Помогите настроить Asterisk для sipnet.ru !!!
Странно себя ведёт Asteriskwin32 два дня мучаюсь ну просто ни как не настроить!
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Откуда: Saint-Petersburg
Сообщений: 3
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Помогите настроить Asterisk для sipnet.ru !!!
Добрый день! Вы моя последняя надежда в решеннии моей проблемы!
А проблема следующая:
Началось всё с того что мне надо было настроить сервер да так что можно было управлять не которыми сервисами с телефона(например звонишь на какой то номер после чего слышишь например hello world, набираешь 666 необходимый сервис останавливаеться, набираешь 777 сервис запускаеться, набираешь 111 перезагрузка ну т.д.)
Я скрипты написал всё работает осталось только вот найти софт который их мог бы запускать при звонке.
Купили мы номер, я выбрал asterisk(т.к. больше ни чего в голову не приходило) но тут проблема у меня winows server 2003x64 sp2!
Думаю ладно попробую настроить asteriskwin32, поставил конфиги и настойки всё те же как и на обычном!
Создал диалплан вот такой:
[general]
static=yes
writeprotect=yes
autofallthrough=yes
clearglobalvars=no
priorityjumping=yes
;[globals]
[incoming]
exten => s,1,verbose(1|Unrouted call handler)
exten => s,n,Answer
exten => s,n,Playback(hello-world)
;exten => s,n,Background(hello-world)
exten => s,n,WaitExten()
exten => s,n,Hangup
exten => 0029273190,1,Answer
exten => 0029273190,n,Playback(hello-world)
exten => 0029273190,n,WaitExten(20)
exten => 0029273190,n,Hangup
exten => 6,1,Answer
exten => 6,n,System('C:/shutdown.bat')
exten => 6,n,Playback(vm-goodbye)
exten => 6,n,Hangup
exten => 7,1,Answer
exten => 7,n,System('C:/restart.bat')
exten => 7,n,Playback(vm-goodbye)
exten => 7,n,Hangup
exten => 1,1,Answer
exten => 1,n,System('C:/test.bat')
exten => 1,n,Playback(vm-goodbye)
exten => 1,n,Hangup
exten => i,1,playback(pbx-invalid)
exten => i,n,HangUp
Далее настроил sip.conf вот так:
[general]
context = incoming ; Default context for incoming calls
bindport = 5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr = 0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
srvlookup = no ; Enable DNS SRV lookups on outbound calls
canreinvite = nonat
externrefresh = 60
nat = no
externip = мой внешний ip
register => 0029273190:пароль@sipnet.ru/0029273190
[authentication]
[sipnet]
type = peer
username = 0029273190
secret = password
host = sipnet.ru
nat = no
fromuser = 0029273190
fromdomain = sipnet.ru
dtmfmode = info
insecure = invite
context = incoming
contact = 0029273190
disallow = all
allow = ulaw
allow = g729
allow = alaw
canreinvite = nonat
А потом началось сначало всё как мне показалось работате но потом по не понятным причинам дозваниваешься то сразу короткие гудки то просто тишина! И вот я уже с этим бъюсь второй день даже и не знаю что делать! Всё уже перепробывал!
