[asterisk-users] RTP not being switched between both SIP endpoints

[Gareth Blades]

On 18/09/13 12:40, Kenny Watson wrote:
> Hi,
>
> Since opensips is not handling media (i presume) is the audio not already going direct from asterisk to the other endpoint?
>
> Thanks
>
> Kenny

Opensips wasnt handling the media so the audio was between the original 
caller and asterisk (with the signalling being relayed by opensips). It 
was just when we dialled onto the final destination via SIP asterisk 
stayed in the loop and didnt issue a reinvite.

Its all fixed now. Although we weren't using any features the AGI 
application was setting DYNAMIC_FEATURES to an empty string which was 
enough to keep asterisk in a loop. We stopped the AGI from setting the 
variable if there were no features and it started working.

Thanks
Gareth