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Звонок через SIPNET

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Сообщений: 16

Звонок через SIPNET

Добрый день всем.

настраиваю * для междугородних звонков через SIPNET. Звоню себе на сотовый телефон. вызов начинает идти. я на сотовом ничего не нажимаю. Сбрасываю звонок с Xlite - а вызов продолжает идти...идти.. и по какому то долгому таймату сам по себе прекращается

asterisk 1.4.26.2
fedora 10

extensions.conf
[globals]

[general]

[default]
exten => s,1,verbose(1|Unrouted call handler)
exten => s,n,Answer()
exten => s,n,Wait(1)
exten => s,n,Playback(tt-weasels)
exten => s,n,Hangup()

[sipnet-in]
exten => s,1,Answer()
exten => s,n,Dial(Sip/1000,10)
exten => s,n,Hangup()

[internal]
exten => 500,1,Answer()
;exten => 500,n,Echo()
exten => 500,n,SayDigits(${EXTEN})
exten => 500,n,Hangup()

exten => _XXXX,1,Verbose(1|Extension 1001)
exten => _XXXX,n,Dial(SIP/${EXTEN},30)
exten => _XXXX,n,Hangup()

[external-call]
exten => _7XXXXXXXXXX,1,Dial(SIP/sipnet/${EXTEN})
exten => _7XXXXXXXXXX,n,Congestion()
exten => _7XXXXXXXXXX,n,Hangup()

exten => i,1,playback(pbx-invalid)
exten => i,n,HangUp

[phones]
include =>internal
include => external-call

sip.conf
[general]
bindaddr=XX1.XX1.XX1.XX1
;
; необходимо использовать что-то одно - или externip или externhost
;
externip = XX1.XX1.XX1.XX1
; ваш внешний IP-адрес, если он является постоянным
;
;externhost = MyServer.MyDomain.tld
; ваше доменное имя, если у вас динамический внешний IP-адрес и вы пользуетесь DynDNS
;
externrefresh = 60
;
localnet = 192.168.10.0/255.255.255.0 ; ваша внутренняя подсеть
;
nat = no
canreinvite = no

videosupport=no

useragent=SipPhone

register=sipnet_user:sipnet_pass@sipnet.ru/3581350

[sipnet]
type=friend
username=sipnet_user
secret=sipnet_pass
host=sipnet.ru
nat=no
fromuser=3581350
fromdomain=sipnet.ru
dtmfmode=rfc2833
insecure=very
canreinvite=no
context=sipnet-in

disallow=all
allow=g729
allow=g723.1
allow=ulaw
allow=alaw



[sets](!)
type = friend
context = phones
host = dynamic
disallow=all
allow=g729
allow=g723.1
allow=ulaw
allow=alaw


dtmfmode=rfc2833
secret=1234
nat=yes


[1000](sets)
mailbox=1000

[1001](sets)
mailbox=1001

[1002](sets)
mailbox=1002

[1003](sets)
mailbox=1003
nat=no


вырезки из лога с ошибкой....
<--- SIP read from XX2.XX2.XX2.XX2:49418 --->
CANCEL sip:79106602021@XX1.XX1.XX1.XX1 SIP/2.0
Via: SIP/2.0/UDP XX2.XX2.XX2.XX2:49418;branch=z9hG4bK-d8754z-d85689148e41625e-1---d8754z-;rport
To: "79106602021"<sip:79106602021@XX1.XX1.XX1.XX1>
From: "D"<sip:1003@XX1.XX1.XX1.XX1>;tag=d87cb269
Call-ID: Yzk4NjE2ZDVkMWNhZjgzZmExYTQxNmRkZGJhMWE5MjM.
CSeq: 2 CANCEL
Proxy-Authorization: Digest username="1003",realm="asterisk",nonce="1addd0ad",uri="sip:79106602021@XX1.XX1.XX1.XX1",response="d2
cee3b883d6e3f7d35311cfef9ae90c",algorithm=MD5
User-Agent: X-Lite release 1103k stamp 53621
Content-Length: 0