Вот Вам sip debug что говорит но я ни чего не понимаю:
Aug 5 17:40:42 DEBUG[7948] chan_sip.c: Scheduled a registration timeout for sipnet.ru id #2
Aug 5 17:40:42 DEBUG[7948] chan_sip.c: Stopping retransmission on '022f3afa46ded8fd19b2dbf1757cc1c4@5.1.11.12' of Request 102: Match Found
Aug 5 17:40:42 DEBUG[7948] chan_sip.c: Stopping retransmission on '022f3afa46ded8fd19b2dbf1757cc1c4@5.1.11.12' of Request 103: Match Found
Aug 5 17:40:42 DEBUG[7948] chan_sip.c: Registration successful
Aug 5 17:40:42 DEBUG[7948] chan_sip.c: Cancelling timeout 2
Aug 5 17:41:03 VERBOSE[7948] logger.c:
<-- SIP read from 212.53.40.40:5060:
INVITE sip:0029273190@5.1.11.12 SIP/2.0
Via: SIP/2.0/UDP 212.53.40.40:5060;branch=z9hG4bK221376-kmbdctj;cgp=etc.tario.ru;upaddr=212.53.35.244;rport
P-CGP-Redirector: pupk@sipnet.ru
Record-Route: <sip:212.53.40.40:5060;lr>
Record-Route: <sip:192.168.40.71:5060;lr>
Record-Route: <sip:rev.832117-192.168.40.71.dialog.cgatepro;lr>
Via: SIP/2.0/UDP 212.53.35.244:61348;branch=z9hG4bK-2900a800-16291809
Max-Forwards: 69
From: "St. Petersburg" <sip:+78120000@tario.net>;tag=687908864-16291809
To: <sip:0029273190@sipnet.ru>
Call-ID: IWF-D224B734-9FCD-11DF-A517-9ADC4C3CA230@212.53.35.244
Contact: <sip:212.53.35.244:61348>
CSeq: 47669 INVITE
User-Agent: TarioSoftswitch/3.2.11
Content-Type: application/sdp
X-Ringcost: 2000
Content-Length: 218
v=0
o=Tario-SIPUA 16291809 100 IN IP4 91.198.130.135
s=SIP Call
c=IN IP4 91.198.130.135
t=0 0
m=audio 18688 RTP/AVP 8 18
c=IN IP4 91.198.130.135
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
Aug 5 17:41:03 VERBOSE[7948] logger.c: --- (17 headers 10 lines) ---
Aug 5 17:41:03 VERBOSE[7948] logger.c: Using INVITE request as basis request - IWF-D224B734-9FCD-11DF-A517-9ADC4C3CA230@212.53.35.244
Aug 5 17:41:03 VERBOSE[7948] logger.c: Sending to 212.53.40.40 : 5060 (NAT)
Aug 5 17:41:03 VERBOSE[7948] logger.c: Found peer 'sipnet'
Aug 5 17:41:03 DEBUG[7948] chan_sip.c: Setting NAT on RTP to 0
Aug 5 17:41:03 VERBOSE[7948] logger.c: Found RTP audio format 8
Aug 5 17:41:03 VERBOSE[7948] logger.c: Found RTP audio format 18
Aug 5 17:41:03 VERBOSE[7948] logger.c: Peer audio RTP is at port 91.198.130.135:18688
Aug 5 17:41:03 DEBUG[7948] chan_sip.c: Peer audio RTP is at port 91.198.130.135:18688
Aug 5 17:41:03 VERBOSE[7948] logger.c: Found description format PCMA
Aug 5 17:41:03 VERBOSE[7948] logger.c: Found description format G729
Aug 5 17:41:03 VERBOSE[7948] logger.c: Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x108 (alaw|g729)/video=0x0 (nothing), combined - 0x108 (alaw|g729)
Aug 5 17:41:03 VERBOSE[7948] logger.c: Non-codec capabilities: us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)
