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Проблема при одновременных вызовах

Проблемы Asterisk без вэб-оболочек и их решения

Модераторы: april22, Zavr2008

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RamonSantiago
Сообщения: 14
Зарегистрирован: 14 май 2014, 15:29

Проблема при одновременных вызовах

Сообщение RamonSantiago »

Суть проблемы:
Один звонок проходит нормально (и входящий и исходящий), при поступлении второго входящего, пропадает звук в обоих каналах.
При втором исходящем вызове в первом пропадает звук, второй канал при этом живет.
Два внутренних звонка проходят без проблем.
Внешине каналы сипнет и ростелеком по sip

Проблема возникла без видимых причин, т.е. никаких изменений в конфигурации оборудования, ос или самого астериска не было.

Что делал
перенес конфигурацию астериск с виртуальной машины на реальное железо.

Подскажите куда копать?

Заранее благодарен.
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april22
Сообщения: 2187
Зарегистрирован: 09 июл 2012, 09:47

Re: Проблема при одновременных вызовах

Сообщение april22 »

в логи
дампы
Своими вопросами , вы загоняете меня в ГУГЛЬ.
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RamonSantiago
Сообщения: 14
Зарегистрирован: 14 май 2014, 15:29

Re: Проблема при одновременных вызовах

Сообщение RamonSantiago »

Логи вроде без криминала...
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

Код: Выделить всё

[Apr  5 17:34:40] WARNING[948] chan_sip.c: Retransmission timeout reached on transmission 002155d6-e30f0040-44f9437e-67f0e0a6@192.168.175.101 for seqno 102 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6400ms with no response

[Apr  7 14:01:27] WARNING[948] chan_sip.c: Retransmission timeout reached on transmission 002155d6-e30f0003-8443912e-58fef481@192.168.175.101 for seqno 102 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6400ms with no response
Остальные сообщения связаны с появлением и отключением пиров.
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RamonSantiago
Сообщения: 14
Зарегистрирован: 14 май 2014, 15:29

Re: Проблема при одновременных вызовах

Сообщение RamonSantiago »

verbose 9
debug 9
простыня получилась здоровая...
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

