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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