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<TITLE>Re: [Freeswitch-users] Configure Freeswitch as an IMS application server</TITLE>
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<FONT FACE="Monaco, Courier New"><SPAN STYLE='font-size:11pt'>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>
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On 11/20/13 11:04 PM, "Thomas Titty" <<a href="ttitty@kineto.com">ttitty@kineto.com</a>> wrote:<BR>
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</SPAN></FONT><BLOCKQUOTE><SPAN STYLE='font-size:11pt'><FONT FACE="Calibri, Verdana, Helvetica, Arial">Hello,<BR>
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I installed Freeswitch 1.5.7 and am trying to configure it as an telephony application server in an IMS network. <BR>
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Here is the high-level signaling flow that I am considering.<BR>
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1. Two SIP users (User-A and User-B) registered with S-CSCF in the IMS network (not with Freeswitch)<BR>
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</FONT><FONT FACE="Calibri, Verdana, Helvetica, Arial">2. User-A makes a voice call to User-B.<BR>
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</FONT><FONT FACE="Calibri, Verdana, Helvetica, Arial">3. S-CSCF forwards the SIP INVITE to Freeswitch.<BR>
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</FONT><FONT FACE="Calibri, Verdana, Helvetica, Arial">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>
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</FONT><FONT FACE="Calibri, Verdana, Helvetica, Arial">5. SIP 200 OK from User-B follows the same path as INVITE in reverse direction.<BR>
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</FONT><FONT FACE="Calibri, Verdana, Helvetica, Arial"> <BR>
I went through the dial-plan documentation and could not figure out whether such a configuration is possible.<BR>
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Please let me know whether such a configuration is feasible and any broad pointers on how to make it work would be appreciated.<BR>
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Thanks,<BR>
Thomas<BR>
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</SPAN></FONT></FONT><FONT FACE="Monaco, Courier New"><SPAN STYLE='font-size:11pt'>-- <BR>
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