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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'>Realtime is probably better for administration, but do you want to throw a layer of complication into such a large undertaking? I wouldn’t want 20,000 people screaming at me because MYSQL crapped out.<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><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"'> asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] <b>On Behalf Of </b>Ing CIP. Alejandro Celi Mariátegui<br><b>Sent:</b> Friday, May 25, 2012 2:40 PM<br><b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br><b>Subject:</b> Re: [asterisk-users] twenty thousands (20, 000) users, which asterisk and how many servers<o:p></o:p></span></p></div></div><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal style='margin-bottom:12.0pt'><br>Asterisk Realtime is better for administration. <br><br>Performance, IMHO is the same issue. I'm not lucky to made large implementations to test these.<br><br>Regards,<o:p></o:p></p><table class=MsoNormalTable border=0 cellspacing=0 cellpadding=0 width="100%" style='width:100.0%'><tr><td style='padding:0in 0in 0in 0in'><p class=MsoNormal>-- <br>Ing CIP. Alejandro Celi Mariátegui <br><<a href="mailto:alex@linux.org.pe">alex@linux.org.pe</a>><br><a href="http://cipher.pe/web/nuestra-experiencia.html">http://cipher.pe/web/nuestra-experiencia.html</a> <o:p></o:p></p></td></tr></table><p class=MsoNormal><br><br><br>El vie, 25-05-2012 a las 07:06 -0700, bilal ghayyad escribió: <o:p></o:p></p><pre><o:p> </o:p></pre><pre><span style='color:black'>Hi John;</span><o:p></o:p></pre><pre><o:p> </o:p></pre><pre><span style='color:black'>For 20,000 users: Is it better to use Asterisk realtime configuration or to use conf files?</span><o:p></o:p></pre><pre><o:p> </o:p></pre><pre><span style='color:black'>I readed the below link but did not understand which GUI that works with asterisk realtime?</span><o:p></o:p></pre><pre><o:p> </o:p></pre><pre><span style='color:black'><a href="http://www.freepbx.org/trac/wiki/AsteriskRealtime">http://www.freepbx.org/trac/wiki/AsteriskRealtime</a></span><o:p></o:p></pre><pre><o:p> </o:p></pre><pre><o:p> </o:p></pre><pre><span style='color:black'>Regards</span><o:p></o:p></pre><pre><span style='color:black'>Bilal</span><o:p></o:p></pre><pre><o:p> </o:p></pre><pre><span style='color:black'>------------</span><o:p></o:p></pre><pre><span style='color:black'>> </span><o:p></o:p></pre><pre><span style='color:black'>> > My question is:</span><o:p></o:p></pre><pre><span style='color:black'>> ></span><o:p></o:p></pre><pre><span style='color:black'>> > Is it really possible to have the asterisk</span><o:p></o:p></pre><pre><span style='color:black'>> configuration in the database server instead of having it in</span><o:p></o:p></pre><pre><span style='color:black'>> conf files? HOW? I am asking this because what I noticed in</span><o:p></o:p></pre><pre><span style='color:black'>> AsteriskNow and in A2Billing and Vicidial or Goautodial that</span><o:p></o:p></pre><pre><span style='color:black'>> whatever I do configuration in the GUI, then the</span><o:p></o:p></pre><pre><span style='color:black'>> configuration will be generated in the conf files, so</span><o:p></o:p></pre><pre><span style='color:black'>> Asterisk will read from the conf files and not from the</span><o:p></o:p></pre><pre><span style='color:black'>> database directly. Is it right or I am confused and there is</span><o:p></o:p></pre><pre><span style='color:black'>> something else?</span><o:p></o:p></pre><pre><span style='color:black'>> ></span><o:p></o:p></pre><pre><span style='color:black'>> > If there is a method to let the configuration to be</span><o:p></o:p></pre><pre><span style='color:black'>> taken from the database (and not from the configuration),</span><o:p></o:p></pre><pre><span style='color:black'>> then HOW? Because even in AsteriskNow, the configuration</span><o:p></o:p></pre><pre><span style='color:black'>> will be generated in a conf files.