Aug 5 17:41:03 DEBUG[7948] chan_sip.c: Checking SIP call limits for device 0029273190
Aug 5 17:41:03 VERBOSE[7948] logger.c: Looking for 0029273190 in incoming (domain 5.1.11.12)
Aug 5 17:41:04 DEBUG[7948] chan_sip.c: build_route: Record-Route hop: <sip:212.53.40.40:5060;lr>
Aug 5 17:41:04 DEBUG[7948] chan_sip.c: build_route: Record-Route hop: <sip:192.168.40.71:5060;lr>
Aug 5 17:41:04 DEBUG[7948] chan_sip.c: build_route: Record-Route hop: <sip:rev.832117-192.168.40.71.dialog.cgatepro;lr>
Aug 5 17:41:04 VERBOSE[7948] logger.c: list_route: hop: <sip:212.53.40.40:5060;lr>
Aug 5 17:41:04 VERBOSE[7948] logger.c: list_route: hop: <sip:192.168.40.71:5060;lr>
Aug 5 17:41:04 VERBOSE[7948] logger.c: list_route: hop: <sip:rev.832117-192.168.40.71.dialog.cgatepro;lr>
Aug 5 17:41:04 VERBOSE[7948] logger.c: Transmitting (no NAT) to 212.53.40.40:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 212.53.40.40:5060;branch=z9hG4bK221376-kmbdctj;cgp=etc.tario.ru;upaddr=212.53.35.244;received=212.53.40.40;rport=5060
Via: SIP/2.0/UDP 212.53.35.244:61348;branch=z9hG4bK-2900a800-16291809
From: "St. Petersburg" <sip:+78120000@tario.net>;tag=687908864-16291809
To: <sip:0029273190@sipnet.ru>
Call-ID: IWF-D224B734-9FCD-11DF-A517-9ADC4C3CA230@212.53.35.244
CSeq: 47669 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:0029273190@5.1.11.12>
Content-Length: 0
---
Aug 5 17:41:04 VERBOSE[7948] logger.c: We're at 5.3.11.12 port 14956
Aug 5 17:41:04 VERBOSE[7948] logger.c: Adding codec 0x100 (g729) to SDP
Aug 5 17:41:04 VERBOSE[7948] logger.c: Adding codec 0x8 (alaw) to SDP
Aug 5 17:41:04 VERBOSE[7948] logger.c: Reliably Transmitting (no NAT) to 212.53.40.40:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 212.53.40.40:5060;branch=z9hG4bK221376-kmbdctj;cgp=etc.tario.ru;upaddr=212.53.35.244;received=212.53.40.40;rport=5060
Via: SIP/2.0/UDP 212.53.35.244:61348;branch=z9hG4bK-2900a800-16291809
Record-Route: <sip:212.53.40.40:5060;lr>
Record-Route: <sip:192.168.40.71:5060;lr>
Record-Route: <sip:rev.832117-192.168.40.71.dialog.cgatepro;lr>
From: "St. Petersburg" <sip:+78120000@tario.net>;tag=687908864-16291809
To: <sip:0029273190@sipnet.ru>;tag=as6f0bf49f
Call-ID: IWF-D224B734-9FCD-11DF-A517-9ADC4C3CA230@212.53.35.244
CSeq: 47669 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:0029273190@5.1.11.12>
Content-Type: application/sdp
Content-Length: 207
v=0
o=root 7948 7948 IN IP4 5.1.11.12
s=session
c=IN IP4 5.1.11.12
t=0 0
m=audio 14956 RTP/AVP 18 8
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - -Aug 5 17:41:04 WARNING[7948] rtp.c: RTP Read error: Connection reset by peer. Hanging up now.
