Blades
[mailinglist+aster...@dns99.co.uk]
Sent: 17 September 2013 11:17
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] RTP not being switched between both SIP endpoints
We have a system where calls are coming in from telcos via an opensips
server and then being redirected out
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
We have a system where calls are coming in from telcos via an opensips
server and then being redirected out to a regular sip destination.
There is no NAT, DTMF features, call recording, or codec translation
being performed so I would expect asterisk to issue a reinvite after the
call is