Код: Выделить всё

  == Using SIP RTP CoS mark 5
[Apr  7 15:13:20] DEBUG[948][C-00000108]: chan_sip.c:5736 do_setnat: Setting NAT on RTP to Off
[Apr  7 15:13:20] DEBUG[948][C-00000108]: chan_sip.c:10049 process_sdp: Processing session-level SDP v=0... UNSUPPORTED OR FAILED.
[Apr  7 15:13:20] DEBUG[948][C-00000108]: chan_sip.c:10049 process_sdp: Processing session-level SDP o=Cisco-SIPUA 27353 0 IN IP4 192.168.175.132... OK.
[Apr  7 15:13:20] DEBUG[948][C-00000108]: chan_sip.c:10049 process_sdp: Processing session-level SDP s=SIP Call... UNSUPPORTED OR FAILED.
[Apr  7 15:13:20] DEBUG[948][C-00000108]: chan_sip.c:10049 process_sdp: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED.
[Apr  7 15:13:20] DEBUG[948][C-00000108]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 8 based on m type on 0xb409f1c0
[Apr  7 15:13:20] DEBUG[948][C-00000108]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 0 based on m type on 0xb409f1c0
[Apr  7 15:13:20] DEBUG[948][C-00000108]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 18 based on m type on 0xb409f1c0
[Apr  7 15:13:20] DEBUG[948][C-00000108]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 101 based on m type on 0xb409f1c0
[Apr  7 15:13:20] DEBUG[948][C-00000108]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.175.132' into...
[Apr  7 15:13:20] DEBUG[948][C-00000108]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.175.132' and port ''.
[Apr  7 15:13:20] DEBUG[948][C-00000108]: chan_sip.c:10482 process_sdp: Processing media-level (audio) SDP c=IN IP4 192.168.175.132... OK.
[Apr  7 15:13:20] DEBUG[948][C-00000108]: chan_sip.c:10482 process_sdp: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK.
[Apr  7 15:13:20] DEBUG[948][C-00000108]: chan_sip.c:10482 process_sdp: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK.
[Apr  7 15:13:20] DEBUG[948][C-00000108]: chan_sip.c:10482 process_sdp: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK.
[Apr  7 15:13:20] DEBUG[948][C-00000108]: chan_sip.c:10482 process_sdp: Processing media-level (audio) SDP a=fmtp:18 annexb=no... OK.
[Apr  7 15:13:20] DEBUG[948][C-00000108]: chan_sip.c:10482 process_sdp: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK.
[Apr  7 15:13:20] DEBUG[948][C-00000108]: chan_sip.c:10482 process_sdp: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED OR FAILED.
[Apr  7 15:13:20] DEBUG[948][C-00000108]: chan_sip.c:10482 process_sdp: Processing media-level (audio) SDP a=sendrecv... OK.
[Apr  7 15:13:20] DEBUG[948][C-00000108]: res_rtp_asterisk.c:3994 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0xb6b21084'
[Apr  7 15:13:20] DEBUG[948][C-00000108]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 0 from 0xb409f1c0 to 0xb6b21230
[Apr  7 15:13:20] DEBUG[948][C-00000108]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 8 from 0xb409f1c0 to 0xb6b21230
[Apr  7 15:13:20] DEBUG[948][C-00000108]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 18 from 0xb409f1c0 to 0xb6b21230
[Apr  7 15:13:20] DEBUG[948][C-00000108]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 101 from 0xb409f1c0 to 0xb6b21230
[Apr  7 15:13:20] DEBUG[948][C-00000108]: res_rtp_asterisk.c:3913 ast_rtp_prop_set: Ignoring duplicate RTCP property on RTP instance '0xb6b21084'
[Apr  7 15:13:20] DEBUG[948][C-00000108]: chan_sip.c:10753 process_sdp: We're settling with these formats: (alaw)
[Apr  7 15:13:20] DEBUG[948][C-00000108]: chan_sip.c:25403 handle_request_invite: Checking SIP call limits for device 104
[Apr  7 15:13:20] DEBUG[948][C-00000108]: chan_sip.c:6680 update_call_counter: Updating call counter for incoming call
[Apr  7 15:13:20] DEBUG[948][C-00000108]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.175.15' into...
[Apr  7 15:13:20] DEBUG[948][C-00000108]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.175.15' and port ''.
[Apr  7 15:13:20] DEBUG[948][C-00000108]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.175.15' into...
[Apr  7 15:13:20] DEBUG[948][C-00000108]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.175.15' and port ''.
[Apr  7 15:13:20] DEBUG[948][C-00000108]: chan_sip.c:7951 sip_new: *** Our native formats are (alaw)
[Apr  7 15:13:20] DEBUG[948][C-00000108]: chan_sip.c:7952 sip_new: *** Joint capabilities are (alaw)
[Apr  7 15:13:20] DEBUG[948][C-00000108]: chan_sip.c:7953 sip_new: *** Our capabilities are (alaw)
[Apr  7 15:13:20] DEBUG[948][C-00000108]: chan_sip.c:7954 sip_new: *** AST_CODEC_CHOOSE formats are alaw
[Apr  7 15:13:20] DEBUG[948][C-00000108]: chan_sip.c:7982 sip_new: This channel will not be able to handle video.
[Apr  7 15:13:20] DEBUG[948][C-00000108]: chan_sip.c:16241 build_route: build_route: Contact hop: <sip:%5bEmployee%20phone%20number%5d@192.168.175.132:5060;transport=udp>
[Apr  7 15:13:20] DEBUG[948][C-00000108]: chan_sip.c:25715 handle_request_invite: SIP/104-00000320: New call is still down.... Trying...
[Apr  7 15:13:20] DEBUG[930]: devicestate.c:344 _ast_device_state: No provider found, checking channel drivers for SIP - 104
[Apr  7 15:13:20] DEBUG[948][C-00000108]: chan_sip.c:3875 __sip_xmit: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.175.132:5060
[Apr  7 15:13:20] DEBUG[930]: chan_sip.c:29602 sip_devicestate: Checking device state for peer 104
[Apr  7 15:13:20] DEBUG[930]: devicestate.c:467 do_state_change: Changing state for SIP/104 - state 3 (Busy)
[Apr  7 15:13:20] DEBUG[930]: devicestate.c:442 devstate_event: device 'SIP/104' state '3'
[Apr  7 15:13:20] DEBUG[930]: devicestate.c:344 _ast_device_state: No provider found, checking channel drivers for SIP - 104
[Apr  7 15:13:20] DEBUG[930]: chan_sip.c:29602 sip_devicestate: Checking device state for peer 104
[Apr  7 15:13:20] DEBUG[930]: devicestate.c:467 do_state_change: Changing state for SIP/104 - state 3 (Busy)
[Apr  7 15:13:20] DEBUG[930]: devicestate.c:442 devstate_event: device 'SIP/104' state '3'
[Apr  7 15:13:20] DEBUG[688][C-00000108]: pbx.c:4890 pbx_extension_helper: Launching 'NoOp'