<------------->
[Nov 8 01:24:29] VERBOSE[27950] logger.c: --- (9 headers 0 lines) ---
[Nov 8 01:24:29] VERBOSE[27950] logger.c: Sending to XX2.XX2.XX2.XX2 : 49418 (NAT)
[Nov 8 01:24:29] DEBUG[27950] chan_sip.c: Setting SIP_ALREADYGONE on dialog Yzk4NjE2ZDVkMWNhZjgzZmExYTQxNmRkZGJhMWE5MjM.
[Nov 8 01:24:29] VERBOSE[27950] logger.c:
<--- Reliably Transmitting (NAT) to XX2.XX2.XX2.XX2:49418 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP XX2.XX2.XX2.XX2:49418;branch=z9hG4bK-d8754z-d85689148e41625e-1---d8754z-;received=XX2.XX2.XX2.XX2;rport=49418
From: "D"<sip:1003@XX1.XX1.XX1.XX1>;tag=d87cb269
To: "79106602021"<sip:79106602021@XX1.XX1.XX1.XX1>;tag=as16b463e2
Call-ID: Yzk4NjE2ZDVkMWNhZjgzZmExYTQxNmRkZGJhMWE5MjM.
CSeq: 2 INVITE
User-Agent: SipPhone
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


[Nov 8 01:24:29] VERBOSE[27950] logger.c:
<--- Transmitting (NAT) to XX2.XX2.XX2.XX2:49418 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP XX2.XX2.XX2.XX2:49418;branch=z9hG4bK-d8754z-d85689148e41625e-1---d8754z-;received=XX2.XX2.XX2.XX2;rport=49418
From: "D"<sip:1003@XX1.XX1.XX1.XX1>;tag=d87cb269
To: "79106602021"<sip:79106602021@XX1.XX1.XX1.XX1>;tag=as16b463e2
Call-ID: Yzk4NjE2ZDVkMWNhZjgzZmExYTQxNmRkZGJhMWE5MjM.
CSeq: 2 CANCEL
User-Agent: SipPhone
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:79106602021@XX1.XX1.XX1.XX1>
Content-Length: 0


[Nov 8 01:24:29] DEBUG[28826] rtp.c: Channel '<unspecified>' has no RTP, not doing anything
[Nov 8 01:24:29] DEBUG[28826] channel.c: Hanging up channel 'SIP/sipnet-09578908'
[Nov 8 01:24:29] DEBUG[28826] chan_sip.c: Hangup call SIP/sipnet-09578908, SIP callid 0ebb39a85a03602a65dc025e4190ab20@sipnet
.ru)
[Nov 8 01:24:29] VERBOSE[28826] logger.c: Scheduling destruction of SIP dialog '0ebb39a85a03602a65dc025e4190ab20@sipnet.ru' i
n 32000 ms (Method: INVITE)
[Nov 8 01:24:29] DEBUG[28826] chan_sip.c: Acked pending invite 103
[Nov 8 01:24:29] DEBUG[28826] chan_sip.c: Stopping retransmission on '0ebb39a85a03602a65dc025e4190ab20@sipnet.ru' of Request
103: Match Found
[Nov 8 01:24:29] VERBOSE[28826] logger.c: Reliably Transmitting (no NAT) to 212.53.40.40:5060:
CANCEL sip:79106602021@sipnet.ru SIP/2.0
Via: SIP/2.0/UDP XX1.XX1.XX1.XX1:5060;branch=z9hG4bK652a61dd;rport
From: "D" <sip:3581350@sipnet.ru>;tag=as15e45a8f
To: <sip:79106602021@sipnet.ru>
Call-ID: 0ebb39a85a03602a65dc025e4190ab20@sipnet.ru
CSeq: 103 CANCEL
User-Agent: SipPhone
Max-Forwards: 70
Content-Length: 0

---
[Nov 8 01:24:29] VERBOSE[27950] logger.c:
<--- SIP read from XX2.XX2.XX2.XX2:49418 --->
ACK sip:79106602021@XX1.XX1.XX1.XX1 SIP/2.0
Via: SIP/2.0/UDP XX2.XX2.XX2.XX2:49418;branch=z9hG4bK-d8754z-d85689148e41625e-1---d8754z-;rport
To: "79106602021"<sip:79106602021@XX1.XX1.XX1.XX1>;tag=as16b463e2
From: "D"<sip:1003@XX1.XX1.XX1.XX1>;tag=d87cb269
Call-ID: Yzk4NjE2ZDVkMWNhZjgzZmExYTQxNmRkZGJhMWE5MjM.
CSeq: 2 ACK
Content-Length: 0


