[Freeswitch-users] Call Transfer Question
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. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] call transfer question
I have a call scenario that involves transferring the call and dropping out of the SIP/RTP stream. I need to accept the SIP call, play a prompt, and retrieve a pin code. After a database lookup, I need to transfer the call to another FS server and drop out of the SIP path. I have done this with the RTP media stream previously. I am not sure what I need to do to drop out of the SIP path. Is this possible on FS? Jonathan ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] call transfer question
You can use deflect to accomplish this.. it will do a refer to the other FS box. /b On Dec 12, 2008, at 11:08 AM, jonathan augenstine wrote: I have a call scenario that involves transferring the call and dropping out of the SIP/RTP stream. I need to accept the SIP call, play a prompt, and retrieve a pin code. After a database lookup, I need to transfer the call to another FS server and drop out of the SIP path. I have done this with the RTP media stream previously. I am not sure what I need to do to drop out of the SIP path. Is this possible on FS? Jonathan ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] call transfer question
Thank you, that is exactly what I need. On Fri, Dec 12, 2008 at 9:14 AM, Brian West br...@freeswitch.org wrote: You can use deflect to accomplish this.. it will do a refer to the other FS box. /b On Dec 12, 2008, at 11:08 AM, jonathan augenstine wrote: I have a call scenario that involves transferring the call and dropping out of the SIP/RTP stream. I need to accept the SIP call, play a prompt, and retrieve a pin code. After a database lookup, I need to transfer the call to another FS server and drop out of the SIP path. I have done this with the RTP media stream previously. I am not sure what I need to do to drop out of the SIP path. Is this possible on FS? Jonathan ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
