I tried all of the variables (sip_network_destination, sip_route_uri, sip_invite_req_uri) that you suggested but the SIP INVITE is not being routed to the p-cscf. I have 2 sip users (say) user-a@example.com<mailto:user-a@example.com> and user-b@example.com<mailto:user-b@example.com>, and P-CSCF at p-cscf.com I would like the SIP INVITE from FS to look like INVITE sip:user-b@example.com SIP/2.0 From: "user-a" <sip:user-a@example.com>;tag=a7c50F74399yc To: <sip:userba@p-cscf.com:5060> Here is what my dial-plan looks like <extension name="pcscf_outbound"> <condition field="destination_number" expression="^(\d{10})$"> <action application="bridge" data="{ sip_network_destination=$1@example.com,sip_route_uri=sip:$1@example.com,sip_invite,sip_invite_req_uri=sip:$1@example.com}sofia/external/$1@p-cscf.com:5060"/> </condition> </extension> With this setting the SIP INVITE looks okay, but is being sent to example.com rather than p-cscf.com. From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Ken Rice Sent: Thursday, 21 November 2013 10:12 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Configure Freeswitch as an IMS application server 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 ________________________________ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org G+ ClueCon : http://fs0.us/cluecon-gplus FB ClueCon : http://fs0.us/cluecon-fb G+ FreeSwitch : http://fs0.us/freeswitch-gplus FB FreeSWITCH : http://fs0.us/freeswitch-fb Twitter : @FreeSWITCH_WIRE irc.freenode.net #freeswitch