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<title>Re: [Freeswitch-users] Configure Freeswitch as an IMS application server</title>
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<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D;mso-fareast-language:EN-US">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.<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D;mso-fareast-language:EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D;mso-fareast-language:EN-US">I have 2 sip users (say)
<a href="mailto:user-a@example.com">user-a@example.com</a> and <a href="mailto:user-b@example.com">
user-b@example.com</a>, and P-CSCF at p-cscf.com <o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D;mso-fareast-language:EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D;mso-fareast-language:EN-US">I would like the SIP INVITE from FS to look like<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D;mso-fareast-language:EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D;mso-fareast-language:EN-US">INVITE sip:user-b@example.com SIP/2.0<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D;mso-fareast-language:EN-US">From: "user-a" <sip:user-a@example.com>;tag=a7c50F74399yc<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D;mso-fareast-language:EN-US">To: <<a href="sip:userba@p-cscf.com:5060">sip:userba@p-cscf.com:5060</a>><o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D;mso-fareast-language:EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D;mso-fareast-language:EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D;mso-fareast-language:EN-US">Here is what my dial-plan looks like<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D;mso-fareast-language:EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D;mso-fareast-language:EN-US"> <extension name="pcscf_outbound"><o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D;mso-fareast-language:EN-US"> <condition field="destination_number" expression="^(\d{10})$"><o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D;mso-fareast-language:EN-US"> <action application="bridge" data="{</span>
<span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D;mso-fareast-language:EN-US">
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"/><o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D;mso-fareast-language:EN-US"> </condition><o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D;mso-fareast-language:EN-US"> </extension><o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D;mso-fareast-language:EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D;mso-fareast-language:EN-US">With this setting the SIP INVITE looks okay, but is being sent to example.com rather than p-cscf.com.<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D;mso-fareast-language:EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D;mso-fareast-language:EN-US"><o:p> </o:p></span></p>
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<p class="MsoNormal"><b><span lang="EN-US" style="font-size:11.0pt;font-family:"Calibri","sans-serif"">From:</span></b><span lang="EN-US" style="font-size:11.0pt;font-family:"Calibri","sans-serif""> freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org]
<b>On Behalf Of </b>Ken Rice<br>
<b>Sent:</b> Thursday, 21 November 2013 10:12 AM<br>
<b>To:</b> FreeSWITCH Users Help<br>
<b>Subject:</b> Re: [Freeswitch-users] Configure Freeswitch as an IMS application server<o:p></o:p></span></p>
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<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal" style="margin-bottom:12.0pt"><span style="font-size:11.0pt;font-family:"Courier New"">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
<a href="http://wiki.freeswitch.org/wiki/Variable_sip_network_destination">http://wiki.freeswitch.org/wiki/Variable_sip_network_destination</a> for 1 way with links to others<br>
<br>
<br>
On 11/20/13 11:04 PM, "Thomas Titty" <<a href="ttitty@kineto.com">ttitty@kineto.com</a>> wrote:</span><o:p></o:p></p>
<blockquote style="margin-top:5.0pt;margin-bottom:5.0pt">
<p class="MsoNormal" style="margin-bottom:12.0pt"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif"">Hello,<br>
<br>
I installed Freeswitch 1.5.7 and am trying to configure it as an telephony application server in an IMS network. <br>
<br>
Here is the high-level signaling flow that I am considering.<br>
<br>
1. Two SIP users (User-A and User-B) registered with S-CSCF in the IMS network (not with Freeswitch)<br>
</span><span style="font-size:11.0pt;font-family:"Courier New""><br>
</span><span style="font-size:11.0pt;font-family:"Calibri","sans-serif"">2. User-A makes a voice call to User-B.<br>
</span><span style="font-size:11.0pt;font-family:"Courier New""><br>
</span><span style="font-size:11.0pt;font-family:"Calibri","sans-serif"">3. S-CSCF forwards the SIP INVITE to Freeswitch.<br>
</span><span style="font-size:11.0pt;font-family:"Courier New""><br>
</span><span style="font-size:11.0pt;font-family:"Calibri","sans-serif"">4. Freeswitch acting as a B2BUA changes SDP to anchor the media flow and sends SIP INVITE to User-B back to S-CSCF.<br>
</span><span style="font-size:11.0pt;font-family:"Courier New""><br>
</span><span style="font-size:11.0pt;font-family:"Calibri","sans-serif"">5. SIP 200 OK from User-B follows the same path as INVITE in reverse direction.<br>
</span><span style="font-size:11.0pt;font-family:"Courier New""><br>
</span><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""><br>
I went through the dial-plan documentation and could not figure out whether such a configuration is possible.<br>
<br>
Please let me know whether such a configuration is feasible and any broad pointers on how to make it work would be appreciated.<br>
<br>
Thanks,<br>
Thomas<br>
<br>
<br>
<br>
<br>
<br>
</span><span style="font-size:11.0pt;font-family:"Courier New""><o:p></o:p></span></p>
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<p class="MsoNormal"><span style="font-size:10.0pt;font-family:Consolas">_________________________________________________________________________<br>
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</blockquote>
<p class="MsoNormal"><span style="font-size:10.0pt;font-family:Consolas"><br>
</span><span style="font-size:11.0pt;font-family:"Courier New"">-- <br>
Ken<br>
<a href="http://www.FreeSWITCH.org">http://www.FreeSWITCH.org</a><br>
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