[Asterisk-Users] SIP Trunk please help

[Diyanat Ali]

yes

$AGI->exec('Dial', "SIP/$EXTEN\@seruser");


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 Mailing List - Non-Commercial 
>Discussion<asterisk-users at lists.digium.com>, 
>asterisk-users at lists.digium.com
>Subject: RE: [Asterisk-Users] SIP Trunk please help
>Date: Fri, 16 Dec 2005 13:56:09 +0800
>MIME-Version: 1.0
>X-OriginalArrivalTime: 16 Dec 2005 05:58:00.0170 (UTC) 
>FILETIME=[AB7B14A0:01C60205]
>
>Hi,
>         Thanks for the reply... Actually I'm using AGI to do it instead of 
>defining it on extensions.conf... Would it be the same in extensions.conf? 
>Should I write $AGI->exec('Dial', 'SIP/$EXTEN\@seruser'); to dial it from 
>AGI script (perl), is this correct?
>
>Thank you very much,
>Ryan
>
>At 01:45 PM 12/16/05, Diyanat Ali wrote:
>>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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>>
>>
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