Aug 5 17:41:04 DEBUG[7948] chan_sip.c: update_call_counter(0029273190) - decrement call limit counter
Aug 5 17:41:04 VERBOSE[7948] logger.c: Scheduling destruction of call 'IWF-D224B734-9FCD-11DF-A517-9ADC4C3CA230@212.53.35.244' in 32000 ms
Aug 5 17:41:04 DEBUG[7948] pbx.c: Function result is '"St. Petersburg" <+78120000>'
Aug 5 17:41:04 DEBUG[7948] pbx.c: Function result is '+78120000'
Aug 5 17:41:04 DEBUG[7948] pbx.c: Function result is '0029273190'
Aug 5 17:41:04 DEBUG[7948] pbx.c: Function result is 'incoming'
Aug 5 17:41:04 DEBUG[7948] pbx.c: Function result is 'SIP/0029273190-009aae68'
Aug 5 17:41:04 DEBUG[7948] pbx.c: Function result is '(null)'
Aug 5 17:41:04 DEBUG[7948] pbx.c: Function result is 'Playback'
Aug 5 17:41:04 DEBUG[7948] pbx.c: Function result is 'hello-world'
Aug 5 17:41:04 DEBUG[7948] pbx.c: Function result is '2010-08-05 17:41:04'
Aug 5 17:41:04 DEBUG[7948] pbx.c: Function result is '2010-08-05 17:41:04'
Aug 5 17:41:04 DEBUG[7948] pbx.c: Function result is '2010-08-05 17:41:04'
Aug 5 17:41:04 DEBUG[7948] pbx.c: Function result is '0'
Aug 5 17:41:04 DEBUG[7948] pbx.c: Function result is '0'
Aug 5 17:41:04 DEBUG[7948] pbx.c: Function result is 'ANSWERED'
Aug 5 17:41:04 DEBUG[7948] pbx.c: Function result is 'DOCUMENTATION'
Aug 5 17:41:04 DEBUG[7948] pbx.c: Function result is '(null)'
Aug 5 17:41:04 DEBUG[7948] pbx.c: Function result is '1281015664.0'
Aug 5 17:41:04 DEBUG[7948] pbx.c: Function result is '(null)'
Aug 5 17:41:04 VERBOSE[7948] logger.c:
<-- SIP read from 212.53.40.40:5060:
ACK sip:0029273190@5.1.11.12 SIP/2.0
Via: SIP/2.0/UDP 212.53.40.40:5060;branch=z9hG4bK221380-kmbdctj;cgp=etc.tario.ru;upaddr=212.53.35.244;rport
Via: SIP/2.0/UDP 212.53.35.244:61348;branch=z9hG4bK-2900a800-16291809
Max-Forwards: 69
From: <sip:+78120000@tario.net>;tag=687908864-16291809
To: <sip:0029273190@sipnet.ru>;tag=as6f0bf49f
Call-ID: IWF-D224B734-9FCD-11DF-A517-9ADC4C3CA230@212.53.35.244
CSeq: 47669 ACK
User-Agent: TarioSoftswitch/3.2.11
Content-Length: 0
Aug 5 17:41:04 VERBOSE[7948] logger.c: --- (10 headers 0 lines) ---
Aug 5 17:41:04 DEBUG[7948] chan_sip.c: Stopping retransmission on 'IWF-D224B734-9FCD-11DF-A517-9ADC4C3CA230@212.53.35.244' of Response 47669: Match Found
Aug 5 17:41:04 VERBOSE[7948] logger.c: set_destination: Parsing <sip:212.53.40.40:5060;lr> for address/port to send to
Aug 5 17:41:04 VERBOSE[7948] logger.c: set_destination: set destination to 212.53.40.40, port 5060
Aug 5 17:41:04 VERBOSE[7948] logger.c: Reliably Transmitting (no NAT) to 212.53.40.40:5060:
CANCEL sip:0029273190@5.1.11.12 SIP/2.0
Via: SIP/2.0/UDP 8.1.11.12:5060;branch=z9hG4bK2812c49c;rport
Route: <sip:212.53.40.40:5060;lr>,<sip:192.168.40.71:5060;lr>,<sip:rev.832117-192.168.40.71.dialog.cgatepro;lr>
From: <sip:002927319@sipnet.ru>
To: "St. Petersburg" <sip:+78120000@tario.net>;tag=687908864-16291809
Call-ID: IWF-D224B734-9FCD-11DF-A517-9ADC4C3CA230@212.53.35.244
CSeq: 101 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
Aug 5 17:41:04 VERBOSE[7948] logger.c: Scheduling destruction of call 'IWF-D224B734-9FCD-11DF-A517-9ADC4C3CA230@212.53.35.244' in 32000 ms
