Please open a bug on http://jira.freeswitch.org for this issue.
Please test it on current svn trunk first as well.
Mike
On Sep 4, 2009, at 7:54 PM, DJB wrote:
I have a call transfer problem with Freeswitch
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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