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Asterisk 12.4.0 команда channel redirect разрывает оба соединения, звонок не переводится

0

Добрый день! Пытаюсь командой из CLI сделать blind transfer звонка. Звонок поступает в очередь и падает на агента, соединение установлено, разговор идёт:

asterisk*CLI> core show channels 
Channel              Location             State   Application(Data)             
SIP/2001-0000000a    (None)               Up      AppQueue((Outgoing Line))     
SIP/2003-00000009    (None)               Up      Queue(CRM,tc)                 
2 active channels
1 active call
5 calls processed

После этого пытаюсь сделать:
asterisk*CLI> channel redirect SIP/2003-00000009 2002
Channel 'SIP/2003-00000009' successfully redirected to 2002
    -- Channel SIP/2003-00000009 left 'simple_bridge' basic-bridge <8dcb78a3-f502-46c2-abfe-254a33bcc71e>
    -- Channel SIP/2001-0000000a left 'simple_bridge' basic-bridge <8dcb78a3-f502-46c2-abfe-254a33bcc71e>
    -- Auto fallthrough, channel 'SIP/2003-00000009' status is 'UNKNOWN'

SIP конфиг:

[2001]
type=friend
regexten=2001
secret=2001
callerid="2001" <2001>
host=dynamic
directmedia=no
insecure=invite
disallow=all
allow=ulaw
registertrying=yes
context=internal
qualify=100
qualifyfreq=10
call-limit=10
canreinvite=yes

SIP debug:

asterisk*CLI> channel redirect SIP/2003-0000000b 2002
Channel 'SIP/2003-0000000b' successfully redirected to 2002
    -- Channel SIP/2003-0000000b left 'simple_bridge' basic-bridge <0b988562-9eed-4df3-9fe6-34bdfd5a5a7e>
    -- Auto fallthrough, channel 'SIP/2003-0000000b' status is 'UNKNOWN'
Scheduling destruction of SIP dialog '4c35f4f9-6215-e411-9e6d-f0def11a319a@localhost.localdomain' in 6400 ms (Method: ACK)
set_destination: Parsing <sip:2003@10.0.8.171:5060> for address/port to send to
set_destination: set destination to 10.0.8.171:5060
    -- Channel SIP/2001-0000000c left 'simple_bridge' basic-bridge <0b988562-9eed-4df3-9fe6-34bdfd5a5a7e>
Reliably Transmitting (no NAT) to 10.0.8.171:5060:
BYE sip:2003@10.0.8.171:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.15.40:5060;branch=z9hG4bK486ae89d;rport
Max-Forwards: 70
From: <sip:9150@10.0.15.40>;tag=as228803e1
To: "Andrey Lyarskiy" <sip:2003@10.0.15.40>;tag=4a32f4f9-6215-e411-9e6d-f0def11a319a
Call-ID: 4c35f4f9-6215-e411-9e6d-f0def11a319a@localhost.localdomain
CSeq: 102 BYE
User-Agent: Asterisk PBX 12.4.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
Scheduling destruction of SIP dialog '71bd9e5b2ab9af95239b66da72131b61@10.0.15.40:5060' in 6400 ms (Method: INVITE)
set_destination: Parsing <sip:2001@10.0.17.128:5060> for address/port to send to
set_destination: set destination to 10.0.17.128:5060
Reliably Transmitting (no NAT) to 10.0.17.128:5060:
BYE sip:2001@10.0.17.128:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.15.40:5060;branch=z9hG4bK50d6b5af
Max-Forwards: 70
From: "2003" <sip:2003@10.0.15.40>;tag=as49681c37
To: <sip:2001@10.0.17.128:5060>;tag=2784575879
Call-ID: 71bd9e5b2ab9af95239b66da72131b61@10.0.15.40:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 12.4.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---

<--- SIP read from UDP:10.0.8.171:5060 --->
SIP/2.0 200 OK
CSeq: 102 BYE
Via: SIP/2.0/UDP 10.0.15.40:5060;branch=z9hG4bK486ae89d;rport
From: <sip:9150@10.0.15.40>;tag=as228803e1
Call-ID: 4c35f4f9-6215-e411-9e6d-f0def11a319a@localhost.localdomain
To: "Andrey Lyarskiy" <sip:2003@10.0.15.40>;tag=4a32f4f9-6215-e411-9e6d-f0def11a319a
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '4c35f4f9-6215-e411-9e6d-f0def11a319a@localhost.localdomain' Method: ACK

<--- SIP read from UDP:10.0.17.128:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.15.40:5060;branch=z9hG4bK50d6b5af
From: "2003" <sip:2003@10.0.15.40>;tag=as49681c37
To: <sip:2001@10.0.17.128:5060>;tag=2784575879
Call-ID: 71bd9e5b2ab9af95239b66da72131b61@10.0.15.40:5060
CSeq: 103 BYE
Contact: <sip:2001@10.0.17.128:5060>
Supported: replaces
User-Agent: C610A IP/42.076.00.000.000
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '71bd9e5b2ab9af95239b66da72131b61@10.0.15.40:5060' Method: INVITE
Reliably Transmitting (no NAT) to 10.0.17.129:5060:
OPTIONS sip:2002@10.0.17.129:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.15.40:5060;branch=z9hG4bK0c66b3f1
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.0.15.40>;tag=as6a00bc56
To: <sip:2002@10.0.17.129:5060>
Contact: <sip:asterisk@10.0.15.40:5060>
Call-ID: 087025ab1af750f240e0e5247106336e@10.0.15.40:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 12.4.0
Date: Tue, 29 Jul 2014 07:52:56 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:10.0.17.129:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.15.40:5060;branch=z9hG4bK0c66b3f1
From: "asterisk" <sip:asterisk@10.0.15.40>;tag=as6a00bc56
To: <sip:2002@10.0.17.129:5060>;tag=ar7a11cb47
Call-ID: 087025ab1af750f240e0e5247106336e@10.0.15.40:5060
CSeq: 102 OPTIONS
Supported: replaces
User-Agent: C610A IP/42.076.00.000.000
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Accept: application/sdp,application/dtmf-relay,application/simple-message-summary,message/sipfrag
Accept-Encoding: identity
Accept-Language: en
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Really destroying SIP dialog '087025ab1af750f240e0e5247106336e@10.0.15.40:5060' Method: OPTIONS
asterisk*CLI> sip set debug off 
SIP Debugging Disabled

Подскажите, что я делаю не так?

удалить закрыть спам изменить тег редактировать

спросил 2014-07-29 12:02:47 +0400

oxumorron Gravatar oxumorron
29 3 5

Comments

добро пажаловать в мир багов pjsip. наиболее вероятно запрещен reinvite

meral ( 2014-07-29 23:09:27 +0400 )редактировать

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Задан: 2014-07-29 12:02:47 +0400

Просмотрен: 806 раз

Обновлен: Jul 29 '14

Проект компании "АТС Дизайн"
Asterisk® и Digium® являются зарегистрированными торговыми марками компании Digium, Inc., США.
IP АТС Asterisk распространяется под лицензией GNU GPL.