It would be good to You access and search info about two projects: 
Asterisk and Yate. Remeber: google is Your friend...

Both have capabilities to handle SIP _AND_ H323 protocol.... even others 
are suported.

Edson.

P.S.: Ever, always, respond an e-mail started on a list, including the 
list, at least as copied.... ;)

bilal ghayyad escreveu:
> I heared that they were working on SIP to H323 Translator? How can we find 
> such thing that resolve the problem of routing the calls from gnugk to 
> Asterisk? Maybe we need H323 to SIP translator?
> 
> Regards
> Bilal
> 
> 
> --- On Tue, 7/14/09, Edson - Lists <4li...@gmail.com> wrote:
> 
>> From: Edson - Lists <4li...@gmail.com>
>> Subject: Re: gnugk: how to link with Asterisk via SIP or H323
>> To: "bilal ghayyad" <bilmar...@yahoo.com>
>> Date: Tuesday, July 14, 2009, 4:48 PM
>> You got it... GnuGK deals only with
>> H323 protocol family... :)
>>
>> Edson.
>>
>> bilal ghayyad escreveu:
>>> Do u mean that Gnugk does not support SIP at all?
>>>
>>> Regards
>>> Bilal
>>>
>>> ------------------- 
>>>> It isn't.... ;) GnuGK is about H.323... not
>> SIP....
>>>> For SIP routing, take a look on the *SER family
>> (SER,
>>>> KAMAILIO or OPENSIPS)... or one of the SIP PBX
>> like Asterisk, Yate,
>>>> a.s.o.
>>>>
>>>> Edson.
>>>>
>>>> bilal ghayyad escreveu:
>>>>> Hi All;
>>>>>
>>>>> I do not know if any one tried to route calls
>> from
>>>> gnugk to Asterisk, so the best way is to use SIP
>> trunk or to
>>>> use H323 trunk?
>>>>> Because actually I discover that the H323 in
>> Asterisk
>>>> is very weak, so I am thinking is the routing to
>> be via SIP,
>>>> but how is the SIP in the gnugk?
>>>>> Any advise?
>>>>> Regards
>>>>> Bilal
>>>
>>>
>>>        
> 
> 
>       
> 

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