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My PBX Задержка перед переадресацией

Добрый день. Столкнулся со следующей проблемой. Была заведена аналоговая линия в MyPBX и для входящей и для исходящей связи, и были произведены некоторые настройки(какие неизвестно). И все работало все было хорошо. Появилась необходимость заменить исходящую линию на sip. Заменили, добавили транк сменили маршруты. И появились проблемы.

Допустим ситуация следующая:

Приходит звонок от клиента на аналоговую линию, допустим на секретаря. Ему клиент говорит "хочу поговорить с Петей", секретарь нажимает "*3502" (502 - номер Пети), или просто transfer 502. И тут начинаются чудеса. Раньше когда были и входящая и исходящая аналоговые переключение происходило мгновенно, то есть как нажал "*3502", так и 502 зазвонил. А теперь нажимаешь "*3502" - в трубке у секретаря тишина секунд 10-15, и только потом начинаются гудки и звонок у Пети. А если во время этой тишины положить трубку звонок просто срывается. И получается что теперь для того чтобы перевести звонок нужно ждать дополнительные 10 секунд. А это мягко говоря неудобно.

Вообще не очень понятно как переадресация связана со сменой исходящих маршрутов.

Была мысль что это связано с тем что теперь аналог конвертируется в sip и идет какая то нестыковка кодеков.

Есть идеи с чем это может быть связано и как это лечить?

Edited by faq4272

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А если набрать "*3502#" звонок сразу переводится ?

Нет, так же. Я думаю суть не в том как переводить. Так как можно переводить через кнопку transfer и результат не меняется.

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Не пробовали на железке снять dump вызова во время перевода?

В данном случае он многое бы открыл и показал.

Боюсь я не являюсь таким специалистом в телефонии. Я запускал -rvvvv с sip set debug peer 500. И мне мало что стало понятно. Возможно вы конечно правы, видимо просто я не вижу в чем проблема.

 

Я могу выложить логи сюда.

500 номер на который поступал звонок через аналоговую линию. 502 на который переводили.

 

   -- Called 500
MyPBX*CLI>
<--- SIP read from UDP:192.168.7.99:5172 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.7.50:5172;branch=z9hG4bK5d584847;rport
From: "Unknown" <sip:Unknown@192.168.7.50:5172>;tag=as38cd72df
To: <sip:500@192.168.7.99:5172>
Call-ID: 08a6cba9770c2fbe2f56a56c01a420f4@192.168.7.50
CSeq: 102 INVITE
User-Agent: Yealink SIP-T26P 6.60.14.15
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
MyPBX*CLI>
<--- SIP read from UDP:192.168.7.99:5172 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.7.50:5172;branch=z9hG4bK5d584847;rport
From: "Unknown" <sip:Unknown@192.168.7.50:5172>;tag=as38cd72df
To: <sip:500@192.168.7.99:5172>;tag=28620435
Call-ID: 08a6cba9770c2fbe2f56a56c01a420f4@192.168.7.50
CSeq: 102 INVITE
Contact: <sip:500@192.168.7.99:5172>
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
User-Agent: Yealink SIP-T26P 6.60.14.15
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
   -- SIP/500-00000d14 is ringing
MyPBX*CLI>
<--- SIP read from UDP:192.168.7.99:5172 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.7.50:5172;branch=z9hG4bK5d584847;rport
From: "Unknown" <sip:Unknown@192.168.7.50:5172>;tag=as38cd72df
To: <sip:500@192.168.7.99:5172>;tag=28620435
Call-ID: 08a6cba9770c2fbe2f56a56c01a420f4@192.168.7.50
CSeq: 102 INVITE
Contact: <sip:500@192.168.7.99:5172>
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
User-Agent: Yealink SIP-T26P 6.60.14.15
Content-Length: 196

v=0
o=- 23008 23008 IN IP4 192.168.7.99
s=SDP data
c=IN IP4 192.168.7.99
t=0 0
m=audio 11790 RTP/AVP 0 96
a=rtpmap:0 PCMU/8000
a=sendrecv
a=fmtp:96 0-15
a=rtpmap:96 telephone-event/8000

