[asterisk-users] Send SIP Re-invite.

[Olle E Johansson]

8 maj 2007 kl. 15.40 skrev Joshua Colp:

> Rohan Hathiwala wrote:
>> Hi,
>>  I need asterisk to instruct the other side to send RTP to a  
>> conference
>> server running on a different machine. The conference server does not
>> understand SIP so I cannot use the SIP REFER method.
>> I have another question. Suppose when processing a SIP INVITE we  
>> want to use
>> asterisk only for call control and let another server handle the  
>> RTP is
>> there a clean way to do this in asterisk.
>> Regards,
>> Rohan Hathiwala.
>
> Asterisk/chan_sip wasn't designed to be able to do this. You're  
> going to end up modifying things... potentially a lot. If the  
> conference server does SIP though you can just dial it, make sure  
> canreinvite is set to yes, and audio should go direct.
>

...psst...
There's a patch in the bug tracker that I believe is what you want.  
Please test and review, add your comments.

/O