Подключение АТС Yeastar N824 к Asterisk

Материал из Xgu.ru

Перейти к: навигация, поиск
stub.png
Данная страница находится в разработке.
Эта страница ещё не закончена. Информация, представленная здесь, может оказаться неполной или неверной.

Если вы считаете, что её стоило бы доработать как можно быстрее, пожалуйста, скажите об этом.


Автор: Telworks

[править] Подключение АТС Yeastar N824 к Asterisk

Данный документ описывает, как подключить АТС Yeastar N824 к Asterisk, для выполнения звонков с номеров на АТС на номера в Asterisk и обратно напрямую, путем набора номера.

В варианте "из коробки" на АТС Yeastar N824 поддерживает 8 каналов SIP в режиме регистрации на сервере провайдера. При регистрации N824 на номере в Asterisk, звонки, исходящие из станции могут быть направлены на внутренние номера Asterisk. Но все входящие вызовы в АТС не могут быть распределены по внутреннему пулу номеров на Yeastar N824, а могут быть направлены на сервисы АТС например на голосовое меню N824, группу, очередь и т.д.

В следующих версиях ПО Yeastar N824 производитель обещал поддержку данной возможности с помощью каналов SIP без регистрации.


Исходные данные (см. рис.):

1. SIP телефон, зарегистрированный на Asterisk под номером 382

2. АТС Yeastar N824 с подключенным аналоговым телефоном 601

Требуется реализовать: возможность звонков из телефона 382 (и всех телефонов в диапазоне 300-399) напрямую на номер 601 (и все телефоны в диапазоне 601-624, 501-508) и возможность звонков из телефона 601 (и всех телефонов в диапазоне 601-624, 501-508) напрямую на номер 382 (и все телефоны в диапазоне 300-399).

VoipDiagramm.png


Для подключения АТС Yeastar N824 необходимо выполнить следующие действия:

1. Создать в Asterisk аккаунт для регистрации станции Yeastar N824

Аккаунт в Asterisk можно создать в файле:

/etc/asterisk/sip.conf 

................
[ASTER-N824]
type=friend
context=in_office
username = ASTER-N824
secret=pincode
host=dynamic
disallow = all
allow = alaw,ulaw,gsm
..................

2. Создать в Yeastar N824 линию подключения к Asterisk

Trunk-fromN824toAster.jpg

2.1. Найти параметры созданного транка по названию в конфигурационном файле на Yeastar N824

В командной строке на станции Yeastar N824 вывести содержимое файла «pjsip»: 

cat /etc/asterisk/pjsip.conf |less

Найти по названию линии(показано на рис синим цветом):

PJSIPconf on N824 native.jpg

2.2. Настроить контекст в Yeastar N824 для найденной линии

Создать и записать в файл через консоль SSH N824: /persistent/custom-cfg/pjsip_custom.conf

Например, с помощью редактора vi 

vi /persistent/custom-cfg/pjsip_custom.conf



[trunk-TrunkToASTERISK-endpoint](trunk-endpoint-basic)
context = Local_Default_CallingRules

Применить настройку, открытием и сохранением панели настройки линии, и нажав кнопку Применить.

2.3. Проверить, что контекст в линии изменился.

В командной строке на станции Yeastar N824 вывести содержимое файла «pjsip»

В настройках транка должен появиться указанный нами контекст (на рис показано красным цветом):

"context = Local_Default_CallingRules" 

cat /etc/asterisk/pjsip.conf |less

PJSIPconf on N824 custom.jpg

3. Настроить маршрутизацию вызовов из Yeastar N824 в Asterisk

Route-fromN824toAster.jpg

4. Настроить маршрутизацию вызовов из Asterisk в Yeastar N824

В Asterisk маршрутизацию вызовов можно задать в файле плана набора:

/etc/asterisk/extensions.conf

Например, направлять вызовы при наборе 5 и 6 в линию - на N824.

И направлять вызовы из линии N824 АТС, на номера 300-399, на которые зарегистрированы IP-телефоны.