    -- Executing [89109998877@LOCAL-PHONES:1] NoOp("SIP/104-00000320", "") in new stack
[Apr  7 15:13:20] DEBUG[688][C-00000108]: pbx.c:3675 ast_str_retrieve_variable: Result of 'EXTEN' is '89109998877'
[Apr  7 15:13:20] DEBUG[688][C-00000108]: pbx.c:4890 pbx_extension_helper: Launching 'Verbose'
    -- Executing [89109998877@LOCAL-PHONES:2] Verbose("SIP/104-00000320", "########## Исходящий на мобильный номер- 89109998877  ##########") in new stack
########## Исходящий на мобильный номер- 89109998877  ##########
[Apr  7 15:13:20] DEBUG[688][C-00000108]: pbx.c:3675 ast_str_retrieve_variable: Result of 'EXTEN' is '89109998877'
[Apr  7 15:13:20] DEBUG[688][C-00000108]: pbx.c:4890 pbx_extension_helper: Launching 'Dial'
    -- Executing [89109998877@LOCAL-PHONES:3] Dial("SIP/104-00000320", "SIP/CTK/89109998877,120") in new stack
[Apr  7 15:13:20] DEBUG[688][C-00000108]: chan_sip.c:29707 sip_request_call: Asked to create a SIP channel with formats: (alaw)
[Apr  7 15:13:20] DEBUG[688][C-00000108]: chan_sip.c:8770 sip_alloc: Allocating new SIP dialog for 2a7e261408a2184d52d45b0f13003924@127.0.1.1:5060 - INVITE (No RTP)
[Apr  7 15:13:20] DEBUG[688][C-00000108]: rtp_engine.c:283 ast_rtp_instance_new: Using engine 'asterisk' for RTP instance '0x80fcc5f4'
[Apr  7 15:13:20] DEBUG[688][C-00000108]: res_rtp_asterisk.c:1808 ast_rtp_new: Allocated port 23424 for RTP instance '0x80fcc5f4'
[Apr  7 15:13:20] DEBUG[688][C-00000108]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.175.15' into...
[Apr  7 15:13:20] DEBUG[688][C-00000108]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.175.15' and port ''.
[Apr  7 15:13:20] DEBUG[688][C-00000108]: rtp_engine.c:292 ast_rtp_instance_new: RTP instance '0x80fcc5f4' is setup and ready to go
[Apr  7 15:13:20] DEBUG[688][C-00000108]: res_rtp_asterisk.c:3947 ast_rtp_prop_set: Setup RTCP on RTP instance '0x80fcc5f4'
  == Using SIP RTP CoS mark 5
[Apr  7 15:13:20] DEBUG[688][C-00000108]: chan_sip.c:5736 do_setnat: Setting NAT on RTP to Off
[Apr  7 15:13:20] DEBUG[688][C-00000108]: chan_sip.c:8565 change_callid_pvt: SIP call-id changed from '2a7e261408a2184d52d45b0f13003924@127.0.1.1:5060' to '2a7e261408a2184d52d45b0f13003924@0000719.yr.centertelecom.ru'
[Apr  7 15:13:20] DEBUG[688][C-00000108]: acl.c:979 ast_ouraddrfor: For destination '8.8.8.8', our source address is '192.168.175.15'.
[Apr  7 15:13:20] DEBUG[688][C-00000108]: chan_sip.c:4032 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_UDP with address 192.168.175.15:5060
[Apr  7 15:13:20] DEBUG[688][C-00000108]: chan_sip.c:5736 do_setnat: Setting NAT on RTP to Off
[Apr  7 15:13:20] DEBUG[688][C-00000108]: chan_sip.c:8565 change_callid_pvt: SIP call-id changed from '2a7e261408a2184d52d45b0f13003924@0000719.yr.centertelecom.ru' to '7ef59bb00593969443f32b14739fe827@0000719.yr.centertelecom.ru'
[Apr  7 15:13:20] DEBUG[688][C-00000108]: chan_sip.c:7951 sip_new: *** Our native formats are (alaw)
[Apr  7 15:13:20] DEBUG[688][C-00000108]: chan_sip.c:7952 sip_new: *** Joint capabilities are (alaw)
[Apr  7 15:13:20] DEBUG[688][C-00000108]: chan_sip.c:7953 sip_new: *** Our capabilities are (alaw)
[Apr  7 15:13:20] DEBUG[688][C-00000108]: chan_sip.c:7954 sip_new: *** AST_CODEC_CHOOSE formats are alaw
[Apr  7 15:13:20] DEBUG[688][C-00000108]: chan_sip.c:7956 sip_new: *** Our preferred formats from the incoming channel are (alaw)
[Apr  7 15:13:20] DEBUG[688][C-00000108]: chan_sip.c:7982 sip_new: This channel will not be able to handle video.
[Apr  7 15:13:20] DEBUG[688][C-00000108]: channel_internal_api.c:882 ast_channel_callid_set: Channel Call ID changing from [C-00000108] to [C-00000108]
[Apr  7 15:13:20] DEBUG[688][C-00000108]: channel.c:6507 ast_channel_inherit_variables: Not copying variable DIALEDTIME.
[Apr  7 15:13:20] DEBUG[688][C-00000108]: channel.c:6507 ast_channel_inherit_variables: Not copying variable ANSWEREDTIME.
[Apr  7 15:13:20] DEBUG[688][C-00000108]: channel.c:6507 ast_channel_inherit_variables: Not copying variable DIALEDPEERNAME.
[Apr  7 15:13:20] DEBUG[688][C-00000108]: channel.c:6507 ast_channel_inherit_variables: Not copying variable DIALEDPEERNUMBER.
[Apr  7 15:13:20] DEBUG[688][C-00000108]: channel.c:6507 ast_channel_inherit_variables: Not copying variable DIALSTATUS.
[Apr  7 15:13:20] DEBUG[688][C-00000108]: channel.c:6507 ast_channel_inherit_variables: Not copying variable SIPCALLID.
[Apr  7 15:13:20] DEBUG[688][C-00000108]: channel.c:6507 ast_channel_inherit_variables: Not copying variable SIPDOMAIN.
[Apr  7 15:13:20] DEBUG[688][C-00000108]: channel.c:6507 ast_channel_inherit_variables: Not copying variable SIPURI.
[Apr  7 15:13:20] DEBUG[688][C-00000108]: chan_sip.c:6356 sip_call: Outgoing Call for 89109998877
[Apr  7 15:13:20] DEBUG[688][C-00000108]: chan_sip.c:6680 update_call_counter: Updating call counter for outgoing call
[Apr  7 15:13:20] DEBUG[688][C-00000108]: chan_sip.c:13151 add_sdp: ** Our capability: (alaw) Video flag: False Text flag: False
[Apr  7 15:13:20] DEBUG[688][C-00000108]: chan_sip.c:13152 add_sdp: ** Our prefcodec: (alaw)
[Apr  7 15:13:20] DEBUG[688][C-00000108]: chan_sip.c:13288 add_sdp: -- Done with adding codecs to SDP
[Apr  7 15:13:20] DEBUG[688][C-00000108]: chan_sip.c:13482 add_sdp: Done building SDP. Settling with this capability: (alaw)
[Apr  7 15:13:20] DEBUG[688][C-00000108]: chan_sip.c:3518 initialize_initreq: Initializing initreq for method INVITE - callid 7ef59bb00593969443f32b14739fe827@0000719.yr.centertelecom.ru
[Apr  7 15:13:20] DEBUG[688][C-00000108]: chan_sip.c:3875 __sip_xmit: Trying to put 'INVITE sip:' onto UDP socket destined for 8.8.8.8:5060
    -- Called SIP/CTK/89109998877
[Apr  7 15:13:20] DEBUG[948]: chan_sip.c:9167 find_call: = Looking for  Call ID: 7ef59bb00593969443f32b14739fe827@0000719.yr.centertelecom.ru (Checking To) --From tag as2b1ea6b4 --To-tag
[Apr  7 15:13:20] DEBUG[948][C-00000108]: chan_sip.c:4608 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '7ef59bb00593969443f32b14739fe827@0000719.yr.centertelecom.ru' Request 102: Found
[Apr  7 15:13:20] DEBUG[948][C-00000108]: chan_sip.c:22601 handle_response_invite: SIP response 100 to standard invite
[Apr  7 15:13:20] DEBUG[948]: chan_sip.c:9167 find_call: = Looking for  Call ID: 7ef59bb00593969443f32b14739fe827@0000719.yr.centertelecom.ru (Checking To) --From tag as2b1ea6b4 --To-tag q2UNTxsT
[Apr  7 15:13:20] DEBUG[948][C-00000108]: chan_sip.c:4529 __sip_ack: Acked pending invite 102
[Apr  7 15:13:20] DEBUG[948][C-00000108]: chan_sip.c:4567 __sip_ack: Stopping retransmission on '7ef59bb00593969443f32b14739fe827@0000719.yr.centertelecom.ru' of Request 102: Match Found
[Apr  7 15:13:20] DEBUG[948][C-00000108]: chan_sip.c:22601 handle_response_invite: SIP response 401 to standard invite
[Apr  7 15:13:20] DEBUG[948][C-00000108]: chan_sip.c:3875 __sip_xmit: Trying to put 'ACK sip:891' onto UDP socket destined for 8.8.8.8:5060
[Apr  7 15:13:20] DEBUG[948][C-00000108]: chan_sip.c:21723 do_proxy_auth: Auth attempt 1 on INVITE
[Apr  7 15:13:20] DEBUG[948][C-00000108]: chan_sip.c:13151 add_sdp: ** Our capability: (alaw) Video flag: False Text flag: False
[Apr  7 15:13:20] DEBUG[948][C-00000108]: chan_sip.c:13152 add_sdp: ** Our prefcodec: (alaw)
[Apr  7 15:13:20] DEBUG[948][C-00000108]: chan_sip.c:13288 add_sdp: -- Done with adding codecs to SDP
[Apr  7 15:13:20] DEBUG[948][C-00000108]: chan_sip.c:13482 add_sdp: Done building SDP. Settling with this capability: (alaw)
[Apr  7 15:13:20] DEBUG[948][C-00000108]: chan_sip.c:3875 __sip_xmit: Trying to put 'INVITE sip:' onto UDP socket destined for 8.8.8.8:5060
[Apr  7 15:13:20] DEBUG[948]: chan_sip.c:9167 find_call: = Looking for  Call ID: 7ef59bb00593969443f32b14739fe827@0000719.yr.centertelecom.ru (Checking To) --From tag as2b1ea6b4 --To-tag
[Apr  7 15:13:20] DEBUG[948][C-00000108]: chan_sip.c:4608 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '7ef59bb00593969443f32b14739fe827@0000719.yr.centertelecom.ru' Request 103: Found
[Apr  7 15:13:20] DEBUG[948][C-00000108]: chan_sip.c:22601 handle_response_invite: SIP response 100 to standard invite
[Apr  7 15:13:26] DEBUG[948]: chan_sip.c:9167 find_call: = Looking for  Call ID: 7ef59bb00593969443f32b14739fe827@0000719.yr.centertelecom.ru (Checking To) --From tag as2b1ea6b4 --To-tag EGRfvx0j
[Apr  7 15:13:26] DEBUG[948][C-00000108]: chan_sip.c:4608 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '7ef59bb00593969443f32b14739fe827@0000719.yr.centertelecom.ru' Request 103: Found
[Apr  7 15:13:26] DEBUG[948][C-00000108]: chan_sip.c:22601 handle_response_invite: SIP response 180 to standard invite
[Apr  7 15:13:26] DEBUG[948][C-00000108]: chan_sip.c:16241 build_route: build_route: Contact hop: <sip:89109998877@8.8.8.8:5060;transport=udp>
    -- SIP/CTK-00000321 is ringing
[Apr  7 15:13:26] DEBUG[688][C-00000108]: rtp_engine.c:1805 ast_rtp_instance_early_bridge: Setting early bridge SDP of 'SIP/104-00000320' with that of 'SIP/CTK-00000321'
[Apr  7 15:13:26] DEBUG[688][C-00000108]: chan_sip.c:3875 __sip_xmit: Trying to put 'SIP/2.0 180' onto UDP socket destined for 192.168.175.132:5060
[Apr  7 15:13:26] DEBUG[930]: devicestate.c:344 _ast_device_state: No provider found, checking channel drivers for SIP - CTK
[Apr  7 15:13:26] DEBUG[930]: chan_sip.c:29602 sip_devicestate: Checking device state for peer CTK
[Apr  7 15:13:26] DEBUG[930]: devicestate.c:467 do_state_change: Changing state for SIP/CTK - state 1 (Not in use)
[Apr  7 15:13:26] DEBUG[930]: devicestate.c:442 devstate_event: device 'SIP/CTK' state '1'
[Apr  7 15:13:26] DEBUG[948]: chan_sip.c:9167 find_call: = Looking for  Call ID: 7ef59bb00593969443f32b14739fe827@0000719.yr.centertelecom.ru (Checking To) --From tag as2b1ea6b4 --To-tag EGRfvx0j
[Apr  7 15:13:26] DEBUG[948][C-00000108]: chan_sip.c:4608 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '7ef59bb00593969443f32b14739fe827@0000719.yr.centertelecom.ru' Request 103: Found
[Apr  7 15:13:26] DEBUG[948][C-00000108]: chan_sip.c:22601 handle_response_invite: SIP response 180 to standard invite
[Apr  7 15:13:26] DEBUG[948][C-00000108]: chan_sip.c:16241 build_route: build_route: Contact hop: <sip:89109998877@8.8.8.8:5060;transport=udp>
    -- SIP/CTK-00000321 is ringing
[Apr  7 15:13:26] DEBUG[688][C-00000108]: rtp_engine.c:1805 ast_rtp_instance_early_bridge: Setting early bridge SDP of 'SIP/104-00000320' with that of 'SIP/CTK-00000321'
[Apr  7 15:13:31] DEBUG[948]: chan_sip.c:3477 registry_addref: SIP Registry 0000719.yr.centertelecom.ru: refcount now 3
[Apr  7 15:13:31] DEBUG[948]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '0000719.yr.centertelecom.ru' into...
[Apr  7 15:13:31] DEBUG[948]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '0000719.yr.centertelecom.ru' and port ''.
[Apr  7 15:13:31] DEBUG[948]: dnsmgr.c:164 internal_dnsmgr_lookup: doing dnsmgr_lookup for '0000719.yr.centertelecom.ru'
[Apr  7 15:13:31] DEBUG[948]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '0000719.yr.centertelecom.ru' into...
[Apr  7 15:13:31] DEBUG[948]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '0000719.yr.centertelecom.ru' and port ''.
[Apr  7 15:13:31] DEBUG[948]: chan_sip.c:3469 registry_unref: SIP Registry 0000719.yr.centertelecom.ru: refcount now 2
[Apr  7 15:13:31] DEBUG[948]: chan_sip.c:8770 sip_alloc: Allocating new SIP dialog for 5fb9bf156b322e4016a1712864daf34d@127.0.1.1 - REGISTER (No RTP)
[Apr  7 15:13:31] DEBUG[948]: chan_sip.c:3477 registry_addref: SIP Registry 0000719.yr.centertelecom.ru: refcount now 3
[Apr  7 15:13:31] DEBUG[948]: acl.c:979 ast_ouraddrfor: For destination '8.8.8.8', our source address is '192.168.175.15'.
[Apr  7 15:13:31] DEBUG[948]: chan_sip.c:4032 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_UDP with address 192.168.175.15:5060
[Apr  7 15:13:31] DEBUG[948]: chan_sip.c:3477 registry_addref: SIP Registry 0000719.yr.centertelecom.ru: refcount now 4
[Apr  7 15:13:31] DEBUG[948]: chan_sip.c:15296 transmit_register: Scheduled a registration timeout for 0000719.yr.centertelecom.ru id  #360869
[Apr  7 15:13:31] DEBUG[948]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '0000719.yr.centertelecom.ru' into...
[Apr  7 15:13:31] DEBUG[948]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '0000719.yr.centertelecom.ru' and port ''.
[Apr  7 15:13:31] DEBUG[948]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '0000719.yr.centertelecom.ru' into...
[Apr  7 15:13:31] DEBUG[948]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '0000719.yr.centertelecom.ru' and port ''.
[Apr  7 15:13:31] DEBUG[948]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '0000719.yr.centertelecom.ru' into...
[Apr  7 15:13:31] DEBUG[948]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '0000719.yr.centertelecom.ru' and port ''.
[Apr  7 15:13:31] DEBUG[948]: chan_sip.c:3518 initialize_initreq: Initializing initreq for method REGISTER - callid 5fb9bf156b322e4016a1712864daf34d@127.0.1.1
[Apr  7 15:13:31] DEBUG[948]: chan_sip.c:15371 transmit_register: REGISTER attempt 1 to admin@0000719.yr.centertelecom.ru
[Apr  7 15:13:31] DEBUG[948]: chan_sip.c:3875 __sip_xmit: Trying to put 'REGISTER si' onto UDP socket destined for 8.8.8.8:5060
[Apr  7 15:13:31] DEBUG[948]: chan_sip.c:3469 registry_unref: SIP Registry 0000719.yr.centertelecom.ru: refcount now 3
[Apr  7 15:13:31] DEBUG[948]: chan_sip.c:9167 find_call: = Looking for  Call ID: 5fb9bf156b322e4016a1712864daf34d@127.0.1.1 (Checking To) --From tag as4d201077 --To-tag oz8qjvse
[Apr  7 15:13:31] DEBUG[948]: chan_sip.c:4567 __sip_ack: Stopping retransmission on '5fb9bf156b322e4016a1712864daf34d@127.0.1.1' of Request 140: Match Found
[Apr  7 15:13:31] DEBUG[948]: chan_sip.c:23419 handle_response_register: Registration successful
[Apr  7 15:13:31] DEBUG[948]: chan_sip.c:23421 handle_response_register: Cancelling timeout 360869
[Apr  7 15:13:31] DEBUG[948]: chan_sip.c:3469 registry_unref: SIP Registry 0000719.yr.centertelecom.ru: refcount now 2
[Apr  7 15:13:31] DEBUG[948]: chan_sip.c:3469 registry_unref: SIP Registry 0000719.yr.centertelecom.ru: refcount now 1
[Apr  7 15:13:31] DEBUG[948]: chan_sip.c:3477 registry_addref: SIP Registry 0000719.yr.centertelecom.ru: refcount now 2
[Apr  7 15:13:35] DEBUG[948]: chan_sip.c:8770 sip_alloc: Allocating new SIP dialog for 41c6186a2017307f202c3b9e537323d6@127.0.1.1:5060 - OPTIONS (No RTP)
[Apr  7 15:13:35] DEBUG[948]: acl.c:979 ast_ouraddrfor: For destination '192.168.175.154', our source address is '192.168.175.15'.
[Apr  7 15:13:35] DEBUG[948]: chan_sip.c:4032 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_UDP with address 192.168.175.15:5060
[Apr  7 15:13:35] DEBUG[948]: chan_sip.c:8565 change_callid_pvt: SIP call-id changed from '41c6186a2017307f202c3b9e537323d6@127.0.1.1:5060' to '79a1fe312fb15837547b9870164c898f@192.168.175.15:5060'
[Apr  7 15:13:35] DEBUG[948]: chan_sip.c:3518 initialize_initreq: Initializing initreq for method OPTIONS - callid 79a1fe312fb15837547b9870164c898f@192.168.175.15:5060
[Apr  7 15:13:35] DEBUG[948]: chan_sip.c:3875 __sip_xmit: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.175.154:5060
[Apr  7 15:13:35] DEBUG[948]: chan_sip.c:9167 find_call: = Looking for  Call ID: 79a1fe312fb15837547b9870164c898f@192.168.175.15:5060 (Checking To) --From tag as40614588 --To-tag 0018b9101ab832d9b1e9dfdc-6214ba58
[Apr  7 15:13:35] DEBUG[948]: chan_sip.c:4567 __sip_ack: Stopping retransmission on '79a1fe312fb15837547b9870164c898f@192.168.175.15:5060' of Request 102: Match Found
[Apr  7 15:13:35] DEBUG[948]: chan_sip.c:6828 sip_destroy: Destroying SIP dialog 79a1fe312fb15837547b9870164c898f@192.168.175.15:5060
[Apr  7 15:13:39] DEBUG[948]: chan_sip.c:8770 sip_alloc: Allocating new SIP dialog for 635e736c613e1e0c61ac2603457cff3a@127.0.1.1:5060 - OPTIONS (No RTP)
[Apr  7 15:13:39] DEBUG[948]: acl.c:979 ast_ouraddrfor: For destination '4.4.4.4', our source address is '192.168.175.15'.
[Apr  7 15:13:39] DEBUG[948]: chan_sip.c:4032 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_UDP with address 192.168.175.15:5060
[Apr  7 15:13:39] DEBUG[948]: chan_sip.c:8565 change_callid_pvt: SIP call-id changed from '635e736c613e1e0c61ac2603457cff3a@127.0.1.1:5060' to '0406927e4f67cb477b450b5829b863fe@192.168.175.15:5060'
[Apr  7 15:13:39] DEBUG[948]: chan_sip.c:3518 initialize_initreq: Initializing initreq for method OPTIONS - callid 0406927e4f67cb477b450b5829b863fe@192.168.175.15:5060
[Apr  7 15:13:39] DEBUG[948]: chan_sip.c:3875 __sip_xmit: Trying to put 'OPTIONS sip' onto UDP socket destined for 4.4.4.4:5060
[Apr  7 15:13:39] DEBUG[948]: chan_sip.c:9167 find_call: = Looking for  Call ID: 0406927e4f67cb477b450b5829b863fe@192.168.175.15:5060 (Checking To) --From tag as6d8808ed --To-tag 4CD725E4
[Apr  7 15:13:39] DEBUG[948]: chan_sip.c:4567 __sip_ack: Stopping retransmission on '0406927e4f67cb477b450b5829b863fe@192.168.175.15:5060' of Request 102: Match Found
[Apr  7 15:13:39] DEBUG[948]: chan_sip.c:6828 sip_destroy: Destroying SIP dialog 0406927e4f67cb477b450b5829b863fe@192.168.175.15:5060
[Apr  7 15:13:40] DEBUG[948]: chan_sip.c:8770 sip_alloc: Allocating new SIP dialog for 3775924f7af60b4f58ed6cc308d7946a@127.0.1.1:5060 - OPTIONS (No RTP)
[Apr  7 15:13:40] DEBUG[948]: acl.c:979 ast_ouraddrfor: For destination '192.168.175.20', our source address is '192.168.175.15'.
[Apr  7 15:13:40] DEBUG[948]: chan_sip.c:4032 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_UDP with address 192.168.175.15:5060
[Apr  7 15:13:40] DEBUG[948]: chan_sip.c:8565 change_callid_pvt: SIP call-id changed from '3775924f7af60b4f58ed6cc308d7946a@127.0.1.1:5060' to '021209a8381555be2bdb2c101d130cb8@192.168.175.15:5060'
[Apr  7 15:13:40] DEBUG[948]: chan_sip.c:3518 initialize_initreq: Initializing initreq for method OPTIONS - callid 021209a8381555be2bdb2c101d130cb8@192.168.175.15:5060
[Apr  7 15:13:40] DEBUG[948]: chan_sip.c:3875 __sip_xmit: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.175.20:5060
