw ms: 20 len: 160
[Sep 29 15:13:22] DEBUG[15964]: res_rtp_asterisk.c:1697 ast_rtcp_read: Got RTCP report of 36 bytes
[Sep 29 15:13:24] DEBUG[15930]: chan_sip.c:7513 sip_alloc: Allocating new SIP dialog for 18e361395d3e020a123f2e20604626be@x.x.x.134:0 - OPTIONS (No RTP)
[Sep 29 15:13:24] DEBUG[15930]: acl.c:725 ast_ouraddrfor: For destination '172.16.32.206', our source address is 'x.x.x.134'.
[Sep 29 15:13:24] DEBUG[15930]: chan_sip.c:3477 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_UDP with address x.x.x.134:5060
[Sep 29 15:13:24] DEBUG[15930]: chan_sip.c:3052 initialize_initreq: Initializing initreq for method OPTIONS - callid 6efdd3fc1ccef78c5a7a6812025e0ccf@x.x.x.134:5060
[Sep 29 15:13:24] DEBUG[15930]: chan_sip.c:3323 __sip_xmit: Trying to put 'OPTIONS sip' onto UDP socket destined for 172.16.32.206:5060
[Sep 29 15:13:25] DEBUG[15930]: chan_sip.c:3323 __sip_xmit: Trying to put 'OPTIONS sip' onto UDP socket destined for 172.16.32.206:5060
[Sep 29 15:13:25] DEBUG[15930]: netsock2.c:134 ast_sockaddr_split_hostport: Splitting 'sipnet.ru' into...
[Sep 29 15:13:25] DEBUG[15930]: netsock2.c:188 ast_sockaddr_split_hostport: ...host 'sipnet.ru' and port ''.
       > doing dnsmgr_lookup for 'sipnet.ru'
[Sep 29 15:13:25] DEBUG[15930]: netsock2.c:134 ast_sockaddr_split_hostport: Splitting 'sipnet.ru' into...
[Sep 29 15:13:25] DEBUG[15930]: netsock2.c:188 ast_sockaddr_split_hostport: ...host 'sipnet.ru' and port ''.
[Sep 29 15:13:25] DEBUG[15930]: chan_sip.c:7513 sip_alloc: Allocating new SIP dialog for 68378cf5530fc0f06fac883d716059a0@sipnet.ru - REGISTER (No RTP)
[Sep 29 15:13:25] DEBUG[15930]: chan_sip.c:3014 registry_addref: SIP Registry sipnet.ru: refcount now 3
[Sep 29 15:13:25] DEBUG[15930]: acl.c:725 ast_ouraddrfor: For destination '212.53.40.40', our source address is 'x.x.x.134'.
[Sep 29 15:13:25] DEBUG[15930]: chan_sip.c:3477 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_UDP with address x.x.x.134:5060
[Sep 29 15:13:25] DEBUG[15930]: chan_sip.c:3014 registry_addref: SIP Registry sipnet.ru: refcount now 4
[Sep 29 15:13:25] DEBUG[15930]: chan_sip.c:12839 transmit_register: Scheduled a registration timeout for sipnet.ru id  #114 
[Sep 29 15:13:25] DEBUG[15930]: netsock2.c:134 ast_sockaddr_split_hostport: Splitting 'sipnet.ru' into...
[Sep 29 15:13:25] DEBUG[15930]: netsock2.c:188 ast_sockaddr_split_hostport: ...host 'sipnet.ru' and port ''.
[Sep 29 15:13:25] DEBUG[15930]: netsock2.c:134 ast_sockaddr_split_hostport: Splitting 'sipnet.ru' into...
[Sep 29 15:13:25] DEBUG[15930]: netsock2.c:188 ast_sockaddr_split_hostport: ...host 'sipnet.ru' and port ''.
[Sep 29 15:13:25] DEBUG[15930]: chan_sip.c:3052 initialize_initreq: Initializing initreq for method REGISTER - callid 68378cf5530fc0f06fac883d716059a0@sipnet.ru
[Sep 29 15:13:25] DEBUG[15930]: chan_sip.c:12915 transmit_register: REGISTER attempt 1 to 0031754852@sipnet.ru
[Sep 29 15:13:25] DEBUG[15930]: chan_sip.c:3323 __sip_xmit: Trying to put 'REGISTER si' onto UDP socket destined for 212.53.40.40:5060
[Sep 29 15:13:25] DEBUG[15930]: chan_sip.c:3006 registry_unref: SIP Registry sipnet.ru: refcount now 3
[Sep 29 15:13:25] DEBUG[15930]: chan_sip.c:7833 find_call: = Looking for  Call ID: 68378cf5530fc0f06fac883d716059a0@sipnet.ru (Checking To) --From tag as2f237f6e --To-tag AABDBB6A  
[Sep 29 15:13:25] DEBUG[15930]: chan_sip.c:4009 __sip_ack: Stopping retransmission on '68378cf5530fc0f06fac883d716059a0@sipnet.ru' of Request 104: Match Found
[Sep 29 15:13:25] DEBUG[15930]: chan_sip.c:20073 handle_response_register: Registration successful
[Sep 29 15:13:25] DEBUG[15930]: chan_sip.c:20075 handle_response_register: Cancelling timeout 114
[Sep 29 15:13:25] DEBUG[15930]: chan_sip.c:3006 registry_unref: SIP Registry sipnet.ru: refcount now 2
[Sep 29 15:13:25] DEBUG[15930]: chan_sip.c:3006 registry_unref: SIP Registry sipnet.ru: refcount now 1
[Sep 29 15:13:25] DEBUG[15930]: chan_sip.c:3014 registry_addref: SIP Registry sipnet.ru: refcount now 2
[Sep 29 15:13:26] DEBUG[15930]: chan_sip.c:3323 __sip_xmit: Trying to put 'OPTIONS sip' onto UDP socket destined for 172.16.32.206:5060
[Sep 29 15:13:27] DEBUG[15930]: chan_sip.c:3323 __sip_xmit: Trying to put 'OPTIONS sip' onto UDP socket destined for 172.16.32.206:5060
[Sep 29 15:13:28] DEBUG[15930]: chan_sip.c:3323 __sip_xmit: Trying to put 'OPTIONS sip' onto UDP socket destined for 172.16.32.206:5060
[Sep 29 15:13:28] DEBUG[15930]: chan_sip.c:5896 sip_destroy: Destroying SIP dialog 6efdd3fc1ccef78c5a7a6812025e0ccf@x.x.x.134:5060