Aug 5 17:41:04 VERBOSE[7948] logger.c:
<-- SIP read from 212.53.40.40:5060:
SIP/2.0 481 No session found
Via: SIP/2.0/UDP 5.1.11.12:5060;branch=z9hG4bK2812c49c;rport=5060
From: <sip:0029273190@sipnet.ru>
To: "St. Petersburg" <sip:+78120000@tario.net>;tag=687908864-16291809
Call-ID: IWF-D224B734-9FCD-11DF-A517-9ADC4C3CA230@212.53.35.244
CSeq: 101 CANCEL
Server: CommuniGatePro/5.3.8
Content-Length: 0
Aug 5 17:41:04 VERBOSE[7948] logger.c: --- (8 headers 0 lines) ---
Aug 5 17:41:04 DEBUG[7948] chan_sip.c: Stopping retransmission on 'IWF-D224B734-9FCD-11DF-A517-9ADC4C3CA230@212.53.35.244' of Request 101: Match Found
Aug 5 17:41:14 DEBUG[7948] chan_sip.c: Auto destroying call '022f3afa46ded8fd19b2dbf1757cc1c4@5.1.11.12'
Aug 5 17:41:14 VERBOSE[7948] logger.c: Destroying call '022f3afa46ded8fd19b2dbf1757cc1c4@5.1.11.12'
Aug 5 17:41:22 VERBOSE[7948] logger.c:
<-- SIP read from 212.53.40.40:5060:
BYE sip:0029273190@5.1.11.12 SIP/2.0
Via: SIP/2.0/UDP 212.53.40.40:5060;branch=z9hG4bK221542-kmbdctj;cgp=etc.tario.ru;upaddr=212.53.35.244;rport
Via: SIP/2.0/UDP 212.53.35.244:61348;branch=z9hG4bK-2900a800-16291809
Max-Forwards: 69
From: <sip:+78120000@tario.net>;tag=687908864-16291809
To: <sip:0029273190@sipnet.ru>;tag=as6f0bf49f
Call-ID: IWF-D224B734-9FCD-11DF-A517-9ADC4C3CA230@212.53.35.244
CSeq: 47670 BYE
User-Agent: TarioSoftswitch/3.2.11
Content-Length: 0
Aug 5 17:41:22 VERBOSE[7948] logger.c: --- (10 headers 0 lines) ---
Aug 5 17:41:22 VERBOSE[7948] logger.c: Sending to 212.53.40.40 : 5060 (NAT)
Aug 5 17:41:22 VERBOSE[7948] logger.c: Transmitting (NAT) to 212.53.40.40:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 212.53.40.40:5060;branch=z9hG4bK221542-kmbdctj;cgp=etc.tario.ru;upaddr=212.53.35.244;received=212.53.40.40;rport=5060
Via: SIP/2.0/UDP 212.53.35.244:61348;branch=z9hG4bK-2900a800-16291809
From: <sip:+78120000@tario.net>;tag=687908864-16291809
To: <sip:0029273190@sipnet.ru>;tag=as6f0bf49f
Call-ID: IWF-D224B734-9FCD-11DF-A517-9ADC4C3CA230@212.53.35.244
CSeq: 47670 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:0029273190@8.3.11.12>
Content-Length: 0
---
Aug 5 17:41:22 VERBOSE[7948] logger.c: Destroying call 'IWF-D224B734-9FCD-11DF-A517-9ADC4C3CA230@212.53.35.244'
ug 5 17:42:27 NOTICE[7948] chan_sip.c: -- Re-registration for 0029273190@sipnet.ru
Aug 5 17:42:27 DEBUG[7948] chan_sip.c: Scheduled a registration timeout for sipnet.ru id #11
Aug 5 17:42:27 DEBUG[7948] chan_sip.c: >>> Re-using Auth data for 0029273190@sipnet.ru
Aug 5 17:42:27 VERBOSE[7948] logger.c: REGISTER 13 headers, 0 lines
Aug 5 17:42:27 VERBOSE[7948] logger.c: Reliably Transmitting (no NAT) to 212.53.40.40:5060:
REGISTER sip:sipnet.ru SIP/2.0
Via: SIP/2.0/UDP 5.1.11.14:5060;branch=z9hG4bK15392c93;rport
From: <sip:0029273190@sipnet.ru>;tag=as230acca4
To: <sip:0029273190@sipnet.ru>
Call-ID: 022f3afa46ded8fd19b2dbf1757cc1c4@5.1.11.12
CSeq: 104 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="0029273190", realm="etc.tario.ru", algorithm=MD5, uri="sip:sipnet.ru", nonce="82F09A976FDCD9A3F7D3", response="27bc12840ce2f75bd3ad0771836c3b49", opaque="opaqueData", qop=auth, cnonce="1743cb14", nc=00000002
