[Asterisk-Users] From OH323 to SIP or OH323 without gatekeeper

[Guillermo Salas M.]

Bruno Hertz wrote:
> "Joe S" <printingfoot at hotmail.com> writes:
> 
> 
>>Hi,
>>
>>I am new with asterisk. I was wondering if there is a way to call a
>>OH323 user or SIP user using Netmeeting/SJPhone with H323 as the
>>default protocol without having a gatekeeper.
>>
>>I can make a call from SIP to OH323 by specifying it in the
>>extensions.conf file, like:
>>
>>exten=>1001, 1, Dial(OH323/10.10.10.1)
>>
>>so I was wondering if there was a way to call from OH323 to SIP or OH323.
> 
> 
> Sure. Just specify in oh323.conf the context where incoming calls
> should go. That context then can include dial statements for any
> protocol, SIP, H323, IAX, whatever. See the Wiki for details on how to
> setup dial plans.
> 
> Finally, instruct your H323 phone to use asterisk as a gateway
> resp. proxy, not a gatekeeper. Any calls will then go through
> asterisk, and to the context you specified.
> 
> I'm doing that with Gnomemeeting all the time, and it works without
> problems.

Mayabe can you show us a little sample? I can call from Gnomemeeting to 
Xlite, but no from xlite to gnomemeeting.

Best regards,


> 
> Regards, Bruno.
> 
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