In the dialplan you just need to have FS send the invite back to the S-CSCF for user B... This can be done a couple of ways, 1 is specify the URI for the user@s-cscf or to specify to route it via the s-cscf there are a few ways to choose from on this... See http://wiki.freeswitch.org/wiki/Variable_sip_network_destination for 1 way with links to others On 11/20/13 11:04 PM, "Thomas Titty" <ttitty@kineto.com> wrote:
Hello,
I installed Freeswitch 1.5.7 and am trying to configure it as an telephony application server in an IMS network.
Here is the high-level signaling flow that I am considering.
1. Two SIP users (User-A and User-B) registered with S-CSCF in the IMS network (not with Freeswitch)
2. User-A makes a voice call to User-B.
3. S-CSCF forwards the SIP INVITE to Freeswitch.
4. Freeswitch acting as a B2BUA changes SDP to anchor the media flow and sends SIP INVITE to User-B back to S-CSCF.
5. SIP 200 OK from User-B follows the same path as INVITE in reverse direction.
I went through the dial-plan documentation and could not figure out whether such a configuration is possible.
Please let me know whether such a configuration is feasible and any broad pointers on how to make it work would be appreciated.
Thanks, Thomas
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