Expires: 120
Contact: <sip:0029273190@5.1.11.12>
Event: registration
Content-Length: 0
---
Aug 5 17:42:27 VERBOSE[7948] logger.c:
<-- SIP read from 212.53.40.40:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 8.3.11.12:5060;branch=z9hG4bK15392c93;rport=5060
From: <sip:0029273190@sipnet.ru>;tag=as230acca4
To: <sip:0029273190@sipnet.ru>;tag=0EC33EFD
Call-ID: 022f3afa46ded8fd19b2dbf1757cc1c4@5.1.1.12
CSeq: 104 REGISTER
Expires: 120
Contact: <sip:0029273190@8.3.11.12>;expires=116
Event: registration
Date: Thu, 05 Aug 2010 13:41:42 GMT
Allow: PUBLISH,SUBSCRIBE
Supported: path,gruu
Allow-Events: presence,message-summary,reg,dialog,line-seize,keep-alive,refer
Server: CommuniGatePro/5.3.8
Content-Length: 0
Помогите Плиззз! |
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Сообщений: 124
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Re: Помогите настроить Asterisk для sipnet.ru !!!
портянку бы сократить. сип дебуг тут вроде бы ни к чему, а вот лог отладки астериска был бы интересней и информативней.
если так прет астериск и охота минимум гемора, можно вкатить на виртуалку, эмуляторы бесплатные.
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Сообщений: 6521
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Re: Помогите настроить Asterisk для sipnet.ru !!!
ExTazZ, все новые треды на тему Помогите настроить Asterisk для sipnet.ru надо сразу закрывать, а их авторов - кастрировать.
Тема уже избита до невозможности, и сама по себе служит хорошим итестом профпригодности.
Вопрос: может ли АБСОЛЮТНО любой человек скачать, проинсталлировать Астериск и настроить его на Sipnet.ru?
Ответ: разумеется нет, не абсолютно любой!
Вопрос: где критерий, по которму можно моментально определить - справится ли он сам или нет?
Ответ: критерий - время, за которое он сам найдёт в интернете всё необходимое для этого - дистрибутивы, инструкции и пр. Если это время растягивается до 1-2 дня, нужно пересмотреть самооценку человеку к исполнению такой задачи.
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Откуда: Saint-Petersburg
Сообщений: 3
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Re: Помогите настроить Asterisk для sipnet.ru !!!
jr: портянку бы сократить. сип дебуг тут вроде бы ни к чему, а вот лог отладки астериска был бы интересней и информативней.
если так прет астериск и охота минимум гемора, можно вкатить на виртуалку, эмуляторы бесплатные.
Я ба так и сделал только вот как останавливать сервисы в windows из виртуалки?
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Откуда: Уфа
Сообщений: 5856
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Re: Помогите настроить Asterisk для sipnet.ru !!!
человек решил уволиться и подложить свинью работодателю: останавливать сервисы по телефону
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Откуда: Saint-Petersburg
Сообщений: 3
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Re: Помогите настроить Asterisk для sipnet.ru !!!
