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продает голос через 10 секунд разговора

Новичком считается только что прочитавший «Астериск - будущее телефонии»
http://asterisk.ru/knowledgebase/books
и пытающийся сделать большее

Модераторы: april22, Zavr2008

Ответить
ded
Сообщения: 15626
Зарегистрирован: 26 авг 2010, 19:00

Re: продает голос через 10 секунд разговора

Сообщение ded »

Vlad1983 писал(а):снимать сигналку
то есть
sip set debug on
и поймать разговор, при котором пропадает голос через 10 сек. Смотреть в дебаге - что происходит на этой 10-секунде?
Варианты:
- ничего не происходит
- пропадает RTP сам по себе :(
- пробегает какое-то SIP сообщение, вот тогда его тут и обубликовать.
Попробуем разобраться.
Вернуться к началу
telef
Сообщения: 70
Зарегистрирован: 29 июн 2011, 01:52

Re: продает голос через 10 секунд разговора

Сообщение telef »

Запустил debug on
без звонков вижу такую штуку
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:
---
Really destroying SIP dialog '000e408a2ea90c886d86fe1c2b30cd9c@77.241.34.150' Method: OPTIONS
Retransmitting #4 (no NAT) to 192.168.5.92:5060:
OPTIONS sip:192.168.5.92 SIP/2.0
Via: SIP/2.0/UDP 77.241.34.150:5060;branch=z9hG4bK75e71945;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@77.241.34.150>;tag=as6d5ad6d4
To: <sip:192.168.5.92>
Contact: <sip:asterisk@77.241.34.150>
Call-ID: 18a63aae235524102047cc9078b4a29f@77.241.34.150
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.9-2ubuntu2
Date: Sat, 17 Nov 2012 12:27:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
Really destroying SIP dialog '18a63aae235524102047cc9078b4a29f@77.241.34.150' Method: OPTIONS
Retransmitting #4 (no NAT) to 192.168.5.89:5060:
OPTIONS sip:192.168.5.89 SIP/2.0
Via: SIP/2.0/UDP 77.241.34.150:5060;branch=z9hG4bK19a4d9c0;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@77.241.34.150>;tag=as3089b318
To: <sip:192.168.5.89>
Contact: <sip:asterisk@77.241.34.150>
Call-ID: 67fbe14b03e82ce4287e27a35b4e8572@77.241.34.150
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.9-2ubuntu2
Date: Sat, 17 Nov 2012 12:27:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
Really destroying SIP dialog '67fbe14b03e82ce4287e27a35b4e8572@77.241.34.150' Method: OPTIONS
Retransmitting #4 (no NAT) to 192.168.5.89:5060:
OPTIONS sip:192.168.5.89 SIP/2.0
Via: SIP/2.0/UDP 77.241.34.150:5060;branch=z9hG4bK1e0c5ad7;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@77.241.34.150>;tag=as1e133f8c
To: <sip:192.168.5.89>
Contact: <sip:asterisk@77.241.34.150>
Call-ID: 3fbaf0db767fac281ae1c0b15b735176@77.241.34.150
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.9-2ubuntu2
Date: Sat, 17 Nov 2012 12:27:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
Really destroying SIP dialog '3fbaf0db767fac281ae1c0b15b735176@77.241.34.150' Method: OPTIONS
Retransmitting #4 (no NAT) to 192.168.5.89:5060:
OPTIONS sip:192.168.5.89 SIP/2.0
Via: SIP/2.0/UDP 77.241.34.150:5060;branch=z9hG4bK77d60382;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@77.241.34.150>;tag=as562d0859
To: <sip:192.168.5.89>
Contact: <sip:asterisk@77.241.34.150>
Call-ID: 763122c077faf7c83134a68513ac8aaf@77.241.34.150
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.9-2ubuntu2
Date: Sat, 17 Nov 2012 12:27:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
Really destroying SIP dialog '763122c077faf7c83134a68513ac8aaf@77.241.34.150' Method: OPTIONS
Retransmitting #4 (no NAT) to 192.168.5.92:5061:
OPTIONS sip:192.168.5.92 SIP/2.0
Via: SIP/2.0/UDP 77.241.34.150:5060;branch=z9hG4bK3484edd4;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@77.241.34.150>;tag=as197df04d
To: <sip:192.168.5.92>
Contact: <sip:asterisk@77.241.34.150>
Call-ID: 6350ad9122bff8cd6a3a8ccc6075459e@77.241.34.150
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.9-2ubuntu2
Date: Sat, 17 Nov 2012 12:27:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
Really destroying SIP dialog '6350ad9122bff8cd6a3a8ccc6075459e@77.241.34.150' Method: OPTIONS
Retransmitting #4 (no NAT) to 192.168.5.97:5060:
OPTIONS sip:192.168.5.97 SIP/2.0
Via: SIP/2.0/UDP 77.241.34.150:5060;branch=z9hG4bK1e046bb4;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@77.241.34.150>;tag=as77ba894b
To: <sip:192.168.5.97>
Contact: <sip:asterisk@77.241.34.150>
Call-ID: 7a7c0b36148a8d3161914a0633eeb1de@77.241.34.150
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.9-2ubuntu2
Date: Sat, 17 Nov 2012 12:27:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
Really destroying SIP dialog '7a7c0b36148a8d3161914a0633eeb1de@77.241.34.150' Method: OPTIONS
Retransmitting #4 (no NAT) to 192.168.5.82:5060:
OPTIONS sip:192.168.5.82 SIP/2.0
Via: SIP/2.0/UDP 77.241.34.150:5060;branch=z9hG4bK5fd13adc;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@77.241.34.150>;tag=as53441162
To: <sip:192.168.5.82>
Contact: <sip:asterisk@77.241.34.150>
Call-ID: 3e81993368c475767a5a316c5a5c5f79@77.241.34.150
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.9-2ubuntu2
Date: Sat, 17 Nov 2012 12:27:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
Really destroying SIP dialog '3e81993368c475767a5a316c5a5c5f79@77.241.34.150' Method: OPTIONS
Reliably Transmitting (no NAT) to 192.168.5.91:5061:
OPTIONS sip:192.168.5.91 SIP/2.0
Via: SIP/2.0/UDP 77.241.34.150:5060;branch=z9hG4bK13416119;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@77.241.34.150>;tag=as07fd075e
To: <sip:192.168.5.91>
Contact: <sip:asterisk@77.241.34.150>
Call-ID: 7d4c74243c2b7bc82a9066f2012f07a2@77.241.34.150
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.9-2ubuntu2
Date: Sat, 17 Nov 2012 12:27:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
при обрыве звонка появились такие строчки
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:
---
Really destroying SIP dialog '1100a6e8792875c13eb19e3026f54aae@77.241.34.150' Method: OPTIONS
Retransmitting #4 (no NAT) to 192.168.5.97:5061:
OPTIONS sip:192.168.5.97 SIP/2.0
Via: SIP/2.0/UDP 77.241.34.150:5060;branch=z9hG4bK7f8da7d9;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@77.241.34.150>;tag=as6e4d3c45
