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<P style="MARGIN: 0in 0in 0pt" class=MsoNormal><SPAN style="BACKGROUND: white"><FONT size=3><FONT face=Calibri>I have a call transfer problem with Freeswitch <BR><BR>Here is the call flow: <BR><BR>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. <BR><BR>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. <BR><BR>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.<?xml:namespace prefix = o ns = "urn:schemas-microsoft-com:office:office" /><o:p></o:p></FONT></FONT></SPAN></P>
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<P style="MARGIN: 0in 0in 0pt" class=MsoNormal><SPAN style="BACKGROUND: white"><FONT size=3 face=Calibri>Thank you.</FONT></SPAN></P></DIV></div><br>
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