Вот debug asteriska:
Aug 6 09:34:10 DEBUG[5248] chan_sip.c: Stopping retransmission on '2db57e3a4cf691c53426f7051f136c58@5.1.11.12' of Request 103: Match Found
Aug 6 09:34:10 DEBUG[5248] chan_sip.c: Registration successful
Aug 6 09:34:10 DEBUG[5248] chan_sip.c: Cancelling timeout 2
Aug 6 09:34:42 DEBUG[5248] chan_sip.c: Auto destroying call '2db57e3a4cf691c53426f7051f136c58@5.1.11.12'
Aug 6 09:35:53 DEBUG[5248] chan_sip.c: Setting NAT on RTP to 0
Aug 6 09:35:53 DEBUG[5248] chan_sip.c: Checking SIP call limits for device 0029273190
Aug 6 09:35:53 DEBUG[5248] chan_sip.c: build_route: Record-Route hop: <sip:212.53.40.40:5060;lr>
Aug 6 09:35:53 DEBUG[5248] chan_sip.c: build_route: Record-Route hop: <sip:192.168.40.72:5060;lr>
Aug 6 09:35:53 DEBUG[5248] chan_sip.c: build_route: Record-Route hop: <sip:rev.856693-192.168.40.72.dialog.cgatepro;lr>
Aug 6 09:35:53 VERBOSE[5248] logger.c: Sent RTP packet to 91.198.130.135:16932 (type 18, seq 22281, ts 160, len 20)
Aug 6 09:35:53 VERBOSE[5248] logger.c: Sent RTP packet to 91.198.130.135:16932 (type 18, seq 22282, ts 320, len 20)
Aug 6 09:35:53 WARNING[5248] rtp.c: RTP Read error: Connection reset by peer. Hanging up now.
Aug 6 09:35:53 DEBUG[5248] chan_sip.c: update_call_counter(0029273190) - decrement call limit counter
Aug 6 09:35:53 DEBUG[5248] pbx.c: Function result is '"St. Petersburg" <+78120000>'
Aug 6 09:35:53 DEBUG[5248] pbx.c: Function result is '+78120000'
Aug 6 09:35:53 DEBUG[5248] pbx.c: Function result is '0029273190'
Aug 6 09:35:53 DEBUG[5248] pbx.c: Function result is 'incoming'
Aug 6 09:35:53 DEBUG[5248] pbx.c: Function result is 'SIP/0029273190-0098fb98'
Aug 6 09:35:53 DEBUG[5248] pbx.c: Function result is '(null)'
Aug 6 09:35:53 DEBUG[5248] pbx.c: Function result is 'Playback'
Aug 6 09:35:53 DEBUG[5248] pbx.c: Function result is 'hello-world'
Aug 6 09:35:53 DEBUG[5248] pbx.c: Function result is '2010-08-06 09:35:53'
Aug 6 09:35:53 DEBUG[5248] pbx.c: Function result is '2010-08-06 09:35:53'
Aug 6 09:35:53 DEBUG[5248] pbx.c: Function result is '2010-08-06 09:35:53'
Aug 6 09:35:53 DEBUG[5248] pbx.c: Function result is '0'
Aug 6 09:35:53 DEBUG[5248] pbx.c: Function result is '0'
Aug 6 09:35:53 DEBUG[5248] pbx.c: Function result is 'ANSWERED'
Aug 6 09:35:53 DEBUG[5248] pbx.c: Function result is 'DOCUMENTATION'
Aug 6 09:35:53 DEBUG[5248] pbx.c: Function result is '(null)'
Aug 6 09:35:53 DEBUG[5248] pbx.c: Function result is '1281072953.0'
Aug 6 09:35:53 DEBUG[5248] pbx.c: Function result is '(null)'
Aug 6 09:35:53 DEBUG[5248] chan_sip.c: Stopping retransmission on 'IWF-34EEBD3C-A053-11DF-A586-9ADC4C3CA230@212.53.35.244' of Response 36173: Match Found
Aug 6 09:35:53 DEBUG[5248] chan_sip.c: Stopping retransmission on 'IWF-34EEBD3C-A053-11DF-A586-9ADC4C3CA230@212.53.35.244' of Request 101: Match Found