[Sep 29 15:13:29] DEBUG[15930]: chan_sip.c:7833 find_call: = Looking for  Call ID: 6a33c3f719d1eb1a287e7720226014fc@x.x.x.134:5060 (Checking From) --From tag 6cf83e6ce6414de3i1 --To-tag as0ef8ec24  
[Sep 29 15:13:29] DEBUG[15930]: chan_sip.c:24391 handle_incoming: **** Received INFO (13) - Command in SIP INFO
[Sep 29 15:13:29] DEBUG[15930]: chan_sip.c:3323 __sip_xmit: Trying to put 'SIP/2.0 200' onto UDP socket destined for x.x.x.99:62863
[Sep 29 15:13:29] DTMF[15964]: channel.c:3957 __ast_read: DTMF end '#' received on SIP/1777-00000001, duration 100 ms
[Sep 29 15:13:29] DTMF[15964]: channel.c:3983 __ast_read: DTMF begin emulation of '#' with duration 100 queued on SIP/1777-00000001
[Sep 29 15:13:29] DEBUG[15964]: channel.c:7001 ast_generic_bridge: Got DTMF begin on channel (SIP/1777-00000001)
[Sep 29 15:13:29] DEBUG[15964]: res_rtp_asterisk.c:749 ast_rtp_update_source: Setting the marker bit due to a source update
[Sep 29 15:13:29] DEBUG[15964]: res_rtp_asterisk.c:749 ast_rtp_update_source: Setting the marker bit due to a source update
[Sep 29 15:13:29] DEBUG[15964]: channel.c:7394 ast_channel_bridge: Bridge stops bridging channels SIP/1011-00000000 and SIP/1777-00000001
[Sep 29 15:13:29] DEBUG[15964]: features.c:3766 ast_bridge_call: Not passing DTMF through, since it may be a feature code
[Sep 29 15:13:29] DEBUG[15964]: res_rtp_asterisk.c:749 ast_rtp_update_source: Setting the marker bit due to a source update
[Sep 29 15:13:29] DEBUG[15964]: res_rtp_asterisk.c:749 ast_rtp_update_source: Setting the marker bit due to a source update
[Sep 29 15:13:29] DTMF[15964]: channel.c:4119 __ast_read: DTMF end emulation of '#' queued on SIP/1777-00000001
[Sep 29 15:13:29] DEBUG[15964]: channel.c:7001 ast_generic_bridge: Got DTMF end on channel (SIP/1777-00000001)
[Sep 29 15:13:29] DEBUG[15964]: res_rtp_asterisk.c:749 ast_rtp_update_source: Setting the marker bit due to a source update
[Sep 29 15:13:29] DEBUG[15964]: res_rtp_asterisk.c:749 ast_rtp_update_source: Setting the marker bit due to a source update
[Sep 29 15:13:29] DEBUG[15964]: channel.c:7394 ast_channel_bridge: Bridge stops bridging channels SIP/1011-00000000 and SIP/1777-00000001
[Sep 29 15:13:29] DEBUG[15964]: features.c:2871 feature_interpret: Feature interpret: chan=SIP/1011-00000000, peer=SIP/1777-00000001, code=#, sense=2, features=2, dynamic=#
[Sep 29 15:13:29] DEBUG[15964]: features.c:3827 ast_bridge_call: Set feature timer to 1000 ms
[Sep 29 15:13:29] DEBUG[15964]: res_rtp_asterisk.c:749 ast_rtp_update_source: Setting the marker bit due to a source update
[Sep 29 15:13:29] DEBUG[15964]: res_rtp_asterisk.c:749 ast_rtp_update_source: Setting the marker bit due to a source update
[Sep 29 15:13:29] DEBUG[15937]: chan_iax2.c:11729 iax2_do_register: Allocate call number
[Sep 29 15:13:29] DEBUG[15937]: chan_iax2.c:2366 peercnt_add: ip callno count incremented to 3 for x.x.x.98
[Sep 29 15:13:29] DEBUG[15937]: chan_iax2.c:11738 iax2_do_register: Registration created on call 410
[Sep 29 15:13:29] DEBUG[15942]: chan_iax2.c:2714 sched_delay_remove: schedule decrement of callno used for x.x.x.98 in 60 seconds
[Sep 29 15:13:29] DEBUG[15964]: res_rtp_asterisk.c:1697 ast_rtcp_read: Got RTCP report of 36 bytes
[Sep 29 15:13:30] DEBUG[15964]: res_rtp_asterisk.c:749 ast_rtp_update_source: Setting the marker bit due to a source update
[Sep 29 15:13:30] DEBUG[15964]: res_rtp_asterisk.c:749 ast_rtp_update_source: Setting the marker bit due to a source update
[Sep 29 15:13:30] DEBUG[15964]: channel.c:7394 ast_channel_bridge: Bridge stops bridging channels SIP/1011-00000000 and SIP/1777-00000001
[Sep 29 15:13:30] DEBUG[15964]: features.c:3640 ast_bridge_call: Timed out for feature!
[Sep 29 15:13:30] DEBUG[15964]: res_rtp_asterisk.c:749 ast_rtp_update_source: Setting the marker bit due to a source update
[Sep 29 15:13:30] DEBUG[15964]: res_rtp_asterisk.c:749 ast_rtp_update_source: Setting the marker bit due to a source update
[Sep 29 15:13:30] DEBUG[15930]: chan_sip.c:7833 find_call: = Looking for  Call ID: 6a33c3f719d1eb1a287e7720226014fc@x.x.x.134:5060 (Checking From) --From tag 6cf83e6ce6414de3i1 --To-tag as0ef8ec24  
[Sep 29 15:13:30] DEBUG[15930]: chan_sip.c:24391 handle_incoming: **** Received INFO (13) - Command in SIP INFO
[Sep 29 15:13:30] DEBUG[15930]: chan_sip.c:3323 __sip_xmit: Trying to put 'SIP/2.0 200' onto UDP socket destined for x.x.x.99:62863
[Sep 29 15:13:30] DTMF[15964]: channel.c:3957 __ast_read: DTMF end '#' received on SIP/1777-00000001, duration 100 ms
