Check out FreeSwitch to replace Asterisk in your core. On Wed, Feb 18, 2009 at 3:42 AM, michel freiha <[email protected]> wrote:
> Dear Alex, > > Thanks for the reply..Can you please list some of these solutions that you > talked about on your reply? > Even I would like to ask if you had a bad experience with asterisk > regarding simultaneous calls limitation and If I'll send 1k calls to an > asterisk machine with the appropriate hardware what will happen? > Kindly note that no trans coding is done, just pass thru codec > > Regards > > > On Tue, Feb 17, 2009 at 5:34 PM, Alex Balashov > <[email protected]>wrote: > >> No, asterisk on conventional hardware can handle at most a few hundred >> calls. >> >> I would strongly discourage the use of Asterisk purely as a transit >> element for billing. Just because a2billing is available does not mean >> you should. Far more scalable solutions are easily available. >> >> -- >> Sent from mobile device >> >> On Feb 17, 2009, at 10:19 AM, michel freiha <[email protected]> wrote: >> >> > Hi all, >> > >> > I'm planning to build a VOIP solution for handling SIP calls coming >> > from endpoints registered on a specific SIP proxy...I made some >> > research regarding network architecture and found out that the best >> > solution is to use OpenSips as SIP proxy for registration and local >> > calls between registered endpoints and use asterisk server with >> > a2billing for PSTN calls, rating, routing and all other stuff plus a >> > MySQL database... >> > >> > This architecture convinced me, but I have some questions regarding >> > asterisk and I need asterisk expert answers in order to take >> > decision... >> > >> > 1- Is there any Software limitation on asterisk regarding number of >> > simulltaneous calls? >> > 2- Can 1 asterisk server handle 5000 simuitaneous calls if I have >> > the appropriate hardware? >> > 3- It's etter to have one asterisk server for hadling 5k >> > simultaneous calls or divide the load on different servers? >> > >> > >> > Waiting your reply >> > >> > Regards >> > _______________________________________________ >> > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> > >> > asterisk-users mailing list >> > To UNSUBSCRIBE or update options visit: >> > http://lists.digium.com/mailman/listinfo/asterisk-users >> >> _______________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype)
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