</span><o:p></o:p></pre><pre><span style='color:black'>> </span><o:p></o:p></pre><pre><span style='color:black'>> Hi Bilal,</span><o:p></o:p></pre><pre><span style='color:black'>> </span><o:p></o:p></pre><pre><span style='color:black'>> You want to look the Asterisk realtime configuration</span><o:p></o:p></pre><pre><span style='color:black'>> features if you </span><o:p></o:p></pre><pre><span style='color:black'>> want to run your configuration from a database rather than</span><o:p></o:p></pre><pre><span style='color:black'>> configuration </span><o:p></o:p></pre><pre><span style='color:black'>> files.</span><o:p></o:p></pre><pre><span style='color:black'>> </span><o:p></o:p></pre><pre><span style='color:black'>> This should point you in the right direction and get you</span><o:p></o:p></pre><pre><span style='color:black'>> started: </span><o:p></o:p></pre><pre><span style='color:black'>> <a href="http://www.voip-info.org/wiki/view/Asterisk+RealTime">http://www.voip-info.org/wiki/view/Asterisk+RealTime</a></span><o:p></o:p></pre><pre><span style='color:black'>> </span><o:p></o:p></pre><pre><span style='color:black'>> It should be noted that if you're wanting to use AsteriskNow</span><o:p></o:p></pre><pre><span style='color:black'>> (which </span><o:p></o:p></pre><pre><span style='color:black'>> relies on FreePBX for its gui configuration features), then</span><o:p></o:p></pre><pre><span style='color:black'>> Asterisk </span><o:p></o:p></pre><pre><span style='color:black'>> realtime configuration will not work as it is not compatible</span><o:p></o:p></pre><pre><span style='color:black'>> at this </span><o:p></o:p></pre><pre><span style='color:black'>> time. Other web gui's might work, but I am not</span><o:p></o:p></pre><pre><span style='color:black'>> familiar with them. </span><o:p></o:p></pre><pre><span style='color:black'>> FreePBX's sentiment on the subject is shared here: </span><o:p></o:p></pre><pre><span style='color:black'>> <a href="http://www.freepbx.org/trac/wiki/AsteriskRealtime">http://www.freepbx.org/trac/wiki/AsteriskRealtime</a></span><o:p></o:p></pre><pre><span style='color:black'>> </span><o:p></o:p></pre><pre><span style='color:black'>> -John</span><o:p></o:p></pre><pre><span style='color:black'>> </span><o:p></o:p></pre><pre><span style='color:black'>> On 05/24/2012 05:46 PM, bilal ghayyad wrote:</span><o:p></o:p></pre><pre><span style='color:black'>> > Thanks for all for the help and kindly reply.</span><o:p></o:p></pre><pre><span style='color:black'>> ></span><o:p></o:p></pre><pre><span style='color:black'>> > One last point that will help me alot:</span><o:p></o:p></pre><pre><span style='color:black'>> ></span><o:p></o:p></pre><pre><span style='color:black'>> > I am thinking to have 4 Servers running Asterisk and 2</span><o:p></o:p></pre><pre><span style='color:black'>> Servers to be for database. The load to be distributed on</span><o:p></o:p></pre><pre><span style='color:black'>> the 4 Asterisk Servers with ability to be redundant (using</span><o:p></o:p></pre><pre><span style='color:black'>> any redundancy technique). The 4 Asterisk Servers to take</span><o:p></o:p></pre><pre><span style='color:black'>> the configuration from the Database Server and actually</span><o:p></o:p></pre><pre><span style='color:black'>> because there is 2 Database servers, then it will be</span><o:p></o:p></pre><pre><span style='color:black'>> redundant to each other (in case one database failed, the</span><o:p></o:p></pre><pre><span style='color:black'>> other will take over).</span><o:p></o:p></pre><pre><span style='color:black'>> ></span><o:p></o:p></pre><pre><span style='color:black'>> > My question is:</span><o:p></o:p></pre><pre><span style='color:black'>> ></span><o:p></o:p></pre><pre><span style='color:black'>> > Is it really possible to have the asterisk</span><o:p></o:p></pre><pre><span style='color:black'>> configuration in the database server instead of having it in</span><o:p></o:p></pre><pre><span style='color:black'>> conf files? HOW? I am asking this because what I noticed in</span><o:p></o:p></pre><pre><span style='color:black'>> AsteriskNow and in A2Billing and Vicidial or Goautodial that</span><o:p></o:p></pre><pre><span style='color:black'>> whatever I do configuration in the GUI, then the</span><o:p></o:p></pre><pre><span style='color:black'>> configuration will be generated in the conf files, so</span><o:p></o:p></pre><pre><span style='color:black'>> Asterisk will read from the conf files and not from the</span><o:p></o:p></pre><pre><span style='color:black'>> database directly. Is it right or I am confused and there is</span><o:p></o:p></pre><pre><span style='color:black'>> something else?