Фрагмент /etc/asterisk/extensions.conf: 

[in_office]
………………………
exten => _[65].,1,Dial(SIP/ASTER-N824/${EXTEN})
……………………..
exten => _3XX,1,Dial(SIP/${EXTEN})
…………………….

[править] Проверка маршрутизации звонков из Yeastar N824 в Asterisk

На аналоговом телефоне 601 Yeastar N824 набираем номер 382 (на 382 в Asterisk зарегистрирован IP-телефон SIP).

IP-телефон SIP, 382 зарегистрированный на Asterisk должен зазвонить.

На Yeastar N824 вывод лога консоли: 

N824*CLI>
    -- Starting simple switch on 'DAHDI/1-1'
[2015-12-24 12:22:25] WARNING[3090][C-00000000]: chan_dahdi.c:5108 dahdi_ec_enable: Enabled echo cancellation on channel 1
    -- Executing [382@DLPN_DialPlan601:1] Set("DAHDI/1-1", "agiresult=0") in new stack
    -- Executing [382@DLPN_DialPlan601:2] GotoIf("DAHDI/1-1", "0?play-no-balance,382,1") in new stack
    -- Executing [382@DLPN_DialPlan601:3] GotoIf("DAHDI/1-1", "0?play-ext-disabled,382,1") in new stack
    -- Executing [382@DLPN_DialPlan601:4] GotoIf("DAHDI/1-1", "0?ext-no-rate,382,1") in new stack
    -- Executing [382@DLPN_DialPlan601:5] NoOp("DAHDI/1-1", "no pinset") in new stack
    -- Executing [382@DLPN_DialPlan601:6] Set("DAHDI/1-1", "ORGINEXTEN=382") in new stack
    -- Executing [382@DLPN_DialPlan601:7] Set("DAHDI/1-1", "ORGINROUTE=TO_ASTERISK") in new stack
    -- Executing [382@DLPN_DialPlan601:8] Set("DAHDI/1-1", "ORGINCONTEXT=DLPN_DialPlan601") in new stack
    -- Executing [382@DLPN_DialPlan601:9] GetNextOutRouter("DAHDI/1-1", "DLPN_DialPlan601,") in new stack
    -- Executing [382@DLPN_DialPlan601:10] GotoByTimeConditionOutBound("DAHDI/1-1", ",0") in new stack
    -- Executing [382@DLPN_DialPlan601:11] Macro("DAHDI/1-1", "trunkdial-failover-0.3,1,,382,trunk-TrunkToASTERISK-endpoint") in new stack
    -- Executing [s@macro-trunkdial-failover-0.3:1] NoOp("DAHDI/1-1", "do call out") in new stack
    -- Executing [s@macro-trunkdial-failover-0.3:2] GotoIf("DAHDI/1-1", "0?Blacklist-Handle,s,1") in new stack
    -- Executing [s@macro-trunkdial-failover-0.3:3] GotoIf("DAHDI/1-1", "0?6:4)}") in new stack
    -- Goto (macro-trunkdial-failover-0.3,s,4)
    -- Executing [s@macro-trunkdial-failover-0.3:4] GotoIf("DAHDI/1-1", "0?5:6") in new stack
    -- Goto (macro-trunkdial-failover-0.3,s,6)
    -- Executing [s@macro-trunkdial-failover-0.3:6] Set("DAHDI/1-1", "TCOUNT=4") in new stack
    -- Executing [s@macro-trunkdial-failover-0.3:7] Set("DAHDI/1-1", "CDR(userfield)=Outbound") in new stack
    -- Executing [s@macro-trunkdial-failover-0.3:8] Set("DAHDI/1-1", "OldCallerID=601") in new stack
    -- Executing [s@macro-trunkdial-failover-0.3:9] Set("DAHDI/1-1", "OldCallerID=601") in new stack
    -- Executing [s@macro-trunkdial-failover-0.3:10] Set("DAHDI/1-1", "TOUCH_MONITOR=601-382") in new stack
    -- Executing [s@macro-trunkdial-failover-0.3:11] NoOp("DAHDI/1-1", "") in new stack
    -- Executing [s@macro-trunkdial-failover-0.3:12] Set("DAHDI/1-1", "TIMEOUT(absolute)=6000") in new stack
    -- Channel will hangup at 2015-12-24 14:02:26.059 GMT-3.
    -- Executing [s@macro-trunkdial-failover-0.3:13] Set("DAHDI/1-1", "DLSTAT=UNKNOW}") in new stack
    -- Executing [s@macro-trunkdial-failover-0.3:14] SetCktCustom("DAHDI/1-1", "sendrpid,no,no") in new stack
    -- Executing [s@macro-trunkdial-failover-0.3:15] GotoIf("DAHDI/1-1", "1>0?1-dial,1") in new stack
    -- Goto (macro-trunkdial-failover-0.3,1-dial,1)
    -- Executing [1-dial@macro-trunkdial-failover-0.3:1] GotoIf("DAHDI/1-1", "0?nextrouter,1") in new stack
    -- Executing [1-dial@macro-trunkdial-failover-0.3:2] GotoIf("DAHDI/1-1", "0?setdod,1:1-dial,3") in new stack
    -- Goto (macro-trunkdial-failover-0.3,1-dial,3)
