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? [1] http://www.rowetel.com/ucasterisk/ucasterisk.html Regards, -- Diego Quintana a.k.a. RouterMaN IngenierĂa 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 _______________________________________________ --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
