Re: gnugk: how to link with Asterisk via SIP or H323

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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 <4lists@xxxxxxxxx> wrote:
> 
>> From: Edson - Lists <4lists@xxxxxxxxx>
>> Subject: Re: gnugk: how to link with Asterisk via SIP or H323
>> To: "bilal ghayyad" <bilmar_gh@xxxxxxxxx>
>> 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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