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> 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 and 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
 
 
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 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> 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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