Hello, I've installed asterisk 1.4.23 on a hosted Xen guest (CentOS 5) and someone with more exposure to Asterisk than me managed to get a basic echo test to work on his office desktop. I also bought credit for termination with grnvoip.com, who also sent me some Asterisk config snippets.
But that's about it. My firend didn't know how to setup the dial plans to connect the incoming registered sip client to them, and all the documentation I managed to find in the last couple of weeks seems to assume that the reader somehow was born with understanding of the basics and the terminology. I also have troubles setting up a SIