Всем привет! Asterisk 1.4.29 centOS 5.4 При входящем звонке начинает звонить телефон агента, при попытке снять звонок в трубке тишина (примерно 10 сек. потом hangup) через какое то время снова поступает звонок и все повторяется... у звонящего играет мелодия как положено, через каждую минуту ему сообщается какой он в очереди фрагмент extensions
-- Executing [s@callcenter:1] Answer("OOH323/Cisco5300-3410", "") in new stack -- Executing [s@callcenter:2] Ringing("OOH323/Cisco5300-3410", "") in new stack -- Executing [s@callcenter:3] Wait("OOH323/Cisco5300-3410", "2") in new stack -- Executing [s@callcenter:4] Queue("OOH323/Cisco5300-3410", "MyQueue|t|||100") in new stack -- Started music on hold, class 'default', on OOH323/Cisco5300-3410 -- outgoing agentcall, to agent '013', on 'Local/600@callcenter-d0a4,1' -- Executing [600@callcenter:1] Dial("Local/600@callcenter-d0a4,2", "SIP/600") in new stack -- Called 600 -- SIP/600-00000034 is ringing -- SIP/600-00000034 answered Local/600@callcenter-d0a4,2 [Apr 14 12:00:31] DEBUG[5863]: chan_agent.c:612 agent_read: Bridge on 'SIP/600-00000034' being set to 'Agent/013' (3) [Apr 14 12:00:31] DEBUG[5863]: chan_agent.c:499 agent_read: Native formats changing from 256 to 4 [Apr 14 12:00:31] DEBUG[5863]: chan_agent.c:499 agent_read: Resetting read to 256 and write to 256 == Spawn extension (callcenter, 600, 1) exited non-zero on 'Local/600@callcenter-d0a4,2' -- Stopped music on hold on OOH323/Cisco5300-3410 [Apr 14 12:00:36] DEBUG[5863]: chan_agent.c:904 agent_hangup: Hungup, howlong is 8, autologoff is 0 == Spawn extension (callcenter, s, 4) exited non-zero on 'OOH323/Cisco5300-3410'
это кодеки. из вашего дебага имеет ценность только пара строчек:
Quote:
[Apr 14 12:00:31] DEBUG[5863]: chan_agent.c:499 agent_read: Native formats changing from 256 to 4 [Apr 14 12:00:31] DEBUG[5863]: chan_agent.c:499 agent_read: Resetting read to 256 and write to 256
теперь смотрим в таблицу и находим причину:
Code:
The result of core show codecs is as bellow
Disclaimer: this command is for informational purposes only. It does not indicate anything about your configuration. INT BINARY HEX TYPE NAME DESC -------------------------------------------------------------------------------- 1 (1
если честно, то были предположение на это, но думал ошибаюсь, Anest не посчитайте за наглость подскажите как исправить, своими силами не справляюсь, а нужно срочно в sip.conf запретил alaw и ulaw оставил g729, вот что пишет в консоли
Code:
-- Executing [s@callcenter:1] Answer("OOH323/Cisco5300-0573", "") in new stack -- Executing [s@callcenter:2] Ringing("OOH323/Cisco5300-0573", "") in new stack -- Executing [s@callcenter:3] Wait("OOH323/Cisco5300-0573", "2") in new stack -- Executing [s@callcenter:4] Queue("OOH323/Cisco5300-0573", "MyQueue|t|||100") in new stack -- Started music on hold, class 'default', on OOH323/Cisco5300-0573 -- outgoing agentcall, to agent '013', on 'Local/600@callcenter-4695,1' -- Executing [600@callcenter:1] Dial("Local/600@callcenter-4695,2", "SIP/600") in new stack -- Called 600 -- SIP/600-00000000 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'Local/600@callcenter-4695,2' status is 'CONGESTION'
то есть получается что на g729 он не хочет работать??? CISCO работает на g729
Лучше всего отлаживать на 711 кодеке, ибо он есть везде. Как заработает, уже можете пытатся оптимизацией заняться. _________________ Intel Core 2 Duo E6400 @ 2.40GHz / 6GB / 160GB Gentoo Linux 2.6.32-r7 || Asterisk 1.6.2.9 | SFA | Linksys SPA922 + D-Link DPH-300S + D-Link DVG-7111S + 3 x Huawei E1550
я понимаю, но уже все поднято на g729! являемся провайдерами телефонии... тогда все менять(((( эта проблема только на входящие, исходящие гуляют тоже с этой циски в g729
Added after 1 minutes:
artyr_n wrote:
а G729 у Вас установлен?
