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</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Hey, <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Don’t know if this will do the job or not but do take a look at deflect application: </span><a href="http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_deflect">http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_deflect</a> <o:p></o:p></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>This will remove the FS from the call path and tell the originator to go somewhere else. <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>I hope this helps, maybe some other gurus here suggest you anything better than this.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Regards,<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Gohar A.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'><p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] <b>On Behalf Of </b>Sam Govind<br><b>Sent:</b> Monday, August 15, 2011 10:58 AM<br><b>To:</b> FreeSWITCH Users Help<br><b>Subject:</b> Re: [Freeswitch-users] SIP proxy collect DTMF using FS<o:p></o:p></span></p></div><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>Thanks Brian for showing concern. I'm always open for ideas. What I'm trying to achieve is collect DTMF from user and then have my SIP Proxy verify if a particular caller is allowed to dial that destination(input as DTMF).<o:p></o:p></p><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>Obviously I could've done the same checks at FS, BUT I'm <b>required </b>to let SIP Proxy verify instead. FreeSWITCH is only required to get input, release the call and if Proxy allows the call only then call be routed to any other FreeSWITCH (Pool of FS LoadBalanced).<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>This is supposed to simplify the operations of FS , decrease the load volume on FS, and increase the call capacity.<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>If you've any better Ideas do share.<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><div><p class=MsoNormal>On Sun, Aug 14, 2011 at 8:39 AM, Brian West <<a href="mailto:brian@freeswitch.org">brian@freeswitch.org</a>> wrote:<o:p></o:p></p><div><p class=MsoNormal>What are you trying to do exactly because it sounds like you have selected the most painful way to accomplish it?<o:p></o:p></p><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal><span style='color:#888888'>/b<o:p></o:p></span></p></div><div><div><div><p class=MsoNormal><o:p> </o:p></p><div><div><p class=MsoNormal>On Aug 9, 2011, at 2:11 AM, Sam Govind wrote:<o:p></o:p></p></div><p class=MsoNormal><br><br><o:p></o:p></p><p class=MsoNormal><span style='font-size:13.5pt;font-family:"Helvetica","sans-serif"'>Hi guys,<br><br>I'm looking to establish a scenario like this, any idea how to do it, if its<br>possible.<br><br>1- SIP proxy send call to FS where DTMF will be collected. (I'm thinking of<br>using PlayAndGetDigits)<br>2- DTMF collected be sent back to SIP proxy while FS ends the call<br>3- Call at SIP proxy end keeps running for some other processing.<br><br>basically I just need FS to collect DTMF and send those back to SIP Proxy.<br><br>Any ideas are welcome.<br><br>Thanks.</span><o:p></o:p></p></div><p class=MsoNormal><o:p> </o:p></p></div></div></div></div><p class=MsoNormal style='margin-bottom:12.0pt'><br>_______________________________________________<br>Join us at ClueCon 2011, Aug 9-11, Chicago<br><a href="http://www.cluecon.com" target="_blank">http://www.cluecon.com</a> 877-7-4ACLUE<br><br>FreeSWITCH-users mailing list<br><a href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a><br><a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br><a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><o:p></o:p></p></div><p class=MsoNormal><o:p> </o:p></p></div></div></body></html>