2006/9/2, Greg Boehnlein <[EMAIL PROTECTED]>:
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
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