Re: Segmentation Fault in H323 to SIP enviroment.

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Hi Diana,

Is this scenario possible with Yate:

SIP phone (option:register to sip server) -> Yate -> Gnugk (RADIUS for AAA) -> h323 endpoint 

As I read here Yate can do the translation, but is it possible to provide registration of SIP endpoints into Gnugk. The SIP username should be send to gnugk as h323id.

Thanks,
Jove



*********** REPLY SEPARATOR  ***********

On 1/19/2007 at 7:04 PM Diana Cionoiu wrote:

>Asterisk h323 chan is not stable, use Yate insted.
>
>Diana Cionoiu
>
>Pablo Parodi(yahoo) wrote:
>
>> Hi all,
>> My name is Pablo Parodi, and I trying to do this:
>>
>> H323 phone  <--->   GNUGK <---> Asterisk with oh323chan <---> SIP phone
>>
>> Maybe the scenario is a little more complicated, but no much different.
>>
>> If I call from the sip phone to the h323 phone the call is OK!
>>
>> but I got Segmentation Fault in when trying from the h323 to the SIP 
>> phone.
>>
>> I´m new with GNUGK (and Voip in general).  Witch info must I post? 
>> (tetherreal, logs(what is the path for the logs?))
>>
>>
>> THANKS IN ADVANCE!!!
>> ps: the versions I'm using are:
>> Pwlib v1.10.0
>> Openh323 v1.18.0
>> GNUGK v2.2.5
>> Asterisk v1.2.14
>> Asterisk oh323channel v0.7.3
>>
>>
>>
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