[Asterisk-Users] SIP Trunk please help

[Ryan Pagquil]

Hi,
         I already contacted what I inputed on my softphone but we 
both can't hear each other. I used X-lite and the other is a hardware 
SIP phone. What could be the problem?

Thanks,
Ryan

At 03:03 PM 12/16/05, you wrote:
>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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