To: <sip:192.168.5.97>
Contact: <sip:asterisk@77.241.34.150>
Call-ID: 1aad126f7bb8065c67e460126e59053c@77.241.34.150
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.9-2ubuntu2
Date: Sat, 17 Nov 2012 12:29:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
Really destroying SIP dialog '1aad126f7bb8065c67e460126e59053c@77.241.34.150' Method: OPTIONS
Retransmitting #4 (no NAT) to 192.168.5.92:5060:
OPTIONS sip:192.168.5.92 SIP/2.0
Via: SIP/2.0/UDP 77.241.34.150:5060;branch=z9hG4bK71809613;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@77.241.34.150>;tag=as63f3653e
To: <sip:192.168.5.92>
Contact: <sip:asterisk@77.241.34.150>
Call-ID: 0f0383351996d7ec5462d0e05c872739@77.241.34.150
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.9-2ubuntu2
Date: Sat, 17 Nov 2012 12:29:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
Really destroying SIP dialog '0f0383351996d7ec5462d0e05c872739@77.241.34.150' Method: OPTIONS
Retransmitting #4 (no NAT) to 192.168.5.89:5060:
OPTIONS sip:192.168.5.89 SIP/2.0
Via: SIP/2.0/UDP 77.241.34.150:5060;branch=z9hG4bK026a7063;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@77.241.34.150>;tag=as61eaf332
To: <sip:192.168.5.89>
Contact: <sip:asterisk@77.241.34.150>
Call-ID: 2fafc0ae4d46966f75194e321d84fe4a@77.241.34.150
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.9-2ubuntu2
Date: Sat, 17 Nov 2012 12:29:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
Really destroying SIP dialog '2fafc0ae4d46966f75194e321d84fe4a@77.241.34.150' Method: OPTIONS
Retransmitting #4 (no NAT) to 192.168.5.89:5060:
OPTIONS sip:192.168.5.89 SIP/2.0
Via: SIP/2.0/UDP 77.241.34.150:5060;branch=z9hG4bK7296d81d;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@77.241.34.150>;tag=as162c7b4f
To: <sip:192.168.5.89>
Contact: <sip:asterisk@77.241.34.150>
Call-ID: 505a4f2e7cdfda26719c93280018e713@77.241.34.150
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.9-2ubuntu2
Date: Sat, 17 Nov 2012 12:29:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
Really destroying SIP dialog '505a4f2e7cdfda26719c93280018e713@77.241.34.150' Method: OPTIONS
Retransmitting #4 (no NAT) to 192.168.5.89:5060:
OPTIONS sip:192.168.5.89 SIP/2.0
Via: SIP/2.0/UDP 77.241.34.150:5060;branch=z9hG4bK023a1afa;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@77.241.34.150>;tag=as6cba8128
To: <sip:192.168.5.89>
Contact: <sip:asterisk@77.241.34.150>
Call-ID: 2a2a774d592dbacb7e73084037cec5bb@77.241.34.150
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.9-2ubuntu2
Date: Sat, 17 Nov 2012 12:29:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
Really destroying SIP dialog '2a2a774d592dbacb7e73084037cec5bb@77.241.34.150' Method: OPTIONS
Retransmitting #4 (no NAT) to 192.168.5.92:5061:
OPTIONS sip:192.168.5.92 SIP/2.0
Via: SIP/2.0/UDP 77.241.34.150:5060;branch=z9hG4bK58258f9e;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@77.241.34.150>;tag=as273e1ef3
To: <sip:192.168.5.92>
Contact: <sip:asterisk@77.241.34.150>
Call-ID: 756f0eac3510e0497e7ecafa6a9e0ff3@77.241.34.150
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.9-2ubuntu2
Date: Sat, 17 Nov 2012 12:29:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
Really destroying SIP dialog '756f0eac3510e0497e7ecafa6a9e0ff3@77.241.34.150' Method: OPTIONS
Retransmitting #4 (no NAT) to 192.168.5.97:5060:
OPTIONS sip:192.168.5.97 SIP/2.0
Via: SIP/2.0/UDP 77.241.34.150:5060;branch=z9hG4bK75db23c8;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@77.241.34.150>;tag=as78ae484a
To: <sip:192.168.5.97>
Contact: <sip:asterisk@77.241.34.150>
Call-ID: 2cf8ca7f1d9d36f679cfe14a6907ea68@77.241.34.150
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.9-2ubuntu2
Date: Sat, 17 Nov 2012 12:29:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
Really destroying SIP dialog '2cf8ca7f1d9d36f679cfe14a6907ea68@77.241.34.150' Method: OPTIONS
Retransmitting #4 (no NAT) to 192.168.5.82:5060:
OPTIONS sip:192.168.5.82 SIP/2.0
Via: SIP/2.0/UDP 77.241.34.150:5060;branch=z9hG4bK4d94c029;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@77.241.34.150>;tag=as038d684f
To: <sip:192.168.5.82>
Contact: <sip:asterisk@77.241.34.150>
Call-ID: 5f8ce765055dac943a6b73b21516608d@77.241.34.150
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.9-2ubuntu2
Date: Sat, 17 Nov 2012 12:29:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
Really destroying SIP dialog '5f8ce765055dac943a6b73b21516608d@77.241.34.150' Method: OPTIONS
Really destroying SIP dialog '29c338734040b6011119b4dd0cf9e010@sipnet.ru' Method: REGISTER
[Nov 17 15:29:26] WARNING[1013]: chan_sip.c:3785 retrans_pkt: Maximum retries exceeded on transmission 001201f5-d2f4000b-3248ede5-0a71dec8@192.168.5.85 for seqno 102 (Critical Response) -- See doc/sip-retransmit.txt.
[Nov 17 15:29:26] WARNING[1013]: chan_sip.c:3812 retrans_pkt: Hanging up call 001201f5-d2f4000b-3248ede5-0a71dec8@192.168.5.85 - no reply to our critical packet (see doc/sip-retransmit.txt).
Scheduling destruction of SIP dialog '7d5f7c7075eb36656ac298e73d7ebb34@sipnet.ru' in 32000 ms (Method: INVITE)
set_destination: Parsing <sip:212.53.40.40:5060;lr> for address/port to send to
set_destination: set destination to 212.53.40.40, port 5060
Reliably Transmitting (NAT) to 212.53.40.40:5060:
BYE sip:proc-137948@212.53.35.244 SIP/2.0
Via: SIP/2.0/UDP 77.241.34.150:5060;branch=z9hG4bK7905269d;rport
Route: <sip:212.53.40.40:5060;lr>,<sip:192.168.40.79:5060;lr>,<sip:2856104-192.168.40.79.dialog.cgatepro;lr>,<sip:212.53.35.244:5060;lr>
Max-Forwards: 70
From: "Chief Director" <sip:bogdan_ast@sipnet.ru>;tag=as3815d691
To: <sip:+78127150705@sipnet.ru>;tag=a2c35090-137948
Call-ID: 7d5f7c7075eb36656ac298e73d7ebb34@sipnet.ru
CSeq: 104 BYE
User-Agent: Asterisk PBX 1.6.2.9-2ubuntu2
Authorization: Digest username="bogdan_ast", realm="etc.tario.ru", algorithm=MD5, uri="sip:proc-137948@212.53.35.244", nonce="95D4CE55193B0BFB9C48", response="fc6fc8696495a2ea20cfdc7d34d2e268", opaque="opaq", qop=auth, cnonce="3dffcc01", nc=00000002
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---

