[asterisk-users] SIPP problem

[Diego Quintana Cruz]

2006/9/2, Greg Boehnlein <damin at nacs.net>:
> On Sat, 2 Sep 2006, Diego Quintana Cruz wrote:
>
> > Hi everybody,
> > I'm trying to load-test my Asterisk PBX using SIPP, but I always
> > getting errors, I followed the instructions given in [1] which mainly
> > was to create the user sipp in sip.conf and the dialing plan for his
> > context in extensions.conf
> >
> > I'm using Asterisk 1.0.10
> >
> > Any ideas or tutorial on how using SIP?
>
>
> Here are my notes on the subject:
>
> http://lists.digium.com/pipermail/asterisk-dev/2006-June/021162.html

I did what you have there but I'm always getting 503 Service
unavailable, I don't know why.

I'm also using AMPortal, do I have to configure something there?

Regards, and sorry for my bad english
-- 
Diego Quintana a.k.a. RouterMaN
Ingeniería de las Telecomunicaciones
PUCP
Linux Registered User #382615 - http://counter.li.org/
SIP # 1-747-633-6676 Ext. 1011
FWD # 764839 Ext. 1011
http://routerman.blogsome.com
http://planeta.debianperu.org