Code:
show translation
Code:
Asterisk*CLI> show translation Translation times between formats (in milliseconds) for one second of data Source Format (Rows) Destination Format (Columns)
проблема сохраняется! при этом если не использовать агента, то все прекрасно работает! и еще, может тупой вопрос -почему * не конвертирует g729 в ulaw???
Code:
[Apr 14 12:00:31] DEBUG[5863]: chan_agent.c:499 agent_read: Native formats changing from 256 to 4 [Apr 14 12:00:31] DEBUG[5863]: chan_agent.c:499 agent_read: Resetting read to 256 and write to 256
на циске запустил u-law, проблема сохранилась! (контекст callcenter переименован в 417770)
Code:
== Starting OOH323/Cisco5300-a46f at 417770,417770,1 failed so falling back to exten 's' -- Executing [s@417770:1] Answer("OOH323/Cisco5300-a46f", "") in new stack -- Executing [s@417770:2] Ringing("OOH323/Cisco5300-a46f", "") in new stack -- Executing [s@417770:3] Wait("OOH323/Cisco5300-a46f", "2") in new stack -- Executing [s@417770:4] Queue("OOH323/Cisco5300-a46f", "MyQueue|t|||100") in new stack -- Started music on hold, class 'default', on OOH323/Cisco5300-a46f -- outgoing agentcall, to agent '013', on 'Local/013@417770-1e36,1' -- Executing [013@417770:1] Dial("Local/013@417770-1e36,2", "SIP/013") in new stack -- Called 013 -- SIP/013-00000003 is ringing -- SIP/013-00000003 answered Local/013@417770-1e36,2 [Apr 21 12:12:49] DEBUG[2949]: chan_agent.c:612 agent_read: Bridge on 'SIP/013-00000003' being set to 'Agent/013' (3) == Spawn extension (417770, 013, 1) exited non-zero on 'Local/013@417770-1e36,2' -- Nobody picked up in 15000 ms [Apr 21 12:12:56] DEBUG[2949]: chan_agent.c:904 agent_hangup: Hungup, howlong is 15, autologoff is 0
Added after 12 minutes:
Code:
-- outgoing agentcall, to agent '013', on 'Local/013@417770-c9f7,1' [Apr 21 12:23:22] DEBUG[2985]: pbx.c:1859 pbx_extension_helper: Launching 'Dial' -- Executing [013@417770:1] Dial("Local/013@417770-c9f7,2", "SIP/013") in new stack [Apr 21 12:23:22] DEBUG[2985]: chan_sip.c:17313 sip_request_call: Asked to create a SIP channel with formats: 0x4 (ulaw) [Apr 21 12:23:22] DEBUG[2985]: chan_sip.c:4797 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Apr 21 12:23:22] DEBUG[2985]: chan_sip.c:2894 do_setnat: Setting NAT on RTP to On [Apr 21 12:23:22] DEBUG[2985]: rtp.c:1653 ast_rtp_make_compatible: Channel 'Local/013@417770-c9f7,2' has no RTP, not doing anything [Apr 21 12:23:22] DEBUG[2985]: channel.c:3759 ast_channel_inherit_variables: Not copying variable DIALEDTIME. [Apr 21 12:23:22] DEBUG[2985]: channel.c:3759 ast_channel_inherit_variables: Not copying variable ANSWEREDTIME. [Apr 21 12:23:22] DEBUG[2985]: channel.c:3759 ast_channel_inherit_variables: Not copying variable DIALEDPEERNAME. [Apr 21 12:23:22] DEBUG[2985]: channel.c:3759 ast_channel_inherit_variables: Not copying variable DIALEDPEERNUMBER. [Apr 21 12:23:22] DEBUG[2985]: channel.c:3759 ast_channel_inherit_variables: Not copying variable DIALSTATUS. [Apr 21 12:23:22] DEBUG[2985]: chan_sip.c:3185 sip_call: Outgoing Call for 013 -- Called 013 [Apr 21 12:23:22] DEBUG[2985]: channel.c:3150 set_format: Set channel SIP/013-0000000f to read format slin [Apr 21 12:23:22] DEBUG[2985]: channel.c:3150 set_format: Set channel Local/013@417770-c9f7,2 to write format slin [Apr 21 