<--- SIP read from UDP:212.53.40.40:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 77.241.34.150:5060;rport=5060;branch=z9hG4bK7905269d
From: <sip:bogdan_ast@sipnet.ru>;tag=as3815d691
To: <sip:+78127150705@sipnet.ru>;tag=a2c35090-137948
Call-ID: 7d5f7c7075eb36656ac298e73d7ebb34@sipnet.ru
CSeq: 104 BYE
Allow: INVITE,ACK,BYE,CANCEL,INFO,OPTIONS
Server: TarioSoftswitch/3.2.12
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '7d5f7c7075eb36656ac298e73d7ebb34@sipnet.ru' Method: INVITE
Really destroying SIP dialog '001201f5-d2f4000b-3248ede5-0a71dec8@192.168.5.85' Method: INVITE
Reliably Transmitting (no NAT) to 192.168.5.91:5061:
OPTIONS sip:192.168.5.91 SIP/2.0
Via: SIP/2.0/UDP 77.241.34.150:5060;branch=z9hG4bK53d0e085;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@77.241.34.150>;tag=as060725a4
To: <sip:192.168.5.91>
Contact: <sip:asterisk@77.241.34.150>
Call-ID: 46cef08148f2add212732e5a363ab0b0@77.241.34.150
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.9-2ubuntu2
Date: Sat, 17 Nov 2012 12:29:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
Reliably Transmitting (no NAT) to 192.168.5.90:5061:
OPTIONS sip:192.168.5.90 SIP/2.0
Via: SIP/2.0/UDP 77.241.34.150:5060;branch=z9hG4bK2958f520;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@77.241.34.150>;tag=as143994c6
To: <sip:192.168.5.90>
Contact: <sip:asterisk@77.241.34.150>
Call-ID: 295abaa042b028dd21d5665b55041e60@77.241.34.150
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.9-2ubuntu2
Date: Sat, 17 Nov 2012 12:29:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
Вернуться к началу
ded
Сообщения: 15626
Зарегистрирован: 26 авг 2010, 19:00

