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

[Michael Jerris]

https://wiki.freeswitch.org/wiki/Sofia-SIP#Specifying_SIP_Proxy_With_fs_path

On Nov 21, 2013, at 3:58 PM, Thomas Titty <ttitty at 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 at example.com and 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 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 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
>  
>  
>  
>  
>  
>  
> 
> _________________________________________________________________________
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> Ken
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