12:23:22] DEBUG[2985]: channel.c:3150 set_format: Set channel Local/013@417770-c9f7,2 to read format slin [Apr 21 12:23:22] DEBUG[2984]: channel.c:3150 set_format: Set channel OOH323/Cisco5300-0a39 to write format slin [Apr 21 12:23:22] DEBUG[2984]: res_musiconhold.c:261 ast_moh_files_next: OOH323/Cisco5300-0a39 Opened file 1 '/var/lib/asterisk/moh/reno_project-system' [Apr 21 12:23:22] DEBUG[2985]: channel.c:3150 set_format: Set channel SIP/013-0000000f to write format slin [Apr 21 12:23:22] DEBUG[2915]: chan_sip.c:2313 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '510f64775b75c09b09119fd31c2736c9@8xx.23x.xxx.7' Request 102: Found -- SIP/013-0000000f is ringing [Apr 21 12:23:22] DEBUG[2985]: rtp.c:1564 ast_rtp_early_bridge: Channel 'Local/013@417770-c9f7,2' has no RTP, not doing anything [Apr 21 12:23:22] DEBUG[2984]: rtp.c:923 ast_rtcp_read: Got RTCP report of 132 bytes [Apr 21 12:23:22] DEBUG[2915]: chan_sip.c:2239 __sip_ack: Acked pending invite 102 [Apr 21 12:23:22] DEBUG[2915]: chan_sip.c:2271 __sip_ack: Stopping retransmission on '510f64775b75c09b09119fd31c2736c9@8xx.23x.xxx.7' of Request 102: Match Found [Apr 21 12:23:22] DEBUG[2915]: channel.c:3150 set_format: Set channel SIP/013-0000000f to read format slin [Apr 21 12:23:22] DEBUG[2915]: channel.c:3150 set_format: Set channel SIP/013-0000000f to write format slin -- SIP/013-0000000f answered Local/013@417770-c9f7,2 [Apr 21 12:23:22] DEBUG[2985]: rtp.c:1564 ast_rtp_early_bridge: Channel 'Local/013@417770-c9f7,2' has no RTP, not doing anything [Apr 21 12:23:22] DEBUG[2919]: app_queue.c:706 handle_statechange: Device 'Agent/013' changed to state '3' (Busy) [Apr 21 12:23:22] DEBUG[2985]: rtp.c:1233 ast_rtp_read: RTP NAT: Got audio from other end. Now sending to address 8xx.23x.xxx.5:21796 [Apr 21 12:23:22] DEBUG[2985]: channel.c:3692 ast_channel_masquerade: Planning to masquerade channel SIP/013-0000000f into the structure of Local/013@417770-c9f7,1 [Apr 21 12:23:22] DEBUG[2985]: channel.c:3706 ast_channel_masquerade: Done planning to masquerade channel SIP/013-0000000f into the structure of Local/013@417770-c9f7,1 [Apr 21 12:23:22] DEBUG[2985]: chan_local.c:339 local_write: Not posting to queue since already masked on 'Local/013@417770-c9f7,2' [Apr 21 12:23:22] DEBUG[2984]: channel.c:4033 ast_do_masquerade: Putting channel SIP/013-0000000f in 4/4 formats [Apr 21 12:23:22] DEBUG[2984]: channel.c:4081 ast_do_masquerade: Released clone lock on 'Local/013@417770-c9f7,1' [Apr 21 12:23:22] DEBUG[2984]: channel.c:4091 ast_do_masquerade: Done Masquerading SIP/013-0000000f (6) [Apr 21 12:23:22] DEBUG[2984]: chan_agent.c:612 agent_read: Bridge on 'SIP/013-0000000f' being set to 'Agent/013' (3) [Apr 21 12:23:22] DEBUG[2985]: channel.c:4295 ast_generic_bridge: Didn't get a frame from channel: Local/013@417770-c9f7,2 [Apr 21 12:23:22] DEBUG[2985]: channel.c:4662 ast_channel_bridge: Bridge stops bridging channels Local/013@417770-c9f7,2 and Local/013@417770-c9f7,1 [Apr 21 12:23:22] DEBUG[2985]: chan_agent.c:1045 agent_bridgedchannel: Asked for bridged channel on 'SIP/013-0000000f'/'Agent/013', returning '' [Apr 21 12:23:22] DEBUG[2985]: channel.c:1570 ast_hangup: Hanging up zombie 'Local/013@417770-c9f7,1' [Apr 21 12:23:22] DEBUG[2985]: rtp.c:1564 ast_rtp_early_bridge: Channel 'Local/013@417770-c9f7,2' has no RTP, not doing anything [Apr 21 12:23:22] DEBUG[2985]: app_dial.c:1901 dial_exec_full: Exiting with DIALSTATUS=ANSWER. [Apr 21 12:23:22] DEBUG[2985]: pbx.c:2409 __ast_pbx_run: Spawn extension (417770,013,1) exited non-zero on 'Local/013@417770-c9f7,2' == Spawn extension (417770, 013, 1) exited non-zero on 'Local/013@417770-c9f7,2' [Apr 21 12:23:22] DEBUG[2985]: channel.c:1462 ast_softhangup_nolock: Soft-Hanging up channel 'Local/013@417770-c9f7,2' [Apr 21 12:23:22] DEBUG[2985]: channel.c:1565 ast_hangup: Hanging up channel 'Local/013@417770-c9f7,2' [Apr 21 12:23:23] DEBUG[2915]: chan_sip.c:4797 sip_alloc: Allocating new SIP dialog for NmM5NmUzZmVjNzYwYzgwZjVmMmUyZDkxZTJiYzc3YTg. - REGISTER (No RTP) [Apr 21 12:23:23] DEBUG[2915]: res_config_mysql.c:657 mysql_reconnect: MySQL RealTime: Everything is fine. [Apr 21 12:23:23] DEBUG[2915]: res_config_mysql.c:356 update_mysql: MySQL RealTime: Update SQL: UPDATE sip_buddies SET ipaddr = '8xx.23x.xxx.5', port = '4009', regseconds = '1271838293', username = '013', fullcontact = 'sip:013@8xx.23x.xxx.5:4009;rinstance=22843f3242d31aaf' WHERE name = '013' [Apr 21 12:23:23] DEBUG[2915]: res_config_mysql.c:370 update_mysql: MySQL RealTime: Updated 1 rows on table: sip_buddies [Apr 21 12:23:23] DEBUG[2919]: app_queue.c:706 handle_statechange: Device 'Agent/013' changed to state '3' (Busy) [Apr 21 12:23:24] DEBUG[2915]: db.c:240 ast_db_del: Unable to find key '003' in family 'SIP/Registry' [Apr 21 12:23:24] DEBUG[2906]: res_config_mysql.c:657 mysql_reconnect: MySQL RealTime: Everything is fine. [Apr 21 12:23:24] DEBUG[2906]: res_config_mysql.c:140 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_buddies WHERE name = '003' AND host = 'dynamic' [Apr 21 12:23:24] DEBUG[2915]: chan_sip.c:4797 sip_alloc: Allocating new SIP dialog for N2ZkZmFiMDJkYjkwMzhlNzZkNGQ0NjM2Mzg3MWNkNmE. - SUBSCRIBE (No RTP) [Apr 21 12:23:24] DEBUG[2915]: res_config_mysql.c:657 mysql_reconnect: MySQL RealTime: Everything is fine. [Apr 21 12:23:24] DEBUG[2915]: res_config_mysql.c:140 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_buddies WHERE name = '003' AND host = 'dynamic' [Apr 21 12:23:24] DEBUG[2915]: db.c:240 ast_db_del: Unable to find key '003' in family 'SIP/Registry' [Apr 21 12:23:24] DEBUG[2915]: chan_sip.c:18192 build_peer: Bah, we're expired (1271838204/0/1271838204)! [Apr 21 12:23:24] DEBUG[2915]: db.c:196 ast_db_get: Unable to find key '003' in family 'SIP/Registry' [Apr 21 12:23:24] DEBUG[2915]: chan_sip.c:8857 build_route: build_route: Retaining previous route: [Apr 21 12:23:24] NOTICE[2915]: chan_sip.c:16212 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 003 [Apr 21 12:23:24] DEBUG[2984]: rtp.c:923 ast_rtcp_read: Got RTCP report of 60 bytes [Apr 21 12:23:25] DEBUG[2984]: rtp.c:923 ast_rtcp_read: Got RTCP report of 132 bytes [Apr 21 12:23:26] DEBUG[2915]: chan_sip.c:2176 __sip_autodestruct: Auto destroying SIP dialog '201361109411464-28763186710086@192.168.0.100' [Apr 21 12:23:29] DEBUG[2984]: rtp.c:923 ast_rtcp_read: Got RTCP report of 60 bytes [Apr 21 12:23:32] DEBUG[2984]: rtp.c:923 ast_rtcp_read: Got RTCP report of 132 bytes [Apr 21 12:23:32] DEBUG[2915]: chan_sip.c:4797 sip_alloc: Allocating new SIP dialog for 128712559-2294730746@192.168.0.100 - REGISTER (No RTP) [Apr 21 12:23:32] DEBUG[2915]: chan_sip.c:4797 sip_alloc: Allocating new SIP dialog for 261775356-2964429372@192.168.0.100 - REGISTER (No RTP) [Apr 21 12:23:32] DEBUG[2915]: res_config_mysql.c:657 mysql_reconnect: MySQL RealTime: Everything is fine. [Apr 21 12:23:32] DEBUG[2915]: res_config_mysql.c:356 update_mysql: MySQL RealTime: Update SQL: UPDATE sip_buddies SET ipaddr = '8xx.23x.xxx.5', port = '62223', regseconds = '1271838272', username = '001', fullcontact = 'sip:001@83.239.194.5:62223' WHERE name = '001' [Apr 21 12:23:32] DEBUG[2915]: res_config_mysql.c:370 update_mysql: MySQL RealTime: Updated 1 rows on table: sip_buddies [Apr 21 12:23:32] DEBUG[2915]: res_config_mysql.c:657 mysql_reconnect: MySQL RealTime: Everything is fine. [Apr 21 12:23:32] DEBUG[2915]: res_config_mysql.c:356 update_mysql: MySQL RealTime: Update SQL: UPDATE sip_buddies SET ipaddr = '8x.23x.xxx.5', port = '62223', regseconds = '1271838272', username = '002', fullcontact = 'sip:002@8xx.23x.xxx.5:62223' WHERE name = '002' [Apr 21 12:23:32] DEBUG[2915]: res_config_mysql.c:370 update_mysql: MySQL RealTime: Updated 1 rows on table: sip_buddies [Apr 21 12:23:35] DEBUG[2984]: rtp.c:923 ast_rtcp_read: Got RTCP report of 132 bytes [Apr 21 12:23:35] DEBUG[2984]: rtp.c:923 ast_rtcp_read: Got RTCP report of 60 bytes -- Nobody picked up in 15000 ms [Apr 21 12:23:37] DEBUG[2984]: channel.c:1565 ast_hangup: Hanging up channel 'Agent/013' [Apr 21 12:23:37] DEBUG[2984]: chan_agent.c:873 agent_hangup: Hangup called for state Down [Apr 21 12:23:37] DEBUG[2984]: channel.c:1565 ast_hangup: Hanging up channel 'SIP/013-0000000f' [Apr 21 12:23:37] DEBUG[2984]: chan_sip.c:3709 sip_hangup: Hangup call SIP/013-0000000f, SIP callid 510f64775b75c09b09119fd31c2736c9@8xx.23x.xxx.7) [Apr 21 12:23:37] DEBUG[2984]: chan_agent.c:904 agent_hangup: Hungup, howlong is 15, autologoff is 0 [Apr 21 12:23:37] DEBUG[2919]: app_queue.c:706 handle_statechange: Device 'Agent/013' changed to state '1' (Not in use) [Apr 21 12:23:37] DEBUG[2919]: app_queue.c:706 handle_statechange: Device 'Agent/013' changed to state '1' (Not in use) [Apr 21 12:23:37] DEBUG[2919]: app_queue.c:706 handle_statechange: Device 'Agent/013' changed to state '1' (Not in use) [Apr 21 12:23:38] DEBUG[2915]: chan_sip.c:2271 __sip_ack: Stopping retransmission on '510f64775b75c09b09119fd31c2736c9@8xx.23x.xxx.7' of Request 103: Match Found [Apr 21 12:23:39] DEBUG[2915]: chan_sip.c:4797 sip_alloc: Allocating new SIP dialog for NDY3OGNhZDJjMWUyMTIwZDNhNzBjOTMxZDJmNTc3NmQ. - SUBSCRIBE (No RTP)
Может таки стоит присмотреться к строчке Код: [Apr 26 10:43:11] DEBUG[23880] rtp.c: RTP NAT: Got audio from other end. Now sending to address 8x.2xx.1xx.5:63759 из аттача? Плюс, в файле вообще нет rtp-потока от sip-пира, который снял трубку.
в том то и дело что при ответе sip ничего не происходит, по этому и не шлет трафик! у звонящего как играла мелодия, так и играет...
Code:
Got audio from other end. Now sending to address 8x.2xx.1xx.5:63759
sip висит на этом адресе и не отвечает (звонок на него поступает, поднимаю трубку и ничего не происходит), вот я и не пойму что происходит, скорее это глюк этой ветки астериска потому как с 1.6 версией такого не было. Не знаю, может на 1.4.27 перейти?
