[Freeswitch-users] Configure Freeswitch as an IMS application server

[Thomas Titty]

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 at example.com<mailto:user-a at example.com> and user-b at example.com<mailto:user-b at example.com>, and P-CSCF at p-cscf.com

I would like the SIP INVITE from FS to look like

INVITE sip:user-b at example.com SIP/2.0
From: "user-a" <sip:user-a at example.com>;tag=a7c50F74399yc
To: <sip:userba at 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 at example.com,sip_route_uri=sip:$1 at example.com,sip_invite,sip_invite_req_uri=sip:$1 at example.com}sofia/external/$1 at 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 at lists.freeswitch.org [mailto:freeswitch-users-bounces at 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 at 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 at 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 at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at 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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131121/9bcb998d/attachment-0001.html