<------------->
--- (11 headers 10 lines) ---
Found RTP audio format 0
Found RTP audio format 96
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 96
Capabilities: us - 0x30e (gsm|ulaw|alaw|g729|speex), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.7.99:11790
list_route: hop: <sip:500@192.168.7.99:5172>
set_destination: Parsing <sip:500@192.168.7.99:5172> for address/port to send to
set_destination: set destination to 192.168.7.99, port 5172
Transmitting (no NAT) to 192.168.7.99:5172:
ACK sip:500@192.168.7.99:5172 SIP/2.0
Via: SIP/2.0/UDP 192.168.7.50:5172;branch=z9hG4bK025fc9c8;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.7.50:5172>;tag=as38cd72df
To: <sip:500@192.168.7.99:5172>;tag=28620435
Contact: <sip:Unknown@192.168.7.50:5172>
Call-ID: 08a6cba9770c2fbe2f56a56c01a420f4@192.168.7.50
CSeq: 102 ACK
User-Agent: MyPBX
Content-Length: 0


---
   -- SIP/500-00000d14 answered DAHDI/2-1
MyPBX*CLI>
<--- SIP read from UDP:192.168.7.99:5172 --->



<------------->
Reliably Transmitting (no NAT) to 192.168.7.99:5172:
OPTIONS sip:500@192.168.7.99:5172 SIP/2.0
Via: SIP/2.0/UDP 192.168.7.50:5172;branch=z9hG4bK0f1dccbd;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.7.50:5172>;tag=as01f62c23
To: <sip:500@192.168.7.99:5172>
Contact: <sip:Unknown@192.168.7.50:5172>
Call-ID: 155f35a17e7b120f4f35cb2e2b149b8a@192.168.7.50
CSeq: 102 OPTIONS
User-Agent: MyPBX
Date: Fri, 06 Nov 2015 08:35:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
MyPBX*CLI>
<--- SIP read from UDP:192.168.7.99:5172 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.7.50:5172;branch=z9hG4bK0f1dccbd;rport
From: "Unknown" <sip:Unknown@192.168.7.50:5172>;tag=as01f62c23
To: <sip:500@192.168.7.99:5172>;tag=1078175820
Call-ID: 155f35a17e7b120f4f35cb2e2b149b8a@192.168.7.50
CSeq: 102 OPTIONS
User-Agent: Yealink SIP-T26P 6.60.14.15
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '155f35a17e7b120f4f35cb2e2b149b8a@192.168.7.50' Method: OPTIONS
   -- Started music on hold, class 'default', on DAHDI/2-1
   -- <SIP/500-00000d14> Playing 'pbx-transfer.gsm' (language 'ru')
[2015-11-06 11:35:48] NOTICE[405]: utils.c:305 HostPoolUpdater: ======host:sip.comtube.com update dns======
   -- Executing [502@DLPN_DialPlan500:1] Macro("Local/502@DLPN_DialPlan500-d810;2", "stdexten,502,SIP/502") in new stack
   -- Executing [s@macro-stdexten:1] Set("Local/502@DLPN_DialPlan500-d810;2", "IsFromOutside=0") in new stack
   -- Executing [s@macro-stdexten:2] GotoIf("Local/502@DLPN_DialPlan500-d810;2", "0?sys-dial,1)}") in new stack
   -- Executing [s@macro-stdexten:3] GotoIf("Local/502@DLPN_DialPlan500-d810;2", "0?Blacklist-Handle,s,1") in new stack
   -- Executing [s@macro-stdexten:4] Macro("Local/502@DLPN_DialPlan500-d810;2", "realstexten,502,SIP/502,tTkKWwXx") in new stack
   -- Executing [s@macro-realstexten:1] Set("Local/502@DLPN_DialPlan500-d810;2", "DYNAMIC_FEATURES=twstart") in new stack
   -- Executing [s@macro-realstexten:2] Set("Local/502@DLPN_DialPlan500-d810;2", "") in new stack
[2015-11-06 11:35:50] WARNING[22808]: pbx.c:9219 pbx_builtin_setvar_multiple: MSet requires at least one variable name/value pair.
   -- Executing [s@macro-realstexten:3] GotoIf("Local/502@DLPN_DialPlan500-d810;2", "0?Blacklist-Handle,s,1") in new stack
   -- Executing [s@macro-realstexten:4] Set("Local/502@DLPN_DialPlan500-d810;2", "TIMEOUT(absolute)=0") in new stack
Channel hangup cancelled.
   -- Executing [s@macro-realstexten:5] Set("Local/502@DLPN_DialPlan500-d810;2", "CKTSETTRANSFER=0") in new stack
   -- Executing [s@macro-realstexten:6] Set("Local/502@DLPN_DialPlan500-d810;2", "REALARG1=502") in new stack
   -- Executing [s@macro-realstexten:7] GotoIf("Local/502@DLPN_DialPlan500-d810;2", "0>0?follow-me,1") in new stack
   -- Executing [s@macro-realstexten:8] GotoIf("Local/502@DLPN_DialPlan500-d810;2", "0>0?vm-u,1") in new stack
   -- Executing [s@macro-realstexten:9] Set("Local/502@DLPN_DialPlan500-d810;2", "RINGTIME=10") in new stack
   -- Executing [s@macro-realstexten:10] CktStdCall("Local/502@DLPN_DialPlan500-d810;2", "srtpfor,SIP/502,novalue") in new stack
   -- Executing [s@macro-realstexten:11] Set("Local/502@DLPN_DialPlan500-d810;2", "_SIPSRTP=0") in new stack
   -- Executing [s@macro-realstexten:12] Dial("Local/502@DLPN_DialPlan500-d810;2", "SIP/502,10,tTkKWwXx") in new stack
   -- ######## ringgroup params:(null)
 == Using SIP RTP TOS bits 184
 == Using SIP RTP CoS mark 5
   -- Called 502
set_destination: Parsing <sip:500@192.168.7.99:5172> for address/port to send to
set_destination: set destination to 192.168.7.99, port 5172
Reliably Transmitting (no NAT) to 192.168.7.99:5172:
NOTIFY sip:500@192.168.7.99:5172 SIP/2.0
Via: SIP/2.0/UDP 192.168.7.50:5172;branch=z9hG4bK2f5ee8a2;rport
Max-Forwards: 70
From: <sip:502@192.168.7.50:5172>;tag=as12fde11e
To: "VOSTOK-TVER" <sip:500@192.168.7.50:5172>;tag=83564305
Contact: <sip:502@192.168.7.50:5172>
Call-ID: 16541150@192.168.7.99
CSeq: 792 NOTIFY
User-Agent: MyPBX
Event: dialog
Content-Type: application/dialog-info+xml
Subscription-State: active
Content-Length: 494