    -- Executing [1-dial@macro-trunkdial-failover-0.3:3] Set("DAHDI/1-1", "CALLERID(name)=601") in new stack
    -- Executing [1-dial@macro-trunkdial-failover-0.3:4] Set("DAHDI/1-1", "CALLERID(num)=601") in new stack
[2015-12-24 12:22:26] ERROR[3090][C-00000000]: pbx.c:4344 ast_func_read: Function SIPPEER not registered
    -- Executing [1-dial@macro-trunkdial-failover-0.3:5] Set("DAHDI/1-1", "_SIPSRTP=") in new stack
    -- Executing [1-dial@macro-trunkdial-failover-0.3:6] Set("DAHDI/1-1", "OUTDIALOPT=tTkKWwXx") in new stack
    -- Executing [1-dial@macro-trunkdial-failover-0.3:7] NoOp("DAHDI/1-1", "null for std") in new stack
    -- Executing [1-dial@macro-trunkdial-failover-0.3:8] GotoIf("DAHDI/1-1", "0?sys-dial,1)}") in new stack
    -- Executing [1-dial@macro-trunkdial-failover-0.3:9] GotoIf("DAHDI/1-1", "0?1-dial,10:1-dial,11") in new stack
    -- Goto (macro-trunkdial-failover-0.3,1-dial,11)
    -- Executing [1-dial@macro-trunkdial-failover-0.3:11] Set("DAHDI/1-1", "PJSIP_TRUNK=@trunk-TrunkToASTERISK-endpoint") in new stack
    -- Executing [1-dial@macro-trunkdial-failover-0.3:12] Set("DAHDI/1-1", "DIALSTRING=PJSIP/382@trunk-TrunkToASTERISK-endpoint") in new stack
    -- Executing [1-dial@macro-trunkdial-failover-0.3:13] Dial("DAHDI/1-1", "PJSIP/382@trunk-TrunkToASTERISK-endpoint,,tTkKWwXx") in new stack
[2015-12-24 12:22:26] WARNING[3090][C-00000000]: chan_pjsip.c:1873 chan_pjsip_request: SDP of 'PJSIP/trunk-TrunkToASTERISK-endpoint-00000000[0x3e8d6c]', [annexb 0], [annexa 0], [mode 0]
    -- Called PJSIP/382@trunk-TrunkToASTERISK-endpoint
[2015-12-24 12:22:26] WARNING[1707]: res_pjsip_sdp_rtp.c:1032 create_outgoing_sdp_stream: Unable to get rtp codec payload code for none
    -- PJSIP/trunk-TrunkToASTERISK-endpoint-00000000 is ringing
[2015-12-24 12:22:28] WARNING[1706]: res_pjsip_sdp_rtp.c:1032 create_outgoing_sdp_stream: Unable to get rtp codec payload code for none
    -- PJSIP/trunk-TrunkToASTERISK-endpoint-00000000 answered DAHDI/1-1
    -- Channel DAHDI/1-1 joined 'simple_bridge' basic-bridge <53bd78b3-ee64-46a7-92f1-f6283fcb75a8>
    -- Channel PJSIP/trunk-TrunkToASTERISK-endpoint-00000000 joined 'simple_bridge' basic-bridge <53bd78b3-ee64-46a7-92f1-f6283fcb75a8>
[2015-12-24 12:22:28] WARNING[3120][C-00000000]: channel.c:5073 ast_write: Codec mismatch on channel PJSIP/trunk-TrunkToASTERISK-endpoint-00000000 setting write format to slin from ulaw native formats (alaw)
       > 0x4368d0 -- Probation passed - setting RTP source address to 192.168.254.173:12230
       > 0x4368d0 -- Switching RTP source address to 192.168.254.188:12370
[2015-12-24 12:22:30] WARNING[1706]: res_pjsip_sdp_rtp.c:1032 create_outgoing_sdp_stream: Unable to get rtp codec payload code for none
    -- Channel PJSIP/trunk-TrunkToASTERISK-endpoint-00000000 left 'simple_bridge' basic-bridge <53bd78b3-ee64-46a7-92f1-f6283fcb75a8>
    -- Channel DAHDI/1-1 left 'simple_bridge' basic-bridge <53bd78b3-ee64-46a7-92f1-f6283fcb75a8>
  == Spawn extension (macro-trunkdial-failover-0.3, 1-dial, 13) exited non-zero on 'DAHDI/1-1' in macro 'trunkdial-failover-0.3'
  == Spawn extension (DLPN_DialPlan601, 382, 11) exited non-zero on 'DAHDI/1-1'
    -- Executing [h@DLPN_DialPlan601:1] NoOp("DAHDI/1-1", "no thing to do") in new stack
    -- Executing [h@DLPN_DialPlan601:2] Hangup("DAHDI/1-1", "") in new stack
  == Spawn extension (DLPN_DialPlan601, h, 2) exited non-zero on 'DAHDI/1-1'
[2015-12-24 12:22:30] WARNING[3090][C-00000000]: chan_dahdi.c:5152 dahdi_ec_disable: Disabled echo cancellation on channel 1
    -- Hanging up on 'DAHDI/1-1'
    -- Hungup 'DAHDI/1-1'
N824*CLI>