</span><o:p></o:p></pre><pre><span style='color:black'>> ></span><o:p></o:p></pre><pre><span style='color:black'>> > If there is a method to let the configuration to be</span><o:p></o:p></pre><pre><span style='color:black'>> taken from the database (and not from the configuration),</span><o:p></o:p></pre><pre><span style='color:black'>> then HOW? Because even in AsteriskNow, the configuration</span><o:p></o:p></pre><pre><span style='color:black'>> will be generated in a conf files.</span><o:p></o:p></pre><pre><span style='color:black'>> ></span><o:p></o:p></pre><pre><span style='color:black'>> > Special thanks for the advise.</span><o:p></o:p></pre><pre><span style='color:black'>> ></span><o:p></o:p></pre><pre><span style='color:black'>> > Regards</span><o:p></o:p></pre><pre><span style='color:black'>> > Bilal</span><o:p></o:p></pre><pre><span style='color:black'>> > -------------</span><o:p></o:p></pre><pre><span style='color:black'>> ></span><o:p></o:p></pre><pre><span style='color:black'>> >>> Hi All;</span><o:p></o:p></pre><pre><span style='color:black'>> >>></span><o:p></o:p></pre><pre><span style='color:black'>> >>> I need to use Asterisk for 20 000 users, so</span><o:p></o:p></pre><pre><span style='color:black'>> which</span><o:p></o:p></pre><pre><span style='color:black'>> >> asterisk version to be used? Is there asterisk</span><o:p></o:p></pre><pre><span style='color:black'>> version that</span><o:p></o:p></pre><pre><span style='color:black'>> >> supports 20,000 users on one hardware machine?</span><o:p></o:p></pre><pre><span style='color:black'>> >>> Can I use one strong hardware server i7 with 64</span><o:p></o:p></pre><pre><span style='color:black'>> GB RAM</span><o:p></o:p></pre><pre><span style='color:black'>> >> and fast hard desk to handle 20 000 users, and</span><o:p></o:p></pre><pre><span style='color:black'>> concurrent</span><o:p></o:p></pre><pre><span style='color:black'>> >> calls 2000? Or I need multiple servers, how much?</span><o:p></o:p></pre><pre><span style='color:black'>> >>> If I am going to use multiple servers (until</span><o:p></o:p></pre><pre><span style='color:black'>> now I do</span><o:p></o:p></pre><pre><span style='color:black'>> >> not know how much, and I do not know if the barrier</span><o:p></o:p></pre><pre><span style='color:black'>> will be</span><o:p></o:p></pre><pre><span style='color:black'>> >> the asterisk software or the hardware), then do I</span><o:p></o:p></pre><pre><span style='color:black'>> have to</span><o:p></o:p></pre><pre><span style='color:black'>> >> use special SIP proxy or I have to use load</span><o:p></o:p></pre><pre><span style='color:black'>> balancer)? In</span><o:p></o:p></pre><pre><span style='color:black'>> >> this case, I have to use asterisk Database (so all</span><o:p></o:p></pre><pre><span style='color:black'>> the</span><o:p></o:p></pre><pre><span style='color:black'>> >> servers will read/write from the database)?</span><o:p></o:p></pre><pre><span style='color:black'>> >>> What about AsteriskNow, can it support?</span><o:p></o:p></pre><pre><span style='color:black'>> >> AsteriskNOW is a GUI on top of Asterisk; it does</span><o:p></o:p></pre><pre><span style='color:black'>> not change</span><o:p></o:p></pre><pre><span style='color:black'>> >> the ability</span><o:p></o:p></pre><pre><span style='color:black'>> >> of the system to handle call load.</span><o:p></o:p></pre><pre><span style='color:black'>> >></span><o:p></o:p></pre><pre><span style='color:black'>> >> Modern versions of Asterisk can easily handle</span><o:p></o:p></pre><pre><span style='color:black'>> 2,000</span><o:p></o:p></pre><pre><span style='color:black'>> >> simultaneous calls,</span><o:p></o:p></pre><pre><span style='color:black'>> >> even with media (non-transcoded) passing through</span><o:p></o:p></pre><pre><span style='color:black'>> the server.