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</o:shapelayout></xml><![endif]--></head><body lang=DA link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span lang=EN-GB style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>According to this article here:<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-GB style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><a href="http://anders.com/cms/266">http://anders.com/cms/266</a><o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal><span lang=EN-GB style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>the difference mainly lies in how FreeSwitchs handles open channels in comparison with Asterisk. FS uses one thread per channel while * keeps jumping between threads. At least that’s how I understand it.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-GB 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 0cm 0cm 0cm'><p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>Fra:</span></b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] <b>På vegne af </b>virendra bhati<br><b>Sendt:</b> 8. februar 2012 06:34<br><b>Til:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br><b>Emne:</b> Re: [asterisk-users] Asterisk V/s FreeSwitch<o:p></o:p></span></p></div><p class=MsoNormal><o:p> </o:p></p><div><p class=MsoNormal style='margin-bottom:12.0pt'>thanks Gilles, <br><br>After reading these web links. it's pretty clear that FreeSwitch is batter then Asterisk feature, quality wise. But asterisk is easy to used. <br><br>But the question is still open from my end.<br><br><b>How</b> <b>FreeSwitch can support 1000CC but asterisk not</b> ?<br><br>Because FreeSwitch used XML as configuration and asterisk plan text file ? <br>FreeSwitch used sofia_sip and asterisk used sip ?<br>Asterisk is PBX and FreeSwitch is SoftSwitch ?<br><br><o:p></o:p></p><div><p class=MsoNormal>On Tue, Feb 7, 2012 at 9:10 PM, Gilles <<a href="mailto:codecomplete@free.fr">codecomplete@free.fr</a>> wrote:<o:p></o:p></p><p class=MsoNormal>On Tue, 7 Feb 2012 17:08:18 +0530, virendra bhati <<a href="mailto:virbhati@gmail.com">virbhati@gmail.com</a>><br>wrote:<o:p></o:p></p><div><p class=MsoNormal style='margin-bottom:12.0pt'>>Why FreeSwitch can handle more then 1,000CC and asterisk only 25CC ? What<br>>technology FreeSwitch is used and asterisk don't. I don't know it's the<br>>right or wrong but this question come to my mind...<o:p></o:p></p></div><p class=MsoNormal>Provided Asterisk, even in release 1.8 or 10, does handle much fewer<br>concurrent calls than Freeswitch, you might find the answer in those<br>articles:<br><br>"How does FreeSWITCH compare to Asterisk?"<br><a href="http://www.freeswitch.org/node/117" target="_blank">www.freeswitch.org/node/117</a><br><br>"Asterisk vs FreeSWITCH"<br><a href="http://www.richappsconsulting.com/blog/blog-detail/asterisk-vs-freeswitch/" target="_blank">www.richappsconsulting.com/blog/blog-detail/asterisk-vs-freeswitch/</a><br><br>"Asterisk vs. FreeSWITCH"<br><a href="http://www.anders.com/cms/266" target="_blank">www.anders.com/cms/266</a><br><br>"Open Source VoIP: Asterisk or FreeSwitch?"<br><a href="http://www.zdnet.com/blog/greenfield/open-source-voip-asterisk-or-freeswitch/233" target="_blank">www.zdnet.com/blog/greenfield/open-source-voip-asterisk-or-freeswitch/233</a><br><br>"FreeSwitch vs Asterisk"<br><a href="http://www.dslreports.com/forum/r23246683-FreeSwitch-vs-Asterisk" target="_blank">www.dslreports.com/forum/r23246683-FreeSwitch-vs-Asterisk</a><o:p></o:p></p><div><div><p class=MsoNormal><br><br>--<br>_____________________________________________________________________<br>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>New to Asterisk? Join us for a live introductory webinar every Thurs:<br> <a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><o:p></o:p></p></div></div></div><p class=MsoNormal><br><br clear=all><br>-- <o:p></o:p></p><div><p class=MsoNormal><br>Thanks and regards<br><br> Virendra Bhati<br>+91-8885268942<br>Software Engineer<br>E-mail-: <a href="mailto:virbhati@gmail.com" target="_blank">virbhati@gmail.com</a><br>Skype id:- virbhati2<o:p></o:p></p></div><p class=MsoNormal><o:p> </o:p></p></div><p> <o:p></o:p></p></div></body></html>