Re: продает голос через 10 секунд разговора

Сообщение ded »

Как-то тоскливо у вас сконфигурировано всё, не по понятиям.
Астериск стоит на реальном ИП 77.241.34.150, и посылает пакеты OPTIONS на телефоны на внутреннем ИП, которые как бы не за НАТ относительно Астериска?
(no NAT) to 192.168.5.92:5060
(no NAT) to 192.168.5.89:5060
(no NAT) to 192.168.5.97:5061

Код: Выделить всё

OPTIONS sip:192.168.5.97 SIP/2.0
Via: SIP/2.0/UDP 77.241.34.150:5060;branch=z9hG4bK7f8da7d9;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@77.241.34.150>;tag=as6e4d3c45
To: <sip:192.168.5.97>
Как это вообще работает у Вас?
Последний раз редактировалось ded 17 ноя 2012, 17:16, всего редактировалось 1 раз.
Вернуться к началу
telef
Сообщения: 70
Зарегистрирован: 29 июн 2011, 01:52

Re: продает голос через 10 секунд разговора

Сообщение telef »

телефон не за NAt
Вернуться к началу
ded
Сообщения: 15626
Зарегистрирован: 26 авг 2010, 19:00

Re: продает голос через 10 секунд разговора

Сообщение ded »

Или читайте книжки, или - платный суппорт.
У Вас бардак реальный.
Вернуться к началу
telef
Сообщения: 70
Зарегистрирован: 29 июн 2011, 01:52

Re: продает голос через 10 секунд разговора

Сообщение telef »

исправил параметр localnet в extension.conf
все заработало!

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