Не могу настроить шлюз до провайдера
Вот пример для астериска с их сайта
Код:
asterisk
Подключение asterisk через SIP-trunk к нашему softswitch'у производится следующим образом):
/etc/asterisk/sip.conf
; == С регистрацией
; Рекомендуем регистрироваться на обоих серверах
register => tt:password@sip1.voipgw.com
register => tt:password@sip2.voipgw.com
[tt]
type=peer
username=tt
secret=password
host=sip.voipgw.com
nat=no
;canreinvite=yes ; до версии asterisk 1.8
directmedia=yes ; версия asterisk 1.8 и более
dtmfmode=rfc2833
qualify=500
disallow=all
allow=alaw
allow=ulaw
allow=g729
context=context
> sip show registry
Host Username Refresh State
sip1.voipgw.com:5060 tt 45 Registered
sip2.voipgw.com:5060 tt 45 Registered
/etc/asterisk/extensions.conf
exten => QQQQQ,1,Set(CALLERID(num)=7XXXXXXXXXX) ; где 7XXXXXXXXXX - задаваемый номер
exten => QQQQQ,n,Dial(SIP/7812ZZZZZZZ@tt,,g)
exten => QQQQQ,n,Hangup
-- Executing Dial("SIP/XXXXX-29c2", "SIP/7812ZZZZZZZ@tt||g") in new stack
-- Called 7812ZZZZZZZ@tt
-- SIP/tt-569a is making progress passing it to SIP/XXXXX-29c2
-- SIP/tt-569a is making progress passing it to SIP/XXXXX-29c2
-- SIP/tt-569a is ringing
-- SIP/tt-569a is making progress passing it to SIP/XXXXX-29c2
-- SIP/tt-569a answered SIP/XXXXX-29c2
-- Executing Hangup("SIP/XXXXX-29c2", "") in new stack
== Spawn extension (default, QQQQQ, 2) exited non-zero on 'SIP/XXXXX-29c2'
; == Без регистрации
[tt]
type=peer
host=sip.voipgw.com
nat=no
;canreinvite=yes ; до версии asterisk 1.8
directmedia=yes ; версия asterisk 1.8 и более
dtmfmode=rfc2833
qualify=500
disallow=all
allow=alaw
allow=ulaw
allow=g729
context=context
/etc/asterisk/extensions.conf
exten => QQQQQ,1,Set(CALLERID(num)=7XXXXXXXXXX) ; где 7XXXXXXXXXX - задаваемый номер
exten => QQQQQ,n,Dial(SIP/7812ZZZZZZZ@tt,,g)
exten => QQQQQ,n,Hangup
Вывод set verbose 3
-- Executing Dial("SIP/XXXXX-223c", "SIP/7812ZZZZZZZ@tt||g") in new stack
-- Called 087812ZZZZZZZ@tt
-- SIP/tt-148e is making progress passing it to SIP/XXXXX-223c
-- SIP/tt-148e is making progress passing it to SIP/XXXXX-223c
-- SIP/tt-148e is ringing
-- SIP/tt-148e is making progress passing it to SIP/XXXXX-223c
-- SIP/tt-148e answered SIP/XXXXX-223c
-- Executing Hangup("SIP/XXXXX-223c", "") in new stack
== Spawn extension (default, QQQQQ, 2) exited non-zero on 'SIP/XXXXX-223c
как сделать тоже для ФС?