На Asterisk вывод лога консоли: 

ubuntu14-template*CLI>
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Executing [382@in_office:1] NoOp("SIP/ASTER-N824-00000000", "DATETIME= 24122015-12:22:26") in new stack
    -- Executing [382@in_office:2] NoOp("SIP/ASTER-N824-00000000", "CALLERID(all)= "601" <601>") in new stack
    -- Executing [382@in_office:3] NoOp("SIP/ASTER-N824-00000000", "CALLERID(dnid)= 382") in new stack
    -- Executing [382@in_office:4] NoOp("SIP/ASTER-N824-00000000", "CALLERID(name)= 601") in new stack
    -- Executing [382@in_office:5] NoOp("SIP/ASTER-N824-00000000", "CALLERID(num)= 601") in new stack
    -- Executing [382@in_office:6] NoOp("SIP/ASTER-N824-00000000", "CONTEXT= in_office") in new stack
    -- Executing [382@in_office:7] NoOp("SIP/ASTER-N824-00000000", "CHANNEL= SIP/ASTER-N824-00000000") in new stack
    -- Executing [382@in_office:8] NoOp("SIP/ASTER-N824-00000000", "EXTEN= 382") in new stack
    -- Executing [382@in_office:9] Dial("SIP/ASTER-N824-00000000", "SIP/382") in new stack
  == Using SIP VIDEO TOS bits 136
  == Using SIP VIDEO CoS mark 4
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Called SIP/382
    -- SIP/382-00000001 is ringing
    -- SIP/382-00000001 answered SIP/ASTER-N824-00000000
    -- Channel SIP/382-00000001 joined 'simple_bridge' basic-bridge <bf93f204-8d54-4e9d-9e3e-950a50d04887>
    -- Channel SIP/ASTER-N824-00000000 joined 'simple_bridge' basic-bridge <bf93f204-8d54-4e9d-9e3e-950a50d04887>
       > Bridge bf93f204-8d54-4e9d-9e3e-950a50d04887: switching from simple_bridge technology to native_rtp
       > Remotely bridged 'SIP/ASTER-N824-00000000' and 'SIP/382-00000001' - media will flow directly between them
       > Remotely bridged 'SIP/ASTER-N824-00000000' and 'SIP/382-00000001' - media will flow directly between them
       > 0x7f257c005890 -- Probation passed - setting RTP source address to 192.168.254.188:12370
       > 0x7f257c005890 -- Probation passed - setting RTP source address to 192.168.254.188:12370
    -- Channel SIP/382-00000001 left 'native_rtp' basic-bridge <bf93f204-8d54-4e9d-9e3e-950a50d04887>
    -- Channel SIP/ASTER-N824-00000000 left 'native_rtp' basic-bridge <bf93f204-8d54-4e9d-9e3e-950a50d04887>
  == Spawn extension (in_office, 382, 9) exited non-zero on 'SIP/ASTER-N824-00000000'
ubuntu14-template*CLI>

