On Sun, Sep 03, 2006 at 10:03:32AM -0500, Diego Quintana Cruz wrote: > 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?
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:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
