[Asterisk-Users] SIP conections, with RTP not going trough Asterisk

[Olle E Johansson]

12 apr 2006 kl. 14.58 skrev Ronald Wiplinger:

> Tiago Stein D`Agostini wrote:
>> Hi,
>>
>>   Ie been looking for some time how to use asterisk  to initiate  
>> SIP connections between 2 IP phones,  but afetr initiated the  
>> communication making the RTP go directly from one telephone to the  
>> other, without passing by asterisk. Unfortunately I found no  
>> explanations of how to do it.
>>
>> Does anyone care to give a pointer to any explanation about how to  
>> do it?
>>
> canreinvite=yes
> and look at the options for dial()
>>
>> Thanks in advance
>>
Actually, it's the default mode. Just connect your phones to Asterisk  
on the same LAN, and Asterisk will
get out of the media path.

/Olle


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* Olle E. Johansson - oej at edvina.net
* Asterisk Training http://edvina.net/training/