[asterisk-users] SIPP problem

[Tzafrir Cohen]

On Sun, Sep 03, 2006 at 10:03:32AM -0500, Diego Quintana Cruz wrote:
> 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?

Do you use sipp as a standaalone service, or do you also need an
Asterisk to originate calls? If the former, An Asterisk installation is
not really required and shouldn't matter, anyway.

-- 
Tzafrir Cohen         sip:tzafrir at local.xorcom.com
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