Thanks Michael. That worked. But I ran into another issue. I am trying to make FS work as an application server in the IMS network. The IMS network while sending an INVITE to FS include a route header with Original Dialog Identifier (ODI) to indicate that the INVITE should be send back to the IMS network by FS. IMS network uses ODI to correlate the two INVITEs. When I use the dialplan with fs_path, I do not see the Route header in the terminating leg INVITE sent by FS. Is there a way to make FS function this way? Or Is there a way to copy a SIP route header from the originating leg to the terminating leg of a call? Thanks, Thomas From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael Jerris Sent: Thursday, 21 November 2013 1:43 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Configure Freeswitch as an IMS application server https://wiki.freeswitch.org/wiki/Sofia-SIP#Specifying_SIP_Proxy_With_fs_path On Nov 21, 2013, at 3:58 PM, Thomas Titty <ttitty@kineto.com<mailto:ttitty@kineto.com>> wrote: 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<http://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<mailto:sip_network_destination=$1@example.com>,sip_route_uri=sip:$1@example.com<mailto:1@example.com>,sip_invite,sip_invite_req_uri=sip:$1@example.com<mailto:1@example.com>}sofia/external/$1@p-cscf.com<mailto:sofia/external/$1@p-cscf.com>:5060"/> </condition> </extension> With this setting the SIP INVITE looks okay, but is being sent to example.com<http://example.com/> rather than p-cscf.com<http://p-cscf.com/>. From: freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf OfKen 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<x-msg://27/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. 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