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
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