[Asterisk-Users] SIP Trunk please help

[Diyanat Ali]

in the sip.conf have the following enteries

; for regsitering with ser
register:seruser:secret at 0.0.0.0:5060;(put ser machine ip:port)

;add a user for the ser machine
[seruser]
type=friend
host=0.0.0.0 ;(put ser machine ip here)
nat=no ;(change as needed )
canreinvite=yes ;(change as needed)
insecure=very ;(change as needed)
disallow=all
allow=ulaw
allow=gsm
context=sip
dtmfmode=rfc2833

in extensions.conf under contect [sip]

[sip]
;replace extension and the priority  to macth your dial plan
exten => _X.,1,Dial(SIP/${EXTEN:}@seruser) ;(seruser is  defined in 
sip.conf)



Diyanat


>From: Ryan Pagquil <rpagquil at philonline.com>
>Reply-To: Asterisk Users Mailing List - Non-Commercial 
>Discussion<asterisk-users at lists.digium.com>
>To: asterisk-users at lists.digium.com
>Subject: [Asterisk-Users] SIP Trunk please help
>Date: Fri, 16 Dec 2005 10:31:24 +0800
>MIME-Version: 1.0
>
>Hi,
>
>	I've been setting up asterisk for prepaid use. I'm testing to call a SER 
>registered user from the Asterisk just to simulate the prepaid calls. Now, 
>I can already contact Asterisk and it prompts me to input my call card 
>number and after that I dial in the number I want to call (a SER registered 
>device). My question is how can I implement on sip.conf to use my SER as 
>the trunk line? So that calls will be forwarded to it. Do I also need to 
>register asterisk on SER?How?
>
>Please help!
>
>Thanks,
>
>Ryan
>
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