[править] Проверка маршрутизации звонков из Asterisk в Yeastar N824

На IP-телефоне SIP 382 зарегистрированном в Asterisk набираем номер 601 (аналоговый телефон 601 на Yeastar N824 порт 1 ).

Аналоговый телефон 601 включенный в первый порт Yeastar N824 должен зазвонить.

На Asterisk вывод лога: 

ubuntu14-template*CLI>
  == Using SIP VIDEO TOS bits 136
  == Using SIP VIDEO CoS mark 4
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Executing [601@in_office:1] Dial("SIP/382-00000004", "SIP/ASTER-N824/601") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Called SIP/ASTER-N824/601
    -- SIP/ASTER-N824-00000005 is ringing
    -- SIP/ASTER-N824-00000005 is ringing
    -- SIP/ASTER-N824-00000005 answered SIP/382-00000004
    -- Channel SIP/ASTER-N824-00000005 joined 'simple_bridge' basic-bridge <2d01f664-4efb-4dd1-a91a-696322a714db>
    -- Channel SIP/382-00000004 joined 'simple_bridge' basic-bridge <2d01f664-4efb-4dd1-a91a-696322a714db>
       > Bridge 2d01f664-4efb-4dd1-a91a-696322a714db: switching from simple_bridge technology to native_rtp
       > Remotely bridged 'SIP/382-00000004' and 'SIP/ASTER-N824-00000005' - media will flow directly between them
       > Remotely bridged 'SIP/382-00000004' and 'SIP/ASTER-N824-00000005' - media will flow directly between them
       > 0x7f25881d7420 -- Probation passed - setting RTP source address to 192.168.254.188:12374
       > 0x7f25881d7420 -- Probation passed - setting RTP source address to 192.168.254.188:12374
    -- Channel SIP/ASTER-N824-00000005 left 'native_rtp' basic-bridge <2d01f664-4efb-4dd1-a91a-696322a714db>
    -- Channel SIP/382-00000004 left 'native_rtp' basic-bridge <2d01f664-4efb-4dd1-a91a-696322a714db>
  == Spawn extension (in_office, 601, 1) exited non-zero on 'SIP/382-00000004'
ubuntu14-template*CLI>


На Yeastar N824 вывод лога: 