[Sep 29 15:13:30] DTMF[15964]: channel.c:3983 __ast_read: DTMF begin emulation of '#' with duration 100 queued on SIP/1777-00000001
[Sep 29 15:13:30] DEBUG[15964]: channel.c:7001 ast_generic_bridge: Got DTMF begin on channel (SIP/1777-00000001)
[Sep 29 15:13:30] DEBUG[15964]: res_rtp_asterisk.c:749 ast_rtp_update_source: Setting the marker bit due to a source update
[Sep 29 15:13:30] DEBUG[15964]: res_rtp_asterisk.c:749 ast_rtp_update_source: Setting the marker bit due to a source update
[Sep 29 15:13:30] DEBUG[15964]: channel.c:7394 ast_channel_bridge: Bridge stops bridging channels SIP/1011-00000000 and SIP/1777-00000001
[Sep 29 15:13:30] DEBUG[15964]: features.c:3766 ast_bridge_call: Not passing DTMF through, since it may be a feature code
[Sep 29 15:13:30] DEBUG[15964]: res_rtp_asterisk.c:749 ast_rtp_update_source: Setting the marker bit due to a source update
[Sep 29 15:13:30] DEBUG[15964]: res_rtp_asterisk.c:749 ast_rtp_update_source: Setting the marker bit due to a source update
[Sep 29 15:13:31] DTMF[15964]: channel.c:4119 __ast_read: DTMF end emulation of '#' queued on SIP/1777-00000001
[Sep 29 15:13:31] DEBUG[15964]: channel.c:7001 ast_generic_bridge: Got DTMF end on channel (SIP/1777-00000001)
[Sep 29 15:13:31] DEBUG[15964]: res_rtp_asterisk.c:749 ast_rtp_update_source: Setting the marker bit due to a source update
[Sep 29 15:13:31] DEBUG[15964]: res_rtp_asterisk.c:749 ast_rtp_update_source: Setting the marker bit due to a source update
[Sep 29 15:13:31] DEBUG[15964]: channel.c:7394 ast_channel_bridge: Bridge stops bridging channels SIP/1011-00000000 and SIP/1777-00000001
[Sep 29 15:13:31] DEBUG[15964]: features.c:2871 feature_interpret: Feature interpret: chan=SIP/1011-00000000, peer=SIP/1777-00000001, code=#, sense=2, features=2, dynamic=#
[Sep 29 15:13:31] DEBUG[15964]: features.c:3827 ast_bridge_call: Set feature timer to 1000 ms
[Sep 29 15:13:31] DEBUG[15964]: res_rtp_asterisk.c:749 ast_rtp_update_source: Setting the marker bit due to a source update
[Sep 29 15:13:31] DEBUG[15964]: res_rtp_asterisk.c:749 ast_rtp_update_source: Setting the marker bit due to a source update
[Sep 29 15:13:31] DEBUG[15930]: chan_sip.c:7833 find_call: = Looking for  Call ID: 6a33c3f719d1eb1a287e7720226014fc@x.x.x.134:5060 (Checking From) --From tag 6cf83e6ce6414de3i1 --To-tag as0ef8ec24  
[Sep 29 15:13:31] DEBUG[15930]: chan_sip.c:24391 handle_incoming: **** Received INFO (13) - Command in SIP INFO
[Sep 29 15:13:31] DEBUG[15930]: chan_sip.c:3323 __sip_xmit: Trying to put 'SIP/2.0 200' onto UDP socket destined for x.x.x.99:62863
[Sep 29 15:13:31] DTMF[15964]: channel.c:3957 __ast_read: DTMF end '#' received on SIP/1777-00000001, duration 100 ms
[Sep 29 15:13:31] DTMF[15964]: channel.c:3983 __ast_read: DTMF begin emulation of '#' with duration 100 queued on SIP/1777-00000001
[Sep 29 15:13:31] DEBUG[15964]: channel.c:7001 ast_generic_bridge: Got DTMF begin on channel (SIP/1777-00000001)
[Sep 29 15:13:31] DEBUG[15964]: res_rtp_asterisk.c:749 ast_rtp_update_source: Setting the marker bit due to a source update
[Sep 29 15:13:31] DEBUG[15964]: res_rtp_asterisk.c:749 ast_rtp_update_source: Setting the marker bit due to a source update
[Sep 29 15:13:31] DEBUG[15964]: channel.c:7394 ast_channel_bridge: Bridge stops bridging channels SIP/1011-00000000 and SIP/1777-00000001
[Sep 29 15:13:31] DEBUG[15964]: features.c:3766 ast_bridge_call: Not passing DTMF through, since it may be a feature code
[Sep 29 15:13:31] DEBUG[15964]: res_rtp_asterisk.c:749 ast_rtp_update_source: Setting the marker bit due to a source update
[Sep 29 15:13:31] DEBUG[15964]: res_rtp_asterisk.c:749 ast_rtp_update_source: Setting the marker bit due to a source update
[Sep 29 15:13:31] DTMF[15964]: channel.c:4119 __ast_read: DTMF end emulation of '#' queued on SIP/1777-00000001
[Sep 29 15:13:31] DEBUG[15964]: channel.c:7001 ast_generic_bridge: Got DTMF end on channel (SIP/1777-00000001)
[Sep 29 15:13:31] DEBUG[15964]: res_rtp_asterisk.c:749 ast_rtp_update_source: Setting the marker bit due to a source update
[Sep 29 15:13:31] DEBUG[15964]: res_rtp_asterisk.c:749 ast_rtp_update_source: Setting the marker bit due to a source update
[Sep 29 15:13:31] DEBUG[15964]: channel.c:7394 ast_channel_bridge: Bridge stops bridging channels SIP/1011-00000000 and SIP/1777-00000001
[Sep 29 15:13:31] DEBUG[15964]: features.c:2871 feature_interpret: Feature interpret: chan=SIP/1011-00000000, peer=SIP/1777-00000001, code=##, sense=2, features=2, dynamic=#
[Sep 29 15:13:31] DEBUG[15964]: features.c:2755 feature_interpret_helper: Feature detected: fname=Blind Transfer sname=blindxfer exten=##