<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="690" state="full" entity="sip:502@192.168.7.50:5172">
<dialog id="502" call-id="pickup-16541150@192.168.7.99" direction="recipient">
<remote>
<identity display="502">sip:502@192.168.7.50:5172</identity>
<target uri="sip:502@192.168.7.50:5172"/>
</remote>
<local>
<identity>sip:502@192.168.7.50:5172</identity>
<target uri="sip:502@192.168.7.50:5172"/>
</local>
<state>early</state>
</dialog>
</dialog-info>

---
 == Extension Changed 502[extensions-hintcontext] new state Ringing for Notify User 500
MyPBX*CLI>
<--- SIP read from UDP:192.168.7.99:5172 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.7.50:5172;branch=z9hG4bK2f5ee8a2;rport
From: <sip:502@192.168.7.50:5172>;tag=as12fde11e
To: "VOSTOK-TVER" <sip:500@192.168.7.50:5172>;tag=83564305
Call-ID: 16541150@192.168.7.99
CSeq: 792 NOTIFY
User-Agent: Yealink SIP-T26P 6.60.14.15
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
SIP Response message for INCOMING dialog NOTIFY arrived
   -- SIP/502-00000d15 is ringing
   -- Local/502@DLPN_DialPlan500-d810;1 is ringing
   -- SIP/502-00000d15 answered Local/502@DLPN_DialPlan500-d810;2
set_destination: Parsing <sip:500@192.168.7.99:5172> for address/port to send to
set_destination: set destination to 192.168.7.99, port 5172
Reliably Transmitting (no NAT) to 192.168.7.99:5172:
NOTIFY sip:500@192.168.7.99:5172 SIP/2.0
Via: SIP/2.0/UDP 192.168.7.50:5172;branch=z9hG4bK2474d15c;rport
Max-Forwards: 70
From: <sip:502@192.168.7.50:5172>;tag=as12fde11e
To: "VOSTOK-TVER" <sip:500@192.168.7.50:5172>;tag=83564305
Contact: <sip:502@192.168.7.50:5172>
Call-ID: 16541150@192.168.7.99
CSeq: 793 NOTIFY
User-Agent: MyPBX
Event: dialog
Content-Type: application/dialog-info+xml
Subscription-State: active
Content-Length: 209

