[Asterisk-Users] ASTERISK AND 120 CONCURRENT CALLS

[Sebastian Nocetti]

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 at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com] En nombre de mattf
Enviado el: Viernes, 06 de Agosto de 2004 04:07 p.m.
Para: 'asterisk-users at 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 at fibertel.com.ar]
Sent: Friday, August 06, 2004 2:51 PM
To: asterisk-users at 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 at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com] En nombre de mattf Enviado
el: Viernes, 06 de Agosto de 2004 03:44 p.m.
Para: 'asterisk-users at 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 at interband.com.ar]
Sent: Friday, August 06, 2004 12:53 PM
To: asterisk-users at 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.
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