[Sep 29 15:13:31] DEBUG[15964]: res_rtp_asterisk.c:749 ast_rtp_update_source: Setting the marker bit due to a source update
    -- Started music on hold, class 'default', on SIP/1011-00000000
[Sep 29 15:13:31] DEBUG[15964]: channel.c:3480 ast_settimeout: Scheduling timer at (50 requested / 50 actual) timer ticks per second
[Sep 29 15:13:31] DEBUG[15964]: channel.c:5107 set_format: Set channel SIP/1777-00000001 to write format gsm
[Sep 29 15:13:31] DEBUG[15964]: channel.c:3480 ast_settimeout: Scheduling timer at (50 requested / 50 actual) timer ticks per second
    -- <SIP/1777-00000001> Playing 'pbx-transfer.gsm' (language 'ru')
[Sep 29 15:13:31] DEBUG[15965]: channel.c:5107 set_format: Set channel SIP/1011-00000000 to write format slin
[Sep 29 15:13:31] DEBUG[15965]: res_musiconhold.c:338 ast_moh_files_next: SIP/1011-00000000 Opened file 0 '/var/lib/asterisk/mohmp3/macroform-cold_day'
[Sep 29 15:13:31] DEBUG[15965]: res_rtp_asterisk.c:1085 ast_rtp_raw_write: Difference is 1360, ms is 190
[Sep 29 15:13:31] DEBUG[15964]: channel.c:3480 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[Sep 29 15:13:31] DEBUG[15964]: channel.c:3480 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[Sep 29 15:13:31] DEBUG[15964]: channel.c:3480 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[Sep 29 15:13:31] DEBUG[15964]: channel.c:5107 set_format: Set channel SIP/1777-00000001 to write format ulaw
[Sep 29 15:13:31] DEBUG[15964]: channel.c:5107 set_format: Set channel SIP/1777-00000001 to write format slin
[Sep 29 15:13:31] DEBUG[15964]: channel.c:3480 ast_settimeout: Scheduling timer at (50 requested / 50 actual) timer ticks per second
[Sep 29 15:13:32] DEBUG[15930]: chan_sip.c:7833 find_call: = Looking for  Call ID: 6a33c3f719d1eb1a287e7720226014fc@x.x.x.134:5060 (Checking From) --From tag 6cf83e6ce6414de3i1 --To-tag as0ef8ec24  
[Sep 29 15:13:32] DEBUG[15930]: chan_sip.c:24391 handle_incoming: **** Received INFO (13) - Command in SIP INFO
[Sep 29 15:13:32] DEBUG[15930]: chan_sip.c:3323 __sip_xmit: Trying to put 'SIP/2.0 200' onto UDP socket destined for x.x.x.99:62863
[Sep 29 15:13:32] DTMF[15964]: channel.c:3957 __ast_read: DTMF end '1' received on SIP/1777-00000001, duration 100 ms
[Sep 29 15:13:32] DTMF[15964]: channel.c:4026 __ast_read: DTMF end passthrough '1' on SIP/1777-00000001
[Sep 29 15:13:32] DEBUG[15964]: channel.c:5107 set_format: Set channel SIP/1777-00000001 to write format ulaw
[Sep 29 15:13:32] DEBUG[15964]: channel.c:3480 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[Sep 29 15:13:33] DEBUG[15930]: chan_sip.c:7833 find_call: = Looking for  Call ID: 6a33c3f719d1eb1a287e7720226014fc@x.x.x.134:5060 (Checking From) --From tag 6cf83e6ce6414de3i1 --To-tag as0ef8ec24  
[Sep 29 15:13:33] DEBUG[15930]: chan_sip.c:24391 handle_incoming: **** Received INFO (13) - Command in SIP INFO
[Sep 29 15:13:33] DEBUG[15930]: chan_sip.c:3323 __sip_xmit: Trying to put 'SIP/2.0 200' onto UDP socket destined for x.x.x.99:62863
[Sep 29 15:13:33] DTMF[15964]: channel.c:3957 __ast_read: DTMF end '0' received on SIP/1777-00000001, duration 100 ms
[Sep 29 15:13:33] DTMF[15964]: channel.c:4026 __ast_read: DTMF end passthrough '0' on SIP/1777-00000001
[Sep 29 15:13:33] DEBUG[15930]: chan_sip.c:7833 find_call: = Looking for  Call ID: 6a33c3f719d1eb1a287e7720226014fc@x.x.x.134:5060 (Checking From) --From tag 6cf83e6ce6414de3i1 --To-tag as0ef8ec24  
[Sep 29 15:13:33] DEBUG[15930]: chan_sip.c:24391 handle_incoming: **** Received INFO (13) - Command in SIP INFO
[Sep 29 15:13:33] DEBUG[15930]: chan_sip.c:3323 __sip_xmit: Trying to put 'SIP/2.0 200' onto UDP socket destined for x.x.x.99:62863
[Sep 29 15:13:33] DTMF[15964]: channel.c:3957 __ast_read: DTMF end '6' received on SIP/1777-00000001, duration 100 ms
[Sep 29 15:13:33] DTMF[15964]: channel.c:4026 __ast_read: DTMF end passthrough '6' on SIP/1777-00000001
[Sep 29 15:13:34] DEBUG[15930]: chan_sip.c:7833 find_call: = Looking for  Call ID: 6a33c3f719d1eb1a287e7720226014fc@x.x.x.134:5060 (Checking From) --From tag 6cf83e6ce6414de3i1 --To-tag as0ef8ec24  
[Sep 29 15:13:34] DEBUG[15930]: chan_sip.c:24391 handle_incoming: **** Received INFO (13) - Command in SIP INFO
[Sep 29 15:13:34] DEBUG[15930]: chan_sip.c:3323 __sip_xmit: Trying to put 'SIP/2.0 200' onto UDP socket destined for x.x.x.99:62863
[Sep 29 15:13:34] DTMF[15964]: channel.c:3957 __ast_read: DTMF end '0' received on SIP/1777-00000001, duration 100 ms
[Sep 29 15:13:34] DTMF[15964]: channel.c:4026 __ast_read: DTMF end passthrough '0' on SIP/1777-00000001


