[asterisk-users] ASTERISK y AGC

Angelito Manansala avmanansala at gmail.com
Fri Dec 8 00:45:08 MST 2006


IN ENGLISH VERSION:

Good night I have mounted the system of predictive marker ASTGUICLIENT in 2
Servants, in one of them who are a Server HP Proliant G3 350 3GB ram with 11
Slackware and Asterisk 1.2.12.1, this single server is in charge of the
voice. Soon another server a little but modest (HP ML110 3.2GB), that has
Apache and the BD MySQL with ASTGUICLIENT 2.0.1, my agents is connected to
this I complete q already the system draws out there, softphone Eyebeam of
my agents estan connected to server VOIP, I have in addition in the server
voip a card digium to 2 installed ports 2 lines ISDN, In the ASTGUICLIENT I
have several campaigns or working, which remove to calls through lines ISDN,
single one of these campaigns Extraction the calls through a main IAX or SIP
that I have with my branch of another country or my supplier of voice. I
have tip maximum of 25 agents connected, the problem this in which arrives a
little while in which the marker no longer passes calls to the agents,
rather is delayed too much in marking and to pass the call., If one marks
from softphone (eyebeam) in pantallla of same I obtain message TRYNG. And
after several seconds it manages to remove the call, in the case of my
agents q work with the marker, hope by a long time q pass the calls to him.
To that it must east problem, the load of my servants is not much not to
pass of 5% in his load average? I have updated the version of asterisk that
tapeworm and of UNDER thinking q podria to solve but with the new versions
the problem even appears? Sera q with 2 servants even sharing the load
cannot put but of 20 agents to the system? Previously I had 20 agents
everything in a servant, east tapeworm problem but the system was not very
slow and towards very difficult the work thus decidi to divide to him to the
load to the servant clearing to him the BD and the WEB to happen to him to
another servant. Some of you has had east problem, of q forms have solved it
Thanks beforehand for its answers Greetings.

On 12/8/06, Aldo Alexander Leyva Alvarado <aleyva2004 at gmail.com> wrote:
>
> Buenas noches
> Tengo montado el sistema de marcador predictivo ASTGUICLIENT en 2
> Servidores, en uno de ellos que es un Server HP Proliant G3 350 3GB RAM con
> Slackware 11 y Asterisk 1.2.12.1 , dicho server solo se encarga de la voz.
>
> Luego otro server un poco mas modesto (HP ML110 3.2GB), que tiene Apache y
> la BD MySQL con el ASTGUICLIENT 2.0.1, mis agentes se conectan a este
> ultimo ya q el sistema corree alli, los softphone Eyebeam de mis agentes
> estan conectados al server VOIP,
>
> Tengo ademas en el server voip una tarjeta digium de 2 puertos instalados
> 2 lineas ISDN,
> En el ASTGUICLIENT tengo varias campañas ya trabajando, las cuales sacan
> llamadas a traves de las lineas ISDN, solo una de estas campañas Saca las
> llamadas a traves de una troncal IAX o SIP que tengo con mi filial de otro
> pais o mi proveedor de voz.
> Tengo un pico maximo de 25 agentes conectados, el problema esta en que
> llega un momento en que el marcador ya no pasa llamadas a los agentes, mejor
> dicho se demora demasiado en marcar y pasar la llamada.,Si uno marca desde
> un softphone  (eyebeam) en la pantallla de mismo obtengo el mensaje TRYNG
> ..Y despues de varios segundos logra sacar la llamada, en el caso de mi
> agentes q trabajan con el marcador, esperan por un largo tiempo q le pasen
> las llamadas.
> A que se debe este problema, la carga de mis servidores no es mucha no
> pasar de 5% en su load average?
> He actualizado la version del asterisk que tenia y del SO pensando q lo
> podria solucionar pero aun con las nuevas versiones el problema se presenta?
>
> Sera q aun con 2 servidores compartiendo la carga no pueda meter mas de 20
> agentes al sistema?
> Anteriormente tuve 20 agentes todo en un servidor, no tenia este problema
> pero el sistema estaba muy lento y  hacia muy dificil el trabajo por lo cual
> decidi dividirle la carga al servidor quitandole la BD y el WEB para pasarle
> a otro servidor.
>
> Alguno de ustedes ha tenido este problema, de q forma lo han solucionado
>
>
> Gracias de antemano por sus respuestas
>
> Saludos
> Aldo Leyva
>
>
>
>
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-- 
Lito Manansala
www.voicefidelity.net
Mobile: +63.906.437.0459
PSTN: +63.44.790.6292
sip:7001 at 203.65.246.100
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skype: bulcrack
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