I am not sure if my question did not get post correctly earlier. I wonder
whether anyone can give me any recommendations.
Here is the call flow:
I call from the PSTN (A party) into my Polycom phone (B-party) which is
registered to FreeSwtich. The Freeswtich is setup not to route media as I have
an SBC acting as a mirror proxy that will do all the NAT and media routing.
The inbound call is setup fine and there is two way voice. I then blind
transfer from the Polycom to my Cell phone. I see the polycom send a SIP refer
to Freeswitch and it sends a 202 accepted fine and that leg between the Polycom
(B party) and the A party is torn down fine like its supposed to be. The
Freeswitch places the outbound call (the number the call is transferring to
C-party) and that call completes. However now there is one way audio between
the A party and C party . I see RTP streaming back from the egress carrier
where the call was transfered to so the A party can hear the C party but the C
party cannot hear the A party . When I look at the SIP traces of the original
inbound call from the A-party I see a SIP re-invite from free switch to place
the call on hold (contains Freeswitch RTP address to I can hear hold music)
while it is transferring the call and the A-party does hear on hold music from
Freeswitch while the call is being
transferred. However I do not see a second re-invite from freeswitch to pass
the media IP it got from the egress leg back to the original inbound leg. If my
inbound gateway does not get a re-invite from Freeswitch to redirect its media
to the new RTP address of of the egress carrier it will not do so hence the one
way voice.
How do I get the Freeswitch to re-invite the original ingress leg once it gets
the SIP 183 from the egress with the new RTP info ? Free switch is sending the
first SIP re-invite to my inbound gateway with new media IP (IP of itself) so
the A-party can hear on hold music , but does not send a second re-invite to my
inbound gateway after it receives the new RTP address from the egress carrier
for the call that was transferred back out.
Thank you.
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