<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="691" state="full" entity="sip:502@192.168.7.50:5172">
<dialog id="502">
<state>confirmed</state>
</dialog>
</dialog-info>

---
 == Extension Changed 502[extensions-hintcontext] new state InUse for Notify User 500
MyPBX*CLI>
<--- SIP read from UDP:192.168.7.99:5172 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.7.50:5172;branch=z9hG4bK2474d15c;rport
From: <sip:502@192.168.7.50:5172>;tag=as12fde11e
To: "VOSTOK-TVER" <sip:500@192.168.7.50:5172>;tag=83564305
Call-ID: 16541150@192.168.7.99
CSeq: 793 NOTIFY
User-Agent: Yealink SIP-T26P 6.60.14.15
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
SIP Response message for INCOMING dialog NOTIFY arrived
MyPBX*CLI>
<--- SIP read from UDP:192.168.7.99:5172 --->
BYE sip:Unknown@192.168.7.50:5172 SIP/2.0
Via: SIP/2.0/UDP 192.168.7.99:5172;branch=z9hG4bK1896065317
From: <sip:500@192.168.7.99:5172>;tag=28620435
To: "Unknown" <sip:Unknown@192.168.7.50:5172>;tag=as38cd72df
Call-ID: 08a6cba9770c2fbe2f56a56c01a420f4@192.168.7.50
CSeq: 103 BYE
Contact: <sip:500@192.168.7.99:5172>
Max-Forwards: 70
User-Agent: Yealink SIP-T26P 6.60.14.15
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
Sending to 192.168.7.99 : 5172 (no NAT)

<--- Transmitting (no NAT) to 192.168.7.99:5172 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.7.99:5172;branch=z9hG4bK1896065317;received=192.168.7.99
From: <sip:500@192.168.7.99:5172>;tag=28620435
To: "Unknown" <sip:Unknown@192.168.7.50:5172>;tag=as38cd72df
Call-ID: 08a6cba9770c2fbe2f56a56c01a420f4@192.168.7.50
CSeq: 103 BYE
Server: MyPBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
   -- Executing [h@DLPN_DialPlan500:1] NoOp("SIP/500-00000d14", "no thing to do") in new stack
   -- Executing [h@DLPN_DialPlan500:2] Hangup("SIP/500-00000d14", "") in new stack
   -- Stopped music on hold on DAHDI/2-1
[2015-11-06 11:35:54] NOTICE[22807]: channel.c:4678 ast_do_masquerade: atxfer not play tone
   -- <Local/502@DLPN_DialPlan500-d810;1> Playing 'beep.gsm' (language 'ru')
 == Spawn extension (macro-realstexten, s, 12) exited non-zero on 'Transfered/DAHDI/2-1<ZOMBIE>' in macro 'realstexten'
 == Spawn extension (macro-stdexten-fromoutside, s, 2) exited non-zero on 'Transfered/DAHDI/2-1<ZOMBIE>' in macro 'stdexten-fromoutside'
 == Spawn extension (from-outside, 500, 1) exited non-zero on 'Transfered/DAHDI/2-1<ZOMBIE>'
Really destroying SIP dialog '08a6cba9770c2fbe2f56a56c01a420f4@192.168.7.50' Method: BYE
[2015-11-06 11:35:56] NOTICE[405]: utils.c:305 HostPoolUpdater: ======host:voip.anders.ru update dns======
MyPBX*CLI>
<--- SIP read from UDP:192.168.7.99:5172 --->



<------------->
[2015-11-06 11:36:01] NOTICE[519]: chan_sip.c:12011 sip_reregister:    -- Re-registration for  6215@82.138.35.55@82.138.35.55
[2015-11-06 11:36:01] NOTICE[519]: chan_sip.c:19233 handle_response_register: Outbound Registration: Expiry for 82.138.35.55 is 120 sec (Scheduling reregistration in 105 s)
[2015-11-06 11:36:04] NOTICE[405]: utils.c:305 HostPoolUpdater: ======host:MyPBX update dns======
MyPBX*CLI>
Disconnected from Asterisk server
Executing last minute cleanups

Edited by faq4272

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