<------------->
[Nov 8 01:24:29] VERBOSE[27950] logger.c: --- (7 headers 0 lines) ---
[Nov 8 01:24:29] DEBUG[27950] chan_sip.c: Stopping retransmission on 'Yzk4NjE2ZDVkMWNhZjgzZmExYTQxNmRkZGJhMWE5MjM.' of Respon
se 2: Match Found
[Nov 8 01:24:29] VERBOSE[28826] logger.c: Scheduling destruction of SIP dialog '0ebb39a85a03602a65dc025e4190ab20@sipnet.ru' i
n 32000 ms (Method: INVITE)
[Nov 8 01:24:29] DEBUG[28826] devicestate.c: Notification of state change to be queued on device/channel SIP/sipnet
[Nov 8 01:24:29] DEBUG[27943] chan_sip.c: Checking device state for peer sipnet
[Nov 8 01:24:29] DEBUG[27943] devicestate.c: Changing state for SIP/sipnet - state 1 (Not in use)
[Nov 8 01:24:29] DEBUG[27955] app_queue.c: Device 'SIP/sipnet' changed to state '1' (Not in use) but we don't care because th
ey're not a member of any queue.
[Nov 8 01:24:29] DEBUG[28826] app_dial.c: Exiting with DIALSTATUS=CANCEL.
[Nov 8 01:24:29] DEBUG[28826] pbx.c: Spawn extension (phones,79106602021,1) exited non-zero on 'SIP/1003-09574980'
[Nov 8 01:24:29] VERBOSE[28826] logger.c: == Spawn extension (phones, 79106602021, 1) exited non-zero on 'SIP/1003-09574980
'
[Nov 8 01:24:29] DEBUG[28826] channel.c: Soft-Hanging up channel 'SIP/1003-09574980'
[Nov 8 01:24:29] DEBUG[28826] channel.c: Hanging up channel 'SIP/1003-09574980'
[Nov 8 01:24:29] DEBUG[28826] chan_sip.c: Hangup call SIP/1003-09574980, SIP callid Yzk4NjE2ZDVkMWNhZjgzZmExYTQxNmRkZGJhMWE5M
jM.)
[Nov 8 01:24:29] DEBUG[28826] devicestate.c: Notification of state change to be queued on device/channel SIP/1003
[Nov 8 01:24:29] DEBUG[27943] chan_sip.c: Checking device state for peer 1003
[Nov 8 01:24:29] DEBUG[27943] devicestate.c: Changing state for SIP/1003 - state 1 (Not in use)
[Nov 8 01:24:29] DEBUG[27955] app_queue.c: Device 'SIP/1003' changed to state '1' (Not in use) but we don't care because they
're not a member of any queue.
[Nov 8 01:24:29] VERBOSE[27950] logger.c: Really destroying SIP dialog 'Yzk4NjE2ZDVkMWNhZjgzZmExYTQxNmRkZGJhMWE5MjM.' Method:
ACK
[Nov 8 01:24:29] VERBOSE[27950] logger.c:
<--- SIP read from 212.53.40.40:5060 --->
SIP/2.0 481 No session found
Via: SIP/2.0/UDP XX1.XX1.XX1.XX1:5060;branch=z9hG4bK652a61dd;rport=5060

From: "D" <sip:3581350@sipnet.ru>;tag=as15e45a8f
To: <sip:79106602021@sipnet.ru>;tag=D077303A
Call-ID: 0ebb39a85a03602a65dc025e4190ab20@sipnet.ru
CSeq: 103 CANCEL
Server: CommuniGatePro/5.2.17
Content-Length: 0


<------------->
[Nov 8 01:24:29] VERBOSE[27950] logger.c: --- (8 headers 0 lines) ---
[Nov 8 01:24:29] DEBUG[27950] chan_sip.c: Stopping retransmission on '0ebb39a85a03602a65dc025e4190ab20@sipnet.ru' of Request
103: Match Found
[Nov 8 01:24:29] WARNING[27950] chan_sip.c: Remote host can't match request CANCEL to call '0ebb39a85a03602a65dc025e4190ab20@
sipnet.ru'. Giving up.