[Sep 29 15:13:34] DEBUG[15965]: res_rtp_asterisk.c:1697 ast_rtcp_read: Got RTCP report of 36 bytes
[Sep 29 15:13:37] DEBUG[15964]: res_rtp_asterisk.c:749 ast_rtp_update_source: Setting the marker bit due to a source update
    -- Stopped music on hold on SIP/1011-00000000
[Sep 29 15:13:37] DEBUG[15964]: channel.c:5107 set_format: Set channel SIP/1011-00000000 to write format ulaw
[Sep 29 15:13:37] DEBUG[15964]: channel.c:3480 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[Sep 29 15:13:37] DEBUG[15964]: features.c:1911 builtin_blindtransfer: transferer=SIP/1777-00000001; transferee=SIP/1011-00000000; lastapp=; lastdata=; chan=SIP/1777-00000001; dstchan=
[Sep 29 15:13:37] DEBUG[15964]: features.c:1914 builtin_blindtransfer: TRANSFEREE; lastapp=Dial; lastdata=SIP/1777,,trI, chan=SIP/1011-00000000; dstchan=SIP/1777-00000001
[Sep 29 15:13:37] DEBUG[15964]: features.c:1916 builtin_blindtransfer: transferer_real_context=from-internal-xfer; xferto=1060
[Sep 29 15:13:37] DEBUG[15964]: features.c:1930 builtin_blindtransfer: ABOUT TO AST_ASYNC_GOTO, have a pbx... set HANGUP_DONT on chan=SIP/1011-00000000
[Sep 29 15:13:37] DEBUG[15964]: channel.c:2807 ast_hangup: Hanging up channel 'SIP/1777-00000001'
[Sep 29 15:13:37] DEBUG[15964]: chan_sip.c:6121 sip_hangup: Hangup call SIP/1777-00000001, SIP callid 6a33c3f719d1eb1a287e7720226014fc@x.x.x.134:5060
[Sep 29 15:13:37] DEBUG[15964]: chan_sip.c:5748 update_call_counter: Updating call counter for outgoing call
[Sep 29 15:13:37] DEBUG[15964]: res_rtp_asterisk.c:2415 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0xa7e19a8'
[Sep 29 15:13:37] DEBUG[15964]: netsock2.c:134 ast_sockaddr_split_hostport: Splitting '172.16.32.162:5060' into...
[Sep 29 15:13:37] DEBUG[15964]: netsock2.c:188 ast_sockaddr_split_hostport: ...host '172.16.32.162' and port '5060'.
[Sep 29 15:13:37] DEBUG[15964]: chan_sip.c:3323 __sip_xmit: Trying to put 'BYE sip:177' onto UDP socket destined for x.x.x.99:62863
[Sep 29 15:13:37] DEBUG[15922]: devicestate.c:340 _ast_device_state: No provider found, checking channel drivers for SIP - 1777
[Sep 29 15:13:37] DEBUG[15922]: chan_sip.c:25615 sip_devicestate: Checking device state for peer 1777
[Sep 29 15:13:37] DEBUG[15922]: devicestate.c:458 do_state_change: Changing state for SIP/1777 - state 1 (Not in use)
[Sep 29 15:13:37] DEBUG[15922]: devicestate.c:438 devstate_event: device 'SIP/1777' state '1'
[Sep 29 15:13:37] DEBUG[15923]: app_queue.c:1586 extension_state_cb: Extension '1777@ext-local' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Sep 29 15:13:37] DEBUG[15957]: app_queue.c:1491 handle_statechange: Device 'SIP/1777' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Sep 29 15:13:37] DEBUG[15964]: cdr_mysql.c:336 mysql_log: Inserting a CDR record.
[Sep 29 15:13:37] DEBUG[15964]: cdr_mysql.c:339 mysql_log: SQL command as follows: INSERT INTO cdr (`calldate`,`clid`,`src`,`dst`,`dcontext`,`channel`,`dstchannel`,`lastapp`,`lastdata`,`duration`,`billsec`,`disposition`,`amaflags`,`uniqueid`) VALUES ('2011-09-29 15:13:15','\"mynout\" <1011>','1011','1777','from-internal','SIP/1011-00000000','SIP/1777-00000001','Dial','SIP/1777,,trI','22','19','ANSWERED','3','1317294795.0')
[Sep 29 15:13:37] DEBUG[15964]: cdr_radius.c:208 radius_log: Unable to create RADIUS record. CDR not recorded!
[Sep 29 15:13:37] DEBUG[15964]: res_config_sqlite.c:684 find_table: About to query table structure: SELECT sql FROM sqlite_master WHERE type='table' AND tbl_name='ast_cdr'
[Sep 29 15:13:37] DEBUG[15964]: res_config_sqlite.c:613 find_table_cb: Found field: id		INTEGER
[Sep 29 15:13:37] DEBUG[15964]: res_config_sqlite.c:613 find_table_cb: Found field: clid		VARCHAR(80)	NOT NULL	DEFAULT ''
[Sep 29 15:13:37] DEBUG[15964]: res_config_sqlite.c:613 find_table_cb: Found field: src		VARCHAR(80)	NOT NULL	DEFAULT ''
[Sep 29 15:13:37] DEBUG[15964]: res_config_sqlite.c:613 find_table_cb: Found field: dst		VARCHAR(80)	NOT NULL	DEFAULT ''
[Sep 29 15:13:37] DEBUG[15964]: res_config_sqlite.c:613 find_table_cb: Found field: dcontext	VARCHAR(80)	NOT NULL	DEFAULT ''
[Sep 29 15:13:37] DEBUG[15964]: res_config_sqlite.c:613 find_table_cb: Found field: channel		VARCHAR(80)	NOT NULL	DEFAULT ''
[Sep 29 15:13:37] DEBUG[15964]: res_config_sqlite.c:613 find_table_cb: Found field: dstchannel	VARCHAR(80)	NOT NULL	DEFAULT ''
[Sep 29 15:13:37] DEBUG[15964]: res_config_sqlite.c:613 find_table_cb: Found field: lastapp		VARCHAR(80)	NOT NULL	DEFAULT ''
[Sep 29 15:13:37] DEBUG[15964]: res_config_sqlite.c:613 find_table_cb: Found field: lastdata	VARCHAR(80)	NOT NULL	DEFAULT ''
[Sep 29 15:13:37] DEBUG[15964]: res_config_sqlite.c:613 find_table_cb: Found field: start		DATETIME	NOT NULL	DEFAULT '0000-00-00 00:00:00'
[Sep 29 15:13:37] DEBUG[15964]: res_config_sqlite.c:613 find_table_cb: Found field: answer		DATETIME	NOT NULL	DEFAULT '0000-00-00 00:00:00'
[Sep 29 15:13:37] DEBUG[15964]: res_config_sqlite.c:613 find_table_cb: Found field: end		DATETIME	NOT NULL	DEFAULT '0000-00-00 00:00:00'
[Sep 29 15:13:37] DEBUG[15964]: res_config_sqlite.c:613 find_table_cb: Found field: duration	INT(11)		NOT NULL	DEFAULT 0
[Sep 29 15:13:37] DEBUG[15964]: res_config_sqlite.c:613 find_table_cb: Found field: billsec		INT(11)		NOT NULL	DEFAULT 0
[Sep 29 15:13:37] DEBUG[15964]: res_config_sqlite.c:613 find_table_cb: Found field: disposition	VARCHAR(45)	NOT NULL	DEFAULT ''
[Sep 29 15:13:37] DEBUG[15964]: res_config_sqlite.c:613 find_table_cb: Found field: amaflags	INT(11)		NOT NULL	DEFAULT 0
[Sep 29 15:13:37] DEBUG[15964]: res_config_sqlite.c:613 find_table_cb: Found field: accountcode	VARCHAR(20)	NOT NULL	DEFAULT ''
[Sep 29 15:13:37] DEBUG[15964]: res_config_sqlite.c:613 find_table_cb: Found field: uniqueid	VARCHAR(32)	NOT NULL	DEFAULT ''
[Sep 29 15:13:37] DEBUG[15964]: res_config_sqlite.c:613 find_table_cb: Found field: userfield	VARCHAR(255)	NOT NULL	DEFAULT ''
[Sep 29 15:13:37] DEBUG[15964]: res_config_sqlite.c:613 find_table_cb: Found field: PRIMARY KEY	(id)