N824*CLI>
[2015-12-24 12:28:02] WARNING[4256]: chan_pjsip.c:2111 chan_pjsip_incoming_request: SDP of 'PJSIP/trunk-TrunkToASTERISK-endpoint-00000002[0x3d9e14]', [annexb 0], [annexa 0], [mode 0]
    -- Executing [601@Local_Default_CallingRules:1] Macro("PJSIP/trunk-TrunkToASTERISK-endpoint-00000002", "stdexten,601,DIALPARAM_OF_EXTEN601,{DIALOPTIONS}") in new stack
    -- Executing [s@macro-stdexten:1] Set("PJSIP/trunk-TrunkToASTERISK-endpoint-00000002", "IsFromOutside=0") in new stack
    -- Executing [s@macro-stdexten:2] Set("PJSIP/trunk-TrunkToASTERISK-endpoint-00000002", "_YS_CADENCE=1") in new stack
    -- Executing [s@macro-stdexten:3] GotoIf("PJSIP/trunk-TrunkToASTERISK-endpoint-00000002", "0?sys-dial,1)}") in new stack
    -- Executing [s@macro-stdexten:4] GotoIf("PJSIP/trunk-TrunkToASTERISK-endpoint-00000002", "0?Blacklist-Handle,s,1") in new stack
    -- Executing [s@macro-stdexten:5] Set("PJSIP/trunk-TrunkToASTERISK-endpoint-00000002", "DID_INBOUND_TRUNK_VARIABLE=") in new stack
    -- Executing [s@macro-stdexten:6] GotoIf("PJSIP/trunk-TrunkToASTERISK-endpoint-00000002", "0?set:noset") in new stack
    -- Goto (macro-stdexten,s,9)
    -- Executing [s@macro-stdexten:9] Macro("PJSIP/trunk-TrunkToASTERISK-endpoint-00000002", "realstexten,601,DAHDI/1,tTkKWwXx,") in new stack
    -- Executing [s@macro-realstexten:1] Set("PJSIP/trunk-TrunkToASTERISK-endpoint-00000002", "DYNAMIC_FEATURES=twstart") in new stack
    -- Executing [s@macro-realstexten:2] Set("PJSIP/trunk-TrunkToASTERISK-endpoint-00000002", "") in new stack
[2015-12-24 12:28:02] WARNING[4261][C-00000002]: pbx.c:11582 pbx_builtin_setvar: Set requires one variable name/value pair.
    -- Executing [s@macro-realstexten:3] GotoIf("PJSIP/trunk-TrunkToASTERISK-endpoint-00000002", "0?Blacklist-Handle,s,1") in new stack
    -- Executing [s@macro-realstexten:4] Set("PJSIP/trunk-TrunkToASTERISK-endpoint-00000002", "TIMEOUT(absolute)=6000") in new stack
    -- Channel will hangup at 2015-12-24 14:08:02.836 GMT-3.
    -- Executing [s@macro-realstexten:5] Set("PJSIP/trunk-TrunkToASTERISK-endpoint-00000002", "CKTSETTRANSFER=0") in new stack
    -- Executing [s@macro-realstexten:6] Set("PJSIP/trunk-TrunkToASTERISK-endpoint-00000002", "REALARG1=601") in new stack
    -- Executing [s@macro-realstexten:7] GotoIf("PJSIP/trunk-TrunkToASTERISK-endpoint-00000002", "0>0?follow-me,1") in new stack
    -- Executing [s@macro-realstexten:8] GotoIf("PJSIP/trunk-TrunkToASTERISK-endpoint-00000002", "0>0?vm-u,1") in new stack
    -- Executing [s@macro-realstexten:9] Set("PJSIP/trunk-TrunkToASTERISK-endpoint-00000002", "RINGTIME=30") in new stack
    -- Executing [s@macro-realstexten:10] CktStdCall("PJSIP/trunk-TrunkToASTERISK-endpoint-00000002", "srtpfor,DAHDI/1,novalue") in new stack
[2015-12-24 12:28:02] ERROR[4261][C-00000002]: pbx.c:4344 ast_func_read: Function SIPPEER not registered