[Apr  7 15:13:40] DEBUG[948]: chan_sip.c:9167 find_call: = Looking for  Call ID: 021209a8381555be2bdb2c101d130cb8@192.168.175.15:5060 (Checking To) --From tag as1b19c9a3 --To-tag 686551ad46dc5d0i0 
[Apr  7 15:13:40] DEBUG[948]: chan_sip.c:4567 __sip_ack: Stopping retransmission on '021209a8381555be2bdb2c101d130cb8@192.168.175.15:5060' of Request 102: Match Found
[Apr  7 15:13:40] DEBUG[948]: chan_sip.c:6828 sip_destroy: Destroying SIP dialog 021209a8381555be2bdb2c101d130cb8@192.168.175.15:5060
[Apr  7 15:13:40] DEBUG[948]: chan_sip.c:8770 sip_alloc: Allocating new SIP dialog for 1b5267321a57a80f4cf49a6b5e4fb7ce@127.0.1.1:5060 - OPTIONS (No RTP)
[Apr  7 15:13:40] DEBUG[948]: acl.c:979 ast_ouraddrfor: For destination '192.168.175.20', our source address is '192.168.175.15'.
[Apr  7 15:13:40] DEBUG[948]: chan_sip.c:4032 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_UDP with address 192.168.175.15:5060
[Apr  7 15:13:40] DEBUG[948]: chan_sip.c:8565 change_callid_pvt: SIP call-id changed from '1b5267321a57a80f4cf49a6b5e4fb7ce@127.0.1.1:5060' to '7d54010d00a44d176e15207a16a141a5@192.168.175.15:5060'
[Apr  7 15:13:40] DEBUG[948]: chan_sip.c:3518 initialize_initreq: Initializing initreq for method OPTIONS - callid 7d54010d00a44d176e15207a16a141a5@192.168.175.15:5060
[Apr  7 15:13:40] DEBUG[948]: chan_sip.c:3875 __sip_xmit: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.175.20:5061
[Apr  7 15:13:40] DEBUG[948]: chan_sip.c:9167 find_call: = Looking for  Call ID: 7d54010d00a44d176e15207a16a141a5@192.168.175.15:5060 (Checking To) --From tag as02674a01 --To-tag 880a6ac57af9bf98i1
[Apr  7 15:13:40] DEBUG[948]: chan_sip.c:4567 __sip_ack: Stopping retransmission on '7d54010d00a44d176e15207a16a141a5@192.168.175.15:5060' of Request 102: Match Found
[Apr  7 15:13:40] DEBUG[948]: chan_sip.c:6828 sip_destroy: Destroying SIP dialog 7d54010d00a44d176e15207a16a141a5@192.168.175.15:5060
[Apr  7 15:13:44] DEBUG[948]: chan_sip.c:9167 find_call: = Looking for  Call ID: 7ef59bb00593969443f32b14739fe827@0000719.yr.centertelecom.ru (Checking To) --From tag as2b1ea6b4 --To-tag EGRfvx0j
[Apr  7 15:13:44] DEBUG[948][C-00000108]: chan_sip.c:4529 __sip_ack: Acked pending invite 103
[Apr  7 15:13:44] DEBUG[948][C-00000108]: chan_sip.c:4567 __sip_ack: Stopping retransmission on '7ef59bb00593969443f32b14739fe827@0000719.yr.centertelecom.ru' of Request 103: Match Found
[Apr  7 15:13:44] DEBUG[948][C-00000108]: chan_sip.c:22601 handle_response_invite: SIP response 200 to standard invite
[Apr  7 15:13:44] DEBUG[948][C-00000108]: chan_sip.c:10049 process_sdp: Processing session-level SDP v=0... UNSUPPORTED OR FAILED.
[Apr  7 15:13:44] DEBUG[948][C-00000108]: chan_sip.c:10049 process_sdp: Processing session-level SDP o=user 182827 182827 IN IP4 8.8.8.8... OK.
[Apr  7 15:13:44] DEBUG[948][C-00000108]: chan_sip.c:10049 process_sdp: Processing session-level SDP s=call... UNSUPPORTED OR FAILED.
[Apr  7 15:13:44] DEBUG[948][C-00000108]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '8.8.8.8' into...
[Apr  7 15:13:44] DEBUG[948][C-00000108]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '8.8.8.8' and port ''.
[Apr  7 15:13:44] DEBUG[948][C-00000108]: chan_sip.c:10049 process_sdp: Processing session-level SDP c=IN IP4 8.8.8.8... OK.
[Apr  7 15:13:44] DEBUG[948][C-00000108]: chan_sip.c:10049 process_sdp: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED.
[Apr  7 15:13:44] DEBUG[948][C-00000108]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 8 based on m type on 0xb409edc0
[Apr  7 15:13:44] DEBUG[948][C-00000108]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 101 based on m type on 0xb409edc0
[Apr  7 15:13:44] DEBUG[948][C-00000108]: chan_sip.c:10482 process_sdp: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK.
[Apr  7 15:13:44] DEBUG[948][C-00000108]: chan_sip.c:10482 process_sdp: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK.
[Apr  7 15:13:44] DEBUG[948][C-00000108]: chan_sip.c:10482 process_sdp: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED OR FAILED.
[Apr  7 15:13:44] DEBUG[948][C-00000108]: chan_sip.c:10482 process_sdp: Processing media-level (audio) SDP a=ptime:20... OK.
[Apr  7 15:13:44] DEBUG[948][C-00000108]: chan_sip.c:10482 process_sdp: Processing media-level (audio) SDP a=sendrecv... OK.
[Apr  7 15:13:44] DEBUG[948][C-00000108]: res_rtp_asterisk.c:3994 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x80fcc5f4'
[Apr  7 15:13:44] DEBUG[948][C-00000108]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 8 from 0xb409edc0 to 0x80fcc7a0
[Apr  7 15:13:44] DEBUG[948][C-00000108]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 101 from 0xb409edc0 to 0x80fcc7a0
[Apr  7 15:13:44] DEBUG[948][C-00000108]: res_rtp_asterisk.c:3913 ast_rtp_prop_set: Ignoring duplicate RTCP property on RTP instance '0x80fcc5f4'
[Apr  7 15:13:44] DEBUG[948][C-00000108]: chan_sip.c:10753 process_sdp: We're settling with these formats: (alaw)
[Apr  7 15:13:44] DEBUG[948][C-00000108]: chan_sip.c:10760 process_sdp: We have an owner, now see if we need to change this call
[Apr  7 15:13:44] DEBUG[948][C-00000108]: chan_sip.c:6680 update_call_counter: Updating call counter for outgoing call
[Apr  7 15:13:44] DEBUG[948][C-00000108]: chan_sip.c:16241 build_route: build_route: Contact hop: <sip:89109998877@8.8.8.8:5060;transport=udp>
[Apr  7 15:13:44] DEBUG[948][C-00000108]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '8.8.8.8:5060' into...
[Apr  7 15:13:44] DEBUG[948][C-00000108]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '8.8.8.8' and port '5060'.
[Apr  7 15:13:44] DEBUG[948][C-00000108]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '8.8.8.8:5060' into...
[Apr  7 15:13:44] DEBUG[948][C-00000108]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '8.8.8.8' and port '5060'.
[Apr  7 15:13:44] DEBUG[948][C-00000108]: chan_sip.c:3875 __sip_xmit: Trying to put 'ACK sip:891' onto UDP socket destined for 8.8.8.8:5060
    -- SIP/CTK-00000321 answered SIP/104-00000320
