[Asterisk-Users] RTP session traversing Asterisk server ...

[Artur C. Severo]

Dear all,

Concerning the REINVITE discussion...
Does anybody can confirm if the Cisco ATA186 accepts the REINVITE Sip
Message, and if is compatible with this feature?
In other words, is it possible to make the ATA186 change the RTP destination
and start sending the media packets straight to destination point instead of
keep sending to Asterisk?

Thanks & Regards,

Artur C. Severo Eng., M.Sc.
Network Engineer
Tel: 55 51 3328 0636 #242



 "Low, Adam" wrote:
>
> Thanks all,
>
> I spent some time on this last night with packet sniffer in
> hand, the 'canreinvite' option makes sense and seems to work
> well for me (running latest * CVS release) when used between
> 79xx phones and the AS5300 gateway although I get some
> somewhat expected problems with 79xx that are NAT'd behind
> ADSL/cable connections.
>
> I don't seem to be hitting the bug that Dave mentioned below ...
>