</span><o:p></o:p></pre><pre><span style='color:black'>> >> We have a</span><o:p></o:p></pre><pre><span style='color:black'>> >> community member who has improved chan_sip in</span><o:p></o:p></pre><pre><span style='color:black'>> Asterisk 10</span><o:p></o:p></pre><pre><span style='color:black'>> >> (and later) to</span><o:p></o:p></pre><pre><span style='color:black'>> >> be able to handle 10,000 simultaneous calls.</span><o:p></o:p></pre><pre><span style='color:black'>> >></span><o:p></o:p></pre><pre><span style='color:black'>> >> Handling 20,000 registrations is probably more of a</span><o:p></o:p></pre><pre><span style='color:black'>> concern</span><o:p></o:p></pre><pre><span style='color:black'>> >> for Asterisk</span><o:p></o:p></pre><pre><span style='color:black'>> >> at this point; I've never heard of anyone</span><o:p></o:p></pre><pre><span style='color:black'>> attempting to</span><o:p></o:p></pre><pre><span style='color:black'>> >> handle that many</span><o:p></o:p></pre><pre><span style='color:black'>> >> on one system.</span><o:p></o:p></pre><pre><span style='color:black'>> >></span><o:p></o:p></pre><pre><span style='color:black'>> >> In spite of all this, though, the other advice</span><o:p></o:p></pre><pre><span style='color:black'>> you've</span><o:p></o:p></pre><pre><span style='color:black'>> >> received in this</span><o:p></o:p></pre><pre><span style='color:black'>> >> thread is sound: even if a single system can handle</span><o:p></o:p></pre><pre><span style='color:black'>> the</span><o:p></o:p></pre><pre><span style='color:black'>> >> load, doing so</span><o:p></o:p></pre><pre><span style='color:black'>> >> is asking for a major problem if that system</span><o:p></o:p></pre><pre><span style='color:black'>> experiences a</span><o:p></o:p></pre><pre><span style='color:black'>> >> failure.</span><o:p></o:p></pre><pre><span style='color:black'>> >> You'd be much better off to at least split the load</span><o:p></o:p></pre><pre><span style='color:black'>> across</span><o:p></o:p></pre><pre><span style='color:black'>> >> two machines,</span><o:p></o:p></pre><pre><span style='color:black'>> >> both of which should be large enough to handle the</span><o:p></o:p></pre><pre><span style='color:black'>> entire</span><o:p></o:p></pre><pre><span style='color:black'>> >> load when</span><o:p></o:p></pre><pre><span style='color:black'>> >> necessary.</span><o:p></o:p></pre><pre><span style='color:black'>> >></span><o:p></o:p></pre><pre><span style='color:black'>> >> -- </span><o:p></o:p></pre><pre><span style='color:black'>> >> Kevin P. Fleming</span><o:p></o:p></pre><pre><span style='color:black'>> >> Digium, Inc. | Director of Software Technologies</span><o:p></o:p></pre><pre><span style='color:black'>> >> Jabber: <a href="mailto:kfleming@digium.com">kfleming@digium.com</a></span><o:p></o:p></pre><pre><span style='color:black'>> >> | SIP: <a href="mailto:kpfleming@digium.com">kpfleming@digium.com</a></span><o:p></o:p></pre><pre><span style='color:black'>> >> | Skype: kpfleming</span><o:p></o:p></pre><pre><span style='color:black'>> >> 445 Jan Davis Drive NW - Huntsville, AL 35806 -</span><o:p></o:p></pre><pre><span style='color:black'>> USA</span><o:p></o:p></pre><pre><span style='color:black'>> >> Check us out at <a href="http://www.digium.com">www.digium.com</a>& </span><o:p></o:p></pre><pre><span style='color:black'>> <a href="http://www.asterisk.org">www.asterisk.org</a></span><o:p></o:p></pre><pre><o:p> </o:p></pre><pre><span style='color:black'>--</span><o:p></o:p></pre><pre><span style='color:black'>_____________________________________________________________________</span><o:p></o:p></pre><pre><span style='color:black'>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com">http://www.api-digital.com</a> --</span><o:p></o:p></pre><pre><span style='color:black'>New to Asterisk? Join us for a live introductory webinar every Thurs:</span><o:p></o:p></pre><pre><span style='color:black'> <a href="http://www.asterisk.org/hello">http://www.asterisk.org/hello</a></span><o:p></o:p></pre><pre><o:p> </o:p></pre><pre><span style='color:black'>asterisk-users mailing list</span><o:p></o:p></pre><pre><span style='color:black'>To UNSUBSCRIBE or update options visit:</span><o:p></o:p></pre><pre><span style='color:black'> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a></span><o:p></o:p></pre><p class=MsoNormal style='margin-bottom:12.0pt'><o:p> </o:p></p></div></body></html>