[Sep 29 15:13:37] DEBUG[15964]: res_config_sqlite.c:834 cdr_handler: SQL query: INSERT INTO ast_cdr (clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,start,answer,end,duration,billsec,disposition,amaflags,uniqueid) VALUES ('"mynout" <1011>','1011','1777','from-internal','SIP/1011-00000000','SIP/1777-00000001','Dial','SIP/1777,,trI','2011-09-29 15:13:15','2011-09-29 15:13:18','2011-09-29 15:13:37','22','19','ANSWERED','DOCUMENTATION','1317294795.0')
[Sep 29 15:13:37] ERROR[15964]: res_config_sqlite.c:847 cdr_handler: unable to open database file
[Sep 29 15:13:37] DEBUG[15964]: app_dial.c:2886 dial_exec_full: Exiting with DIALSTATUS=ANSWER.
[Sep 29 15:13:37] DEBUG[15964]: app_macro.c:435 _macro_exec: Executed application: Dial
  == Channel 'SIP/1011-00000000' jumping out of macro 'dial-one'
[Sep 29 15:13:37] DEBUG[15964]: app_macro.c:435 _macro_exec: Executed application: Macro
  == Channel 'SIP/1011-00000000' jumping out of macro 'exten-vm'
[Sep 29 15:13:37] DEBUG[15964]: pbx.c:4101 pbx_extension_helper: Launching 'ResetCDR'
    -- Executing [1060@from-internal-xfer:1] ResetCDR("SIP/1011-00000000", "") in new stack
[Sep 29 15:13:37] DEBUG[15964]: pbx.c:4101 pbx_extension_helper: Launching 'NoCDR'
    -- Executing [1060@from-internal-xfer:2] NoCDR("SIP/1011-00000000", "") in new stack
[Sep 29 15:13:37] DEBUG[15964]: pbx.c:4101 pbx_extension_helper: Launching 'Progress'
    -- Executing [1060@from-internal-xfer:3] Progress("SIP/1011-00000000", "") in new stack
[Sep 29 15:13:37] DEBUG[15964]: pbx.c:4101 pbx_extension_helper: Launching 'Wait'
    -- Executing [1060@from-internal-xfer:4] Wait("SIP/1011-00000000", "1") in new stack
[Sep 29 15:13:37] DEBUG[15922]: devicestate.c:340 _ast_device_state: No provider found, checking channel drivers for SIP - 1777
[Sep 29 15:13:37] DEBUG[15922]: chan_sip.c:25615 sip_devicestate: Checking device state for peer 1777
[Sep 29 15:13:37] DEBUG[15922]: devicestate.c:458 do_state_change: Changing state for SIP/1777 - state 1 (Not in use)
[Sep 29 15:13:37] DEBUG[15922]: devicestate.c:438 devstate_event: device 'SIP/1777' state '1'
[Sep 29 15:13:37] DEBUG[15930]: chan_sip.c:7833 find_call: = Looking for  Call ID: 6a33c3f719d1eb1a287e7720226014fc@x.x.x.134:5060 (Checking To) --From tag as0ef8ec24 --To-tag 6cf83e6ce6414de3i1  
[Sep 29 15:13:37] DEBUG[15930]: chan_sip.c:4009 __sip_ack: Stopping retransmission on '6a33c3f719d1eb1a287e7720226014fc@x.x.x.134:5060' of Request 103: Match Found
[Sep 29 15:13:37] DEBUG[15930]: chan_sip.c:5896 sip_destroy: Destroying SIP dialog 6a33c3f719d1eb1a287e7720226014fc@x.x.x.134:5060
[Sep 29 15:13:37] DEBUG[15930]: rtp_engine.c:293 instance_destructor: Destroyed RTP instance '0xa7e19a8'
[Sep 29 15:13:37] DEBUG[15957]: app_queue.c:1491 handle_statechange: Device 'SIP/1777' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Sep 29 15:13:38] DEBUG[15964]: pbx.c:4101 pbx_extension_helper: Launching 'Progress'
    -- Executing [1060@from-internal-xfer:5] Progress("SIP/1011-00000000", "") in new stack
[Sep 29 15:13:38] DEBUG[15964]: pbx.c:4101 pbx_extension_helper: Launching 'Playback'
    -- Executing [1060@from-internal-xfer:6] Playback("SIP/1011-00000000", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack
[Sep 29 15:13:38] DEBUG[15964]: channel.c:5107 set_format: Set channel SIP/1011-00000000 to write format gsm
[Sep 29 15:13:38] DEBUG[15964]: res_rtp_asterisk.c:1085 ast_rtp_raw_write: Difference is 8272, ms is 1054
[Sep 29 15:13:38] DEBUG[15964]: channel.c:3480 ast_settimeout: Scheduling timer at (50 requested / 50 actual) timer ticks per second
    -- <SIP/1011-00000000> Playing 'silence/1.gsm' (language 'ru')
[Sep 29 15:13:38] DEBUG[15930]: chan_sip.c:7513 sip_alloc: Allocating new SIP dialog for 679bafab299336d1037fb6f043f93411@x.x.x.134:0 - OPTIONS (No RTP)
[Sep 29 15:13:38] DEBUG[15930]: acl.c:725 ast_ouraddrfor: For destination '172.16.32.206', our source address is 'x.x.x.134'.
[Sep 29 15:13:38] DEBUG[15930]: chan_sip.c:3477 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_UDP with address x.x.x.134:5060
[Sep 29 15:13:38] DEBUG[15930]: chan_sip.c:3052 initialize_initreq: Initializing initreq for method OPTIONS - callid 3af71ab02f03eb510a40e3fc5ca254cd@x.x.x.134:5060
[Sep 29 15:13:38] DEBUG[15930]: chan_sip.c:3323 __sip_xmit: Trying to put 'OPTIONS sip' onto UDP socket destined for 172.16.32.206:5060
[Sep 29 15:13:39] DEBUG[15964]: channel.c:3480 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[Sep 29 15:13:39] DEBUG[15964]: channel.c:3480 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[Sep 29 15:13:39] DEBUG[15964]: channel.c:3480 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[Sep 29 15:13:39] DEBUG[15964]: channel.c:5107 set_format: Set channel SIP/1011-00000000 to write format ulaw
[Sep 29 15:13:39] DEBUG[15964]: channel.c:5107 set_format: Set channel SIP/1011-00000000 to write format gsm
[Sep 29 15:13:39] DEBUG[15964]: channel.c:3480 ast_settimeout: Scheduling timer at (50 requested / 50 actual) timer ticks per second
    -- <SIP/1011-00000000> Playing 'cannot-complete-as-dialed.gsm' (language 'ru')
[Sep 29 15:13:39] DEBUG[15930]: chan_sip.c:3323 __sip_xmit: Trying to put 'OPTIONS sip' onto UDP socket destined for 172.16.32.206:5060