[Nov 8 01:24:56] VERBOSE[27950] logger.c:
<--- SIP read from XX2.XX2.XX2.XX2:49418 --->

<------------->
[Nov 8 01:25:43] VERBOSE[27950] logger.c: --- (8 headers 0 lines) ---
[Nov 8 01:25:43] DEBUG[27950] chan_sip.c: Invalid SIP message - rejected , no callid, len 329
[Nov 8 01:25:45] VERBOSE[27950] logger.c:
<--- SIP read from 212.53.40.40:5060 --->
SIP/2.0 487 Request cancelled
Via: SIP/2.0/UDP XX1.XX1.XX1.XX1:5060;branch=z9hG4bK07f8d1f1;rport=5060
From: "D" <sip:3581350@sipnet.ru>;tag=as15e45a8f
To: <sip:79106602021@sipnet.ru>;tag=9f31a067-1741292
Call-ID: 0ebb39a85a03602a65dc025e4190ab20@sipnet.ru
CSeq: 103 INVITE
Server: CommuniGatePro/5.2.17
Content-Length: 0
2009-11-08 09:11

Avatara of switch
Откуда: Уфа
Сообщений: 5856

Re: Звонок через SIPNET

когда звонок проходит, голос ходит?
http://www.lynks.ru - Решения телефонии, мини-АТС, VoIP на основе Trixbox и Asterisk
2009-11-08 09:26

Сообщений: 16

Re: Звонок через SIPNET

switch, увы щас нет под рукой микрофона ( на работе остался)

но я как описал - не устанавливаю сессию. не принимаю вызов. на телефоне он повисает как не отвеченный...

между экстеншами голос проходил. я проверял ранее. на сипнет щас нет возможности проверить rtp

если позвонить на автоответчик - я его слышу. вот сейчас проверил
2009-11-08 09:41

Сообщений: 16

Re: Звонок через SIPNET

у сервера реальные адреса.

xlite - 2 варианта
за натом и без ната. результат одинаковый
2009-11-08 09:57

Сообщений: 16

Re: Звонок через SIPNET

проблема решилась.

у меня оказалось на сипнете зарегистрировано 2 устройства под одной учетной записью: сервер * и fxs адаптер. отключив fxs адаптер вызовы перестали повисать.

еще такая проблема есть
при звонке на сотовый номер я отбиваю вызов на сотовом телефоне и Xlite переходит в статус Call failed: Service unavailable.