Aug 6 09:35:55 DEBUG[5248] chan_sip.c: Scheduled a registration timeout for sipnet.ru id #11
Aug 6 09:35:56 DEBUG[5248] chan_sip.c: Stopping retransmission on '2db57e3a4cf691c53426f7051f136c58@5.1.11.12' of Request 104: Match Found
Aug 6 09:35:56 DEBUG[5248] chan_sip.c: Registration successful
Aug 6 09:35:56 DEBUG[5248] chan_sip.c: Cancelling timeout 11
Aug 6 09:36:28 DEBUG[5248] chan_sip.c: Auto destroying call '2db57e3a4cf691c53426f7051f136c58@5.1.11.12'
Aug 6 09:37:41 DEBUG[5248] chan_sip.c: Scheduled a registration timeout for sipnet.ru id #15
Aug 6 09:37:41 DEBUG[5248] chan_sip.c: Stopping retransmission on '2db57e3a4cf691c53426f7051f136c58@5.1.11.12' of Request 105: Match Found
Aug 6 09:37:41 DEBUG[5248] chan_sip.c: Registration successful
Aug 6 09:37:41 DEBUG[5248] chan_sip.c: Cancelling timeout 15
Aug 6 09:38:13 DEBUG[5248] chan_sip.c: Auto destroying call '2db57e3a4cf691c53426f7051f136c58@5.1.11.12'
Aug 6 09:39:26 DEBUG[5248] chan_sip.c: Scheduled a registration timeout for sipnet.ru id #19
Aug 6 09:39:26 DEBUG[5248] chan_sip.c: Stopping retransmission on '2db57e3a4cf691c53426f7051f136c58@5.1.11.12' of Request 106: Match Found
Aug 6 09:39:26 DEBUG[5248] chan_sip.c: Registration successful
Aug 6 09:39:26 DEBUG[5248] chan_sip.c: Cancelling timeout 19
Aug 6 09:39:58 DEBUG[5248] chan_sip.c: Auto destroying call '2db57e3a4cf691c53426f7051f136c58@5.1.11.12'
Aug 6 09:41:11 DEBUG[5248] chan_sip.c: Scheduled a registration timeout for sipnet.ru id #23
Aug 6 09:41:11 DEBUG[5248] chan_sip.c: Stopping retransmission on '2db57e3a4cf691c53426f7051f136c58@5.1.11.12' of Request 107: Match Found
Aug 6 09:41:11 DEBUG[5248] chan_sip.c: Registration successful
Aug 6 09:41:11 DEBUG[5248] chan_sip.c: Cancelling timeout 23
Aug 6 09:41:43 DEBUG[5248] chan_sip.c: Auto destroying call '2db57e3a4cf691c53426f7051f136c58@5.1.11.12'
Aug 6 09:42:56 DEBUG[5248] chan_sip.c: Scheduled a registration timeout for sipnet.ru id #27
Aug 6 09:42:56 DEBUG[5248] chan_sip.c: Stopping retransmission on '2db57e3a4cf691c53426f7051f136c58@5.1.11.12' of Request 108: Match Found
Aug 6 09:42:56 DEBUG[5248] chan_sip.c: Registration successful
Aug 6 09:42:56 DEBUG[5248] chan_sip.c: Cancelling timeout 27
Aug 6 09:43:28 DEBUG[5248] chan_sip.c: Auto destroying call '2db57e3a4cf691c53426f7051f136c58@5.1.11.12' |
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Сообщений: 124
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Re: Помогите настроить Asterisk для sipnet.ru !!!
тачка стоит за натом, проверить настройки проброса портов, фаер. если непоможет, потрясти сипнет и сипнет вики. тема избитая, хотя каждый на сипнете застревал хоть на час.
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