Sebastian Nocetti
2004-Aug-06 09:53 UTC
[Asterisk-Users] ASTERISK AND 120 CONCURRENT CALLS
hello all, does anyone has experiencie using asterisk with a digium CARD using G729 managing 120 concurrent calls with SIP and/or H323??? I wanna know if Asterisk is stable doing this....because we wanna implement it in some locations!! Thanks All!! Sebastian. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040806/e868d188/attachment.htm
Will you have E1s? will you restrict users to 729 or will you allow other codecs? will most calls be from SIP to SIP? or SIP to E1 lines? MATT--- -----Original Message----- From: Sebastian Nocetti [mailto:sebastian@interband.com.ar] Sent: Friday, August 06, 2004 12:53 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] ASTERISK AND 120 CONCURRENT CALLS hello all, does anyone has experiencie using asterisk with a digium CARD using G729 managing 120 concurrent calls with SIP and/or H323??? I wanna know if Asterisk is stable doing this....because we wanna implement it in some locations!! Thanks All!! Sebastian. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040806/e2a1ced1/attachment.htm
Nope, you won't be able to build a server fast enough to handle the transcoding. At the very most we've handled 60 concurrent SIP to T1 conversations on a Dual Athlon MP 2800+ system before it crashed, and I've never heard of anyone having more than 90 concurrent SIP to Zap channels running (and that was in a lab envorinment). If you want to use Asterisk you should look into multiple, fast asterisk servers handling 50 concurrent calls at the most each. MATT--- -----Original Message----- From: Sebastian Nocetti [mailto:snocetti@fibertel.com.ar] Sent: Friday, August 06, 2004 2:51 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] ASTERISK AND 120 CONCURRENT CALLS E1's, only G729 and from SIP to E1 or from E1 to SIP De: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] En nombre de mattf Enviado el: Viernes, 06 de Agosto de 2004 03:44 p.m. Para: 'asterisk-users@lists.digium.com' Asunto: RE: [Asterisk-Users] ASTERISK AND 120 CONCURRENT CALLS Will you have E1s? will you restrict users to 729 or will you allow other codecs? will most calls be from SIP to SIP? or SIP to E1 lines? MATT--- -----Original Message----- From: Sebastian Nocetti [mailto:sebastian@interband.com.ar] Sent: Friday, August 06, 2004 12:53 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] ASTERISK AND 120 CONCURRENT CALLS hello all, does anyone has experiencie using asterisk with a digium CARD using G729 managing 120 concurrent calls with SIP and/or H323??? I wanna know if Asterisk is stable doing this....because we wanna implement it in some locations!! Thanks All!! Sebastian.
If you do testing before you go live I'd love to see how many concurrent calls you get out of that very expensive HP server :) MATT--- -----Original Message----- From: Sebastian Nocetti [mailto:snocetti@fibertel.com.ar] Sent: Friday, August 06, 2004 3:24 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] ASTERISK AND 120 CONCURRENT CALLS Mm, that's bad, wow... Well, I will see howto implement that... Thanks for you comments Aaa... My macbhine is a DUAL XEON 3.4 with 2GB memory Is a HP PROLIANT. -----Mensaje original----- De: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] En nombre de mattf Enviado el: Viernes, 06 de Agosto de 2004 04:07 p.m. Para: 'asterisk-users@lists.digium.com' Asunto: RE: [Asterisk-Users] ASTERISK AND 120 CONCURRENT CALLS Nope, you won't be able to build a server fast enough to handle the transcoding. At the very most we've handled 60 concurrent SIP to T1 conversations on a Dual Athlon MP 2800+ system before it crashed, and I've never heard of anyone having more than 90 concurrent SIP to Zap channels running (and that was in a lab envorinment). If you want to use Asterisk you should look into multiple, fast asterisk servers handling 50 concurrent calls at the most each. MATT--- -----Original Message----- From: Sebastian Nocetti [mailto:snocetti@fibertel.com.ar] Sent: Friday, August 06, 2004 2:51 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] ASTERISK AND 120 CONCURRENT CALLS E1's, only G729 and from SIP to E1 or from E1 to SIP De: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] En nombre de mattf Enviado el: Viernes, 06 de Agosto de 2004 03:44 p.m. Para: 'asterisk-users@lists.digium.com' Asunto: RE: [Asterisk-Users] ASTERISK AND 120 CONCURRENT CALLS Will you have E1s? will you restrict users to 729 or will you allow other codecs? will most calls be from SIP to SIP? or SIP to E1 lines? MATT--- -----Original Message----- From: Sebastian Nocetti [mailto:sebastian@interband.com.ar] Sent: Friday, August 06, 2004 12:53 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] ASTERISK AND 120 CONCURRENT CALLS hello all, does anyone has experiencie using asterisk with a digium CARD using G729 managing 120 concurrent calls with SIP and/or H323??? I wanna know if Asterisk is stable doing this....because we wanna implement it in some locations!! Thanks All!! Sebastian. _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users