[Apr  7 15:13:44] DEBUG[688][C-00000108]: rtp_engine.c:1805 ast_rtp_instance_early_bridge: Setting early bridge SDP of 'SIP/104-00000320' with that of 'SIP/CTK-00000321'
[Apr  7 15:13:44] DEBUG[688][C-00000108]: chan_sip.c:7277 sip_answer: SIP answering channel: SIP/104-00000320
[Apr  7 15:13:44] DEBUG[688][C-00000108]: res_rtp_asterisk.c:2175 ast_rtp_update_source: Setting the marker bit due to a source update
[Apr  7 15:13:44] DEBUG[688][C-00000108]: chan_sip.c:13596 transmit_response_with_sdp: Setting framing from config on incoming call
[Apr  7 15:13:44] DEBUG[688][C-00000108]: chan_sip.c:13151 add_sdp: ** Our capability: (alaw) Video flag: True Text flag: True
[Apr  7 15:13:44] DEBUG[688][C-00000108]: chan_sip.c:13152 add_sdp: ** Our prefcodec: (nothing)
[Apr  7 15:13:44] DEBUG[688][C-00000108]: chan_sip.c:13288 add_sdp: -- Done with adding codecs to SDP
[Apr  7 15:13:44] DEBUG[688][C-00000108]: chan_sip.c:13482 add_sdp: Done building SDP. Settling with this capability: (alaw)
[Apr  7 15:13:44] DEBUG[688][C-00000108]: chan_sip.c:3875 __sip_xmit: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.175.132:5060
[Apr  7 15:13:44] DEBUG[688][C-00000108]: features.c:4429 ast_bridge_call: bridge answer set, chan answer set
[Apr  7 15:13:44] DEBUG[688][C-00000108]: features.c:4250 clear_dialed_interfaces: Removing dialed interfaces datastore on SIP/CTK-00000321 since we're bridging
[Apr  7 15:13:44] DEBUG[688][C-00000108]: res_rtp_asterisk.c:2175 ast_rtp_update_source: Setting the marker bit due to a source update
[Apr  7 15:13:44] DEBUG[688][C-00000108]: res_rtp_asterisk.c:2175 ast_rtp_update_source: Setting the marker bit due to a source update
    -- Locally bridging SIP/104-00000320 and SIP/CTK-00000321
[Apr  7 15:13:44] DEBUG[930]: devicestate.c:344 _ast_device_state: No provider found, checking channel drivers for SIP - CTK
[Apr  7 15:13:44] DEBUG[930]: chan_sip.c:29602 sip_devicestate: Checking device state for peer CTK
[Apr  7 15:13:44] DEBUG[930]: devicestate.c:467 do_state_change: Changing state for SIP/CTK - state 1 (Not in use)
[Apr  7 15:13:44] DEBUG[930]: devicestate.c:442 devstate_event: device 'SIP/CTK' state '1'
[Apr  7 15:13:44] DEBUG[930]: devicestate.c:344 _ast_device_state: No provider found, checking channel drivers for SIP - 104
[Apr  7 15:13:44] DEBUG[930]: chan_sip.c:29602 sip_devicestate: Checking device state for peer 104
[Apr  7 15:13:44] DEBUG[930]: devicestate.c:467 do_state_change: Changing state for SIP/104 - state 3 (Busy)
[Apr  7 15:13:44] DEBUG[930]: devicestate.c:442 devstate_event: device 'SIP/104' state '3'
[Apr  7 15:13:44] DEBUG[688][C-00000108]: res_rtp_asterisk.c:3581 ast_rtp_read: 0xb6b38340 -- Probation learning mode pass with source address 192.168.175.132:20598
       > 0xb6b38340 -- Probation passed - setting RTP source address to 192.168.175.132:20598
[Apr  7 15:13:44] DEBUG[948]: chan_sip.c:9167 find_call: = Looking for  Call ID: 00215503-06d70004-3a295f75-dbd12cb6@192.168.175.132 (Checking From) --From tag 0021550306d70051022e38a3-cc6b89c8 --To-tag as06c7460f
[Apr  7 15:13:44] DEBUG[948][C-00000108]: chan_sip.c:28146 handle_incoming: **** Received ACK (6) - Command in SIP ACK
[Apr  7 15:13:44] DEBUG[948][C-00000108]: chan_sip.c:4567 __sip_ack: Stopping retransmission on '00215503-06d70004-3a295f75-dbd12cb6@192.168.175.132' of Response 102: Match Found
[Apr  7 15:13:44] DEBUG[688][C-00000108]: res_rtp_asterisk.c:3581 ast_rtp_read: 0x80fd0ed8 -- Probation learning mode pass with source address 8.8.8.8:27226
       > 0x80fd0ed8 -- Probation passed - setting RTP source address to 8.8.8.8:27226
[Apr  7 15:13:45] DEBUG[948]: chan_sip.c:8770 sip_alloc: Allocating new SIP dialog for 0a084b09528415f44f19ae7e70a9a1d8@127.0.1.1:5060 - OPTIONS (No RTP)
[Apr  7 15:13:45] DEBUG[948]: acl.c:979 ast_ouraddrfor: For destination '192.168.175.132', our source address is '192.168.175.15'.
[Apr  7 15:13:45] DEBUG[948]: chan_sip.c:4032 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_UDP with address 192.168.175.15:5060
[Apr  7 15:13:45] DEBUG[948]: chan_sip.c:8565 change_callid_pvt: SIP call-id changed from '0a084b09528415f44f19ae7e70a9a1d8@127.0.1.1:5060' to '2a302cc64bc474da43c8a1902c8cd24d@192.168.175.15:5060'
[Apr  7 15:13:45] DEBUG[948]: chan_sip.c:3518 initialize_initreq: Initializing initreq for method OPTIONS - callid 2a302cc64bc474da43c8a1902c8cd24d@192.168.175.15:5060
[Apr  7 15:13:45] DEBUG[948]: chan_sip.c:3875 __sip_xmit: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.175.132:5060
[Apr  7 15:13:45] DEBUG[948]: chan_sip.c:9167 find_call: = Looking for  Call ID: 2a302cc64bc474da43c8a1902c8cd24d@192.168.175.15:5060 (Checking To) --From tag as59a79fdd --To-tag 0021550306d70052cbee0d59-fe179263
[Apr  7 15:13:45] DEBUG[948]: chan_sip.c:4567 __sip_ack: Stopping retransmission on '2a302cc64bc474da43c8a1902c8cd24d@192.168.175.15:5060' of Request 102: Match Found
[Apr  7 15:13:45] DEBUG[948]: chan_sip.c:6828 sip_destroy: Destroying SIP dialog 2a302cc64bc474da43c8a1902c8cd24d@192.168.175.15:5060
Asterisk*CLI> core set debug off
Core debug is now OFF
    -- Executing [h@LOCAL-PHONES:1] NoOp("SIP/100-0000031e", "") in new stack
    -- Executing [h@LOCAL-PHONES:2] Verbose("SIP/100-0000031e", "1,последняя командв 16, время ответа абонента 67, код ошибки ") in new stack
 последняя командв 16, время ответа абонента 67, код ошибки
  == Spawn extension (LOCAL-PHONES, 556677, 2) exited non-zero on 'SIP/100-0000031e'
    -- Executing [h@LOCAL-PHONES:1] NoOp("SIP/104-00000320", "") in new stack
    -- Executing [h@LOCAL-PHONES:2] Verbose("SIP/104-00000320", "1,последняя командв 16, время ответа абонента 47, код ошибки ") in new stack
 последняя командв 16, время ответа абонента 47, код ошибки
  == Spawn extension (LOCAL-PHONES, 89109998877, 3) exited non-zero on 'SIP/104-00000320'
Asterisk*CLI>
Знать бы что искать...
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ded
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Re: Проблема при одновременных вызовах

