[Freeswitch-users] Call Transfer Problem

[Michael Jerris]

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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