    -- Executing [s@macro-realstexten:11] Set("PJSIP/trunk-TrunkToASTERISK-endpoint-00000002", "_SIPSRTP=") in new stack
    -- Executing [s@macro-realstexten:12] Set("PJSIP/trunk-TrunkToASTERISK-endpoint-00000002", "TYPE=1") in new stack
    -- Executing [s@macro-realstexten:13] Set("PJSIP/trunk-TrunkToASTERISK-endpoint-00000002", "DIALSTRING=DAHDI/1") in new stack
    -- Executing [s@macro-realstexten:14] Set("PJSIP/trunk-TrunkToASTERISK-endpoint-00000002", "DIALTIME=30") in new stack
    -- Executing [s@macro-realstexten:15] Dial("PJSIP/trunk-TrunkToASTERISK-endpoint-00000002", "DAHDI/1,30,tTkKWwXx") in new stack
    -- Called DAHDI/1
    -- DAHDI/1-1 is ringing
[2015-12-24 12:28:04] WARNING[4261][C-00000002]: chan_dahdi.c:5108 dahdi_ec_enable: Enabled echo cancellation on channel 1
    -- DAHDI/1-1 answered PJSIP/trunk-TrunkToASTERISK-endpoint-00000002
    -- Channel PJSIP/trunk-TrunkToASTERISK-endpoint-00000002 joined 'simple_bridge' basic-bridge <d7153ed8-1d96-4ddf-abaa-17d9401aa5a3>
    -- Channel DAHDI/1-1 joined 'simple_bridge' basic-bridge <d7153ed8-1d96-4ddf-abaa-17d9401aa5a3>
       > 0x4368d0 -- Probation passed - setting RTP source address to 192.168.254.173:19344
       > 0x4368d0 -- Probation passed - setting RTP source address to 192.168.254.173:19344
       > 0x4368d0 -- Switching RTP source address to 192.168.254.188:12374
[2015-12-24 12:28:04] NOTICE[4261][C-00000002]: channel.c:4140 __ast_read: Dropping incompatible voice frame on PJSIP/trunk-TrunkToASTERISK-endpoint-00000002 of format ulaw since our native format has changed to (alaw)
[2015-12-24 12:28:08] WARNING[4269][C-00000002]: chan_dahdi.c:5152 dahdi_ec_disable: Disabled echo cancellation on channel 1
    -- Channel DAHDI/1-1 left 'simple_bridge' basic-bridge <d7153ed8-1d96-4ddf-abaa-17d9401aa5a3>
    -- Channel PJSIP/trunk-TrunkToASTERISK-endpoint-00000002 left 'simple_bridge' basic-bridge <d7153ed8-1d96-4ddf-abaa-17d9401aa5a3>
  == Spawn extension (macro-realstexten, s, 15) exited non-zero on 'PJSIP/trunk-TrunkToASTERISK-endpoint-00000002' in macro 'realstexten'
    -- Hanging up on 'DAHDI/1-1'
    -- Hungup 'DAHDI/1-1'
  == Spawn extension (macro-stdexten, s, 9) exited non-zero on 'PJSIP/trunk-TrunkToASTERISK-endpoint-00000002' in macro 'stdexten'
  == Spawn extension (Local_Default_CallingRules, 601, 1) exited non-zero on 'PJSIP/trunk-TrunkToASTERISK-endpoint-00000002'
    -- Executing [h@Local_Default_CallingRules:1] NoOp("PJSIP/trunk-TrunkToASTERISK-endpoint-00000002", "no thing to do") in new stack
    -- Executing [h@Local_Default_CallingRules:2] Hangup("PJSIP/trunk-TrunkToASTERISK-endpoint-00000002", "") in new stack
  == Spawn extension (Local_Default_CallingRules, h, 2) exited non-zero on 'PJSIP/trunk-TrunkToASTERISK-endpoint-00000002'
N824*CLI>
Источник — «http://xgu.ru:81/wiki/%D0%9F%D0%BE%D0%B4%D0%BA%D0%BB%D1%8E%D1%87%D0%B5%D0%BD%D0%B8%D0%B5_%D0%90%D0%A2%D0%A1_Yeastar_N824_%D0%BA_Asterisk»