Сообщение ded »

Сами то пробовали читать? Ничего не насторожило? -- Это вы с Гуглем разговариваете?

Код: Выделить всё

[Apr  7 15:13:44] DEBUG[948][C-00000108]: chan_sip.c:6680 update_call_counter: Updating call counter for outgoing call
[Apr  7 15:13:44] DEBUG[948][C-00000108]: chan_sip.c:16241 build_route: build_route: Contact hop: <sip:89109998877@8.8.8.8:5060;transport=udp>
[Apr  7 15:13:44] DEBUG[948][C-00000108]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '8.8.8.8:5060' into...
[Apr  7 15:13:44] DEBUG[948][C-00000108]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '8.8.8.8' and port '5060'.
[Apr  7 15:13:44] DEBUG[948][C-00000108]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '8.8.8.8:5060' into...
[Apr  7 15:13:44] DEBUG[948][C-00000108]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '8.8.8.8' and port '5060'.
[Apr  7 15:13:44] DEBUG[948][C-00000108]: chan_sip.c:3875 __sip_xmit: Trying to put 'ACK sip:891' onto UDP socket destined for 8.8.8.8:5060
    -- SIP/CTK-00000321 answered SIP/104-00000320

[Apr  7 15:13:44] DEBUG[688][C-00000108]: res_rtp_asterisk.c:3581 ast_rtp_read: 0x80fd0ed8 -- Probation learning mode pass with source address 8.8.8.8:27226
       > 0x80fd0ed8 -- Probation passed - setting RTP source address to 8.8.8.8:27226
[Apr  7 15:13:45] DEBUG[948]: chan_sip.c:8770 sip_alloc: Allocating new SIP dialog for 0a084b09528415f44f19ae7e70a9a1d8@127.0.1.1:5060 - OPTIONS (No RTP)
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RamonSantiago
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Re: Проблема при одновременных вызовах

Сообщение RamonSantiago »

Ну если Вы имеете в виду адрес 8.8.8.8, то все белые адреса и номера телефонов сознательно изменены....
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ded
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Re: Проблема при одновременных вызовах

Сообщение ded »

Как об этом можно было догадаться?
Вместо таких выкрутасов достаточно было оставить реальные ИП, изменив только первый октет.
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RamonSantiago
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Re: Проблема при одновременных вызовах

Сообщение RamonSantiago »

Согласен, следовало предупредить о замене адресов.
Меня смущает вот эта часть:

Код: Выделить всё

[Apr  7 15:13:44] DEBUG[948][C-00000108]: chan_sip.c:4529 __sip_ack: Acked pending invite 103
[Apr  7 15:13:44] DEBUG[948][C-00000108]: chan_sip.c:4567 __sip_ack: Stopping retransmission on '7ef59bb00593969443f32b14739fe827@0000719.yr.centertelecom.ru' of Request 103: Match Found
[Apr  7 15:13:44] DEBUG[948][C-00000108]: chan_sip.c:22601 handle_response_invite: SIP response 200 to standard invite
[Apr  7 15:13:44] DEBUG[948][C-00000108]: chan_sip.c:10049 process_sdp: Processing session-level SDP v=0... UNSUPPORTED OR FAILED.
[Apr  7 15:13:44] DEBUG[948][C-00000108]: chan_sip.c:10049 process_sdp: Processing session-level SDP o=user 182827 182827 IN IP4 8.8.8.8... OK.
[Apr  7 15:13:44] DEBUG[948][C-00000108]: chan_sip.c:10049 process_sdp: Processing session-level SDP s=call... UNSUPPORTED OR FAILED.
[Apr  7 15:13:44] DEBUG[948][C-00000108]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '8.8.8.8' into...
[Apr  7 15:13:44] DEBUG[948][C-00000108]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '8.8.8.8' and port ''.
[Apr  7 15:13:44] DEBUG[948][C-00000108]: chan_sip.c:10049 process_sdp: Processing session-level SDP c=IN IP4 8.8.8.8... OK.
[Apr  7 15:13:44] DEBUG[948][C-00000108]: chan_sip.c:10049 process_sdp: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED.
[Apr  7 15:13:44] DEBUG[948][C-00000108]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 8 based on m type on 0xb409edc0
[Apr  7 15:13:44] DEBUG[948][C-00000108]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 101 based on m type on 0xb409edc0
[Apr  7 15:13:44] DEBUG[948][C-00000108]: chan_sip.c:10482 process_sdp: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK.
[Apr  7 15:13:44] DEBUG[948][C-00000108]: chan_sip.c:10482 process_sdp: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK.
[Apr  7 15:13:44] DEBUG[948][C-00000108]: chan_sip.c:10482 process_sdp: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED OR FAILED.
[Apr  7 15:13:44] DEBUG[948][C-00000108]: chan_sip.c:10482 process_sdp: Processing media-level (audio) SDP a=ptime:20... OK.
[Apr  7 15:13:44] DEBUG[948][C-00000108]: chan_sip.c:10482 process_sdp: Processing media-level (audio) SDP a=sendrecv... OK.
[Apr  7 15:13:44] DEBUG[948][C-00000108]: res_rtp_asterisk.c:3994 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x80fcc5f4'
[Apr  7 15:13:44] DEBUG[948][C-00000108]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 8 from 0xb409edc0 to 0x80fcc7a0
[Apr  7 15:13:44] DEBUG[948][C-00000108]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 101 from 0xb409edc0 to 0x80fcc7a0
Но что это значит, я не пойму...
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ded
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Re: Проблема при одновременных вызовах

Сообщение ded »

Я ещё сокращу:

Код: Выделить всё

[Apr  7 15:13:44] DEBUG[948][C-00000108]: chan_sip.c:10049 process_sdp: Processing session-level SDP v=0... UNSUPPORTED OR FAILED.
[Apr  7 15:13:44] DEBUG[948][C-00000108]: chan_sip.c:10049 process_sdp: Processing session-level SDP o=user 182827 182827 IN IP4 8.8.8.8... OK.
[Apr  7 15:13:44] DEBUG[948][C-00000108]: chan_sip.c:10049 process_sdp: Processing session-level SDP s=call... UNSUPPORTED OR FAILED.
[Apr  7 15:13:44] DEBUG[948][C-00000108]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '8.8.8.8' into...
[Apr  7 15:13:44] DEBUG[948][C-00000108]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '8.8.8.8' and port ''.
Адрес есть, а порт не определён в SDP, чтобы понять - нужно spi set debug ip 8.8.8.8 и рассмотреть этот самый инвайт и SDP. Также строки CLI вокруг этого инвайта дадут представление о происходящем. Вы главное читайте, читайте, самостоятельно. Там всё для людей инфа, ибо компютеру разъяснения текстом не нужны.
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