[Nov 8 11:00:17] DEBUG[15240] chan_sip.c: Stopping retransmission on '4e6c98df73309b254489d1af1a484325@sipnet.ru' of Request
103: Match Found
[Nov 8 11:00:17] VERBOSE[15240] logger.c: -- Got SIP response 600 "Busy Everywhere" back from 212.53.40.40
[Nov 8 11:00:17] DEBUG[15240] chan_sip.c: Setting SIP_ALREADYGONE on dialog 4e6c98df73309b254489d1af1a484325@sipnet.ru
[Nov 8 11:00:17] VERBOSE[17962] logger.c: -- SIP/sipnet-09590c88 is busy
[Nov 8 11:00:17] DEBUG[17962] channel.c: Hanging up channel 'SIP/sipnet-09590c88'
[Nov 8 11:00:17] DEBUG[17962] chan_sip.c: Hangup call SIP/sipnet-09590c88, SIP callid 4e6c98df73309b254489d1af1a484325@sipnet
.ru)
[Nov 8 11:00:17] DEBUG[17962] devicestate.c: Notification of state change to be queued on device/channel SIP/sipnet
[Nov 8 11:00:17] DEBUG[15232] chan_sip.c: Checking device state for peer sipnet
[Nov 8 11:00:17] DEBUG[15232] devicestate.c: Changing state for SIP/sipnet - state 1 (Not in use)
[Nov 8 11:00:17] DEBUG[15245] app_queue.c: Device 'SIP/sipnet' changed to state '1' (Not in use) but we don't care because th
ey're not a member of any queue.
[Nov 8 11:00:17] VERBOSE[17962] logger.c: == Everyone is busy/congested at this time (1:1/0/0)
[Nov 8 11:00:17] DEBUG[17962] rtp.c: Channel '<unspecified>' has no RTP, not doing anything
[Nov 8 11:00:17] DEBUG[17962] app_dial.c: Exiting with DIALSTATUS=BUSY.
[Nov 8 11:00:17] DEBUG[17962] pbx.c: Launching 'Congestion'
[Nov 8 11:00:17] VERBOSE[17962] logger.c: -- Executing [79106602021@phones:2] Congestion("SIP/1003-095842b0", "") in new
stack
[Nov 8 11:00:17] DEBUG[17962] chan_sip.c: Setting SIP_ALREADYGONE on dialog Y2YxZjY0YzU0NmNkZmY3NGEwOWQ2MDI2MTMwYWE5MWM.
[Nov 8 11:00:17] DEBUG[17962] channel.c: Soft-Hanging up channel 'SIP/1003-095842b0'
[Nov 8 11:00:17] DEBUG[17962] devicestate.c: Notification of state change to be queued on device/channel SIP/1003
[Nov 8 11:00:17] DEBUG[15232] chan_sip.c: Checking device state for peer 1003
[Nov 8 11:00:17] DEBUG[15232] devicestate.c: Changing state for SIP/1003 - state 1 (Not in use)
[Nov 8 11:00:17] DEBUG[15245] app_queue.c: Device 'SIP/1003' changed to state '1' (Not in use) but we don't care because they
're not a member of any queue.
[Nov 8 11:00:17] DEBUG[17962] pbx.c: Spawn extension (phones,79106602021,2) exited non-zero on 'SIP/1003-095842b0'
[Nov 8 11:00:17] VERBOSE[17962] logger.c: == Spawn extension (phones, 79106602021, 2) exited non-zero on 'SIP/1003-095842b0
'
[Nov 8 11:00:17] DEBUG[17962] channel.c: Soft-Hanging up channel 'SIP/1003-095842b0'
[Nov 8 11:00:17] DEBUG[17962] channel.c: Hanging up channel 'SIP/1003-095842b0'
[Nov 8 11:00:17] DEBUG[17962] chan_sip.c: Hangup call SIP/1003-095842b0, SIP callid Y2YxZjY0YzU0NmNkZmY3NGEwOWQ2MDI2MTMwYWE5M
WM.)
[Nov 8 11:00:17] DEBUG[17962] devicestate.c: Notification of state change to be queued on device/channel SIP/1003
[Nov 8 11:00:17] DEBUG[15232] chan_sip.c: Checking device state for peer 1003
[Nov 8 11:00:17] DEBUG[15232] devicestate.c: Changing state for SIP/1003 - state 1 (Not in use)
[Nov 8 11:00:17] DEBUG[15245] app_queue.c: Device 'SIP/1003' changed to state '1' (Not in use) but we don't care because they
're not a member of any queue.
2009-11-08 11:11

Avatara of switch
Откуда: Уфа
Сообщений: 5856

Re: Звонок через SIPNET

ну вроде как так и должно быть
http://www.lynks.ru - Решения телефонии, мини-АТС, VoIP на основе Trixbox и Asterisk
2009-11-08 11:14

Сообщений: 16

Re: Звонок через SIPNET

а... надо добавить Busy() после Dial
2009-11-08 12:02

Сообщений: 16

Re: Звонок через SIPNET

switch, а сколько я могу делать одновременных вызовов через sipnet с разных exten?

т.е. есть у меня 10 exten'ов - они все могут звонить через одну учетную запись? или я должен создать дополнительные sip id для этих звонков?
2009-11-08 13:46

Откуда: Саратов
Сообщений: 414

Re: Звонок через SIPNET

major_dk:

а сколько я могу делать одновременных вызовов через sipnet с разных exten?
А что говорит техподдержка сипнета по этому поводу? Вы туда обращались?
+7(925)140-7438
2009-11-08 14:07

Avatara of switch
Откуда: Уфа
Сообщений: 5856

Re: Звонок через SIPNET

ну у меня 10-20 разговоров через одну учетку ходит
без проблем.
http://www.lynks.ru - Решения телефонии, мини-АТС, VoIP на основе Trixbox и Asterisk
2009-11-08 15:41

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