[Sep 29 15:13:39] DEBUG[15932]: chan_iax2.c:2396 peercnt_remove: ip callno count decremented to 2 for x.x.x.98
[Sep 29 15:13:40] DEBUG[15930]: chan_sip.c:3323 __sip_xmit: Trying to put 'OPTIONS sip' onto UDP socket destined for 172.16.32.206:5060
[Sep 29 15:13:41] DEBUG[15930]: chan_sip.c:3323 __sip_xmit: Trying to put 'OPTIONS sip' onto UDP socket destined for 172.16.32.206:5060
[Sep 29 15:13:41] DEBUG[15964]: channel.c:3480 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[Sep 29 15:13:41] DEBUG[15964]: channel.c:3480 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[Sep 29 15:13:41] DEBUG[15964]: channel.c:3480 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[Sep 29 15:13:41] DEBUG[15964]: channel.c:5107 set_format: Set channel SIP/1011-00000000 to write format ulaw
[Sep 29 15:13:41] DEBUG[15964]: channel.c:5107 set_format: Set channel SIP/1011-00000000 to write format gsm
[Sep 29 15:13:41] DEBUG[15964]: channel.c:3480 ast_settimeout: Scheduling timer at (50 requested / 50 actual) timer ticks per second
    -- <SIP/1011-00000000> Playing 'check-number-dial-again.gsm' (language 'ru')
[Sep 29 15:13:42] DEBUG[15930]: chan_sip.c:3867 __sip_autodestruct: Auto destroying SIP dialog 'c9bbe0e0-cf3e-433c-a00c-54a1b5407102'
[Sep 29 15:13:42] DEBUG[15930]: chan_sip.c:5896 sip_destroy: Destroying SIP dialog c9bbe0e0-cf3e-433c-a00c-54a1b5407102
[Sep 29 15:13:42] DEBUG[15930]: chan_sip.c:3323 __sip_xmit: Trying to put 'OPTIONS sip' onto UDP socket destined for 172.16.32.206:5060
[Sep 29 15:13:42] DEBUG[15930]: chan_sip.c:5896 sip_destroy: Destroying SIP dialog 3af71ab02f03eb510a40e3fc5ca254cd@x.x.x..134:5060
[Sep 29 15:13:42] DEBUG[15964]: res_rtp_asterisk.c:1697 ast_rtcp_read: Got RTCP report of 36 bytes
[Sep 29 15:13:44] DEBUG[15964]: channel.c:3480 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[Sep 29 15:13:44] DEBUG[15964]: channel.c:3480 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[Sep 29 15:13:44] DEBUG[15964]: channel.c:3480 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[Sep 29 15:13:44] DEBUG[15964]: channel.c:5107 set_format: Set channel SIP/1011-00000000 to write format ulaw
[Sep 29 15:13:44] DEBUG[15964]: pbx.c:4101 pbx_extension_helper: Launching 'Wait'
    -- Executing [1060@from-internal-xfer:7] Wait("SIP/1011-00000000", "1") in new stack
[Sep 29 15:13:45] DEBUG[15964]: pbx.c:4101 pbx_extension_helper: Launching 'Congestion'
    -- Executing [1060@from-internal-xfer:8] Congestion("SIP/1011-00000000", "20") in new stack
[Sep 29 15:13:45] DEBUG[15964]: channel.c:4467 ast_indicate_data: Driver for channel 'SIP/1011-00000000' does not support indication 8, emulating it
[Sep 29 15:13:45] DEBUG[15964]: channel.c:5107 set_format: Set channel SIP/1011-00000000 to write format slin
[Sep 29 15:13:45] DEBUG[15964]: channel.c:3480 ast_settimeout: Scheduling timer at (50 requested / 50 actual) timer ticks per second
[Sep 29 15:13:45] DEBUG[15964]: res_rtp_asterisk.c:1085 ast_rtp_raw_write: Difference is 8384, ms is 1068
[Sep 29 15:13:46] DEBUG[15964]: res_rtp_asterisk.c:1697 ast_rtcp_read: Got RTCP report of 36 bytes
[Sep 29 15:13:49] DEBUG[15930]: chan_sip.c:7513 sip_alloc: Allocating new SIP dialog for 3fba79be4f9f6ec813407958499a3b8d@x.x.x..134:0 - OPTIONS (No RTP)
[Sep 29 15:13:49] DEBUG[15930]: acl.c:725 ast_ouraddrfor: For destination 'x.x.x.99', our source address is 'x.x.x..134'.
[Sep 29 15:13:49] DEBUG[15930]: chan_sip.c:3477 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_UDP with address x.x.x..134:5060
[Sep 29 15:13:49] DEBUG[15930]: chan_sip.c:3052 initialize_initreq: Initializing initreq for method OPTIONS - callid 22bd907a69d801bd6049c24b254680e0@x.x.x.134:5060
[Sep 29 15:13:49] DEBUG[15930]: chan_sip.c:3323 __sip_xmit: Trying to put 'OPTIONS sip' onto UDP socket destined for x.x.x.99:3826
[Sep 29 15:13:49] DEBUG[15930]: chan_sip.c:7513 sip_alloc: Allocating new SIP dialog for 7b7c0403140f3e7847cfeb7e632ac858@x.x.x.134:0 - OPTIONS (No RTP)
[Sep 29 15:13:49] DEBUG[15930]: acl.c:725 ast_ouraddrfor: For destination 'x.x.x.99', our source address is 'x.x.x.134'.
[Sep 29 15:13:49] DEBUG[15930]: chan_sip.c:3477 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_UDP with address x.x.x.134:5060
[Sep 29 15:13:49] DEBUG[15930]: chan_sip.c:3052 initialize_initreq: Initializing initreq for method OPTIONS - callid 52cec7b272e62259536d6c281e690b30@x.x.x.134:5060
[Sep 29 15:13:49] DEBUG[15930]: chan_sip.c:3323 __sip_xmit: Trying to put 'OPTIONS sip' onto UDP socket destined for x.x.x.99:62863
[Sep 29 15:13:49] DEBUG[15930]: chan_sip.c:7513 sip_alloc: Allocating new SIP dialog for 0a46104b732e44f66baafe32215d4038@x.x.x.134:0 - OPTIONS (No RTP)
[Sep 29 15:13:49] DEBUG[15930]: acl.c:725 ast_ouraddrfor: For destination 'x.x.x.99', our source address is 'x.x.x.134'.
[Sep 29 15:13:49] DEBUG[15930]: chan_sip.c:3477 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_UDP with address x.x.x.134:5060
[Sep 29 15:13:49] DEBUG[15930]: chan_sip.c:3052 initialize_initreq: Initializing initreq for method OPTIONS - callid 68610d74654ccb58590fced8771aca70@x.x.x.134:5060
[Sep 29 15:13:49] DEBUG[15930]: chan_sip.c:3323 __sip_xmit: Trying to put 'OPTIONS sip' onto UDP socket destined for x.x.x.99:43938
[Sep 29 15:13:49] DEBUG[15930]: chan_sip.c:7513 sip_alloc: Allocating new SIP dialog for 5330b80f19e0276f02fd19cb50280b54@x.x.x.134:0 - OPTIONS (No RTP)
[Sep 29 15:13:49] DEBUG[15930]: acl.c:725 ast_ouraddrfor: For destination 'x.x.x.99', our source address is 'x.x.x.134'.
[Sep 29 15:13:49] DEBUG[15930]: chan_sip.c:3477 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_UDP with address x.x.x.134:5060
[Sep 29 15:13:49] DEBUG[15930]: chan_sip.c:3052 initialize_initreq: Initializing initreq for method OPTIONS - callid 139f9f590a9ecbca16b4fe115c3cb347@x.x.x.134:5060
[Sep 29 15:13:49] DEBUG[15930]: chan_sip.c:3323 __sip_xmit: Trying to put 'OPTIONS sip' onto UDP socket destined for x.x.x.99:54562
[Sep 29 15:13:49] DEBUG[15930]: chan_sip.c:7833 find_call: = Looking for  Call ID: 22bd907a69d801bd6049c24b254680e0@x.x.x.134:5060 (Checking To) --From tag as7e1f6c53 --To-tag d19578d2  
[Sep 29 15:13:49] DEBUG[15930]: chan_sip.c:4009 __sip_ack: Stopping retransmission on '22bd907a69d801bd6049c24b254680e0@x.x.x.134:5060' of Request 102: Match Found
[Sep 29 15:13:49] DEBUG[15930]: chan_sip.c:5896 sip_destroy: Destroying SIP dialog 22bd907a69d801bd6049c24b254680e0@x.x.x.134:5060
[Sep 29 15:13:49] DEBUG[15930]: chan_sip.c:7833 find_call: = Looking for  Call ID: 52cec7b272e62259536d6c281e690b30@x.x.x.134:5060 (Checking To) --From tag as3362ce03 --To-tag fd2f4aa413fbcd50i1  
[Sep 29 15:13:49] DEBUG[15930]: chan_sip.c:4009 __sip_ack: Stopping retransmission on '52cec7b272e62259536d6c281e690b30@x.x.x.134:5060' of Request 102: Match Found
[Sep 29 15:13:49] DEBUG[15930]: chan_sip.c:5896 sip_destroy: Destroying SIP dialog 52cec7b272e62259536d6c281e690b30@x.x.x.134:5060
[Sep 29 15:13:49] DEBUG[15930]: chan_sip.c:7833 find_call: = Looking for  Call ID: 68610d74654ccb58590fced8771aca70@x.x.x.134:5060 (Checking To) --From tag as52475d5f --To-tag b62272ba1821d58di2  
[Sep 29 15:13:49] DEBUG[15930]: chan_sip.c:4009 __sip_ack: Stopping retransmission on '68610d74654ccb58590fced8771aca70@x.x.x.134:5060' of Request 102: Match Found
[Sep 29 15:13:49] DEBUG[15930]: chan_sip.c:5896 sip_destroy: Destroying SIP dialog 68610d74654ccb58590fced8771aca70@x.x.x.134:5060
[Sep 29 15:13:49] DEBUG[15930]: chan_sip.c:7833 find_call: = Looking for  Call ID: 139f9f590a9ecbca16b4fe115c3cb347@x.x.x.134:5060 (Checking To) --From tag as34f7b62c --To-tag 3921e5925ba13f2ai2  
[Sep 29 15:13:49] DEBUG[15930]: chan_sip.c:4009 __sip_ack: Stopping retransmission on '139f9f590a9ecbca16b4fe115c3cb347@x.x.x.134:5060' of Request 102: Match Found
[Sep 29 15:13:49] DEBUG[15930]: chan_sip.c:5896 sip_destroy: Destroying SIP dialog 139f9f590a9ecbca16b4fe115c3cb347@x.x.x.134:5060
[Sep 29 15:13:49] DEBUG[15932]: chan_iax2.c:2396 peercnt_remove: ip callno count decremented to 1 for x.x.x.98
[Sep 29 15:13:49] DEBUG[15935]: chan_iax2.c:2366 peercnt_add: ip callno count incremented to 2 for x.x.x.98
[Sep 29 15:13:49] DEBUG[15936]: chan_iax2.c:2714 sched_delay_remove: schedule decrement of callno used for x.x.x.98 in 60 seconds
[Sep 29 15:13:49] DEBUG[15936]: chan_iax2.c:10854 socket_process: Peer sip.miko.ru: got pong, lastms 1, historicms 1, maxms 2000
[Sep 29 15:13:52] DEBUG[15930]: chan_sip.c:7513 sip_alloc: Allocating new SIP dialog for 5e8a94681b58062c4b735c7c43e61162@x.x.x.134:0 - OPTIONS (No RTP)
[Sep 29 15:13:52] DEBUG[15930]: acl.c:725 ast_ouraddrfor: For destination '172.16.32.206', our source address is 'x.x.x.134'.
[Sep 29 15:13:52] DEBUG[15930]: chan_sip.c:3477 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_UDP with address x.x.x.134:5060
[Sep 29 15:13:52] DEBUG[15930]: chan_sip.c:3052 initialize_initreq: Initializing initreq for method OPTIONS - callid 29ca70de66cb10b7241398585487c460@x.x.x.134:5060
[Sep 29 15:13:52] DEBUG[15930]: chan_sip.c:3323 __sip_xmit: Trying to put 'OPTIONS sip' onto UDP socket destined for 172.16.32.206:5060
[Sep 29 15:13:53] DEBUG[15930]: chan_sip.c:3323 __sip_xmit: Trying to put 'OPTIONS sip' onto UDP socket destined for 172.16.32.206:5060
[Sep 29 15:13:53] DEBUG[15964]: res_rtp_asterisk.c:1697 ast_rtcp_read: Got RTCP report of 36 bytes
[Sep 29 15:13:54] DEBUG[15930]: chan_sip.c:3323 __sip_xmit: Trying to put 'OPTIONS sip' onto UDP socket destined for 172.16.32.206:5060

nelgondar · Сообщение **nelgondar** » 30 сен 2011, 15:25

Всем спасибо. выяснил что xfer контекст на знал никаких экстейшенов

ded · Сообщение **ded** » 30 сен 2011, 15:45

[from-internal-xfer]??
Как такое может быть с продуктом из коробки?

Кстати:

Хорошо полуркавший нуб перестает быть нубом и становится полноценным членом сообщества, в котором он луркал. Поскольку нубов не очень любят, то по идее лурканье надо всевозможно поощрять. Однако обычно пожелание высказывается в грубой форме: «иди луркай»! Скорее всего, это связано с тем, что нуб, старающийся перестать быть нубом, луркает сам, без всяческих дополнительных призывов к лурканью.

*+freepbx не работает перевод звонков.

Re: *+freepbx не работает перевод звонков.

Re: *+freepbx не работает перевод звонков.

Re: *+freepbx не работает перевод звонков.

Re: *+freepbx не работает перевод звонков.

Re: *+freepbx не работает перевод звонков.

Re: *+freepbx не работает перевод звонков.

Re: *+freepbx не работает перевод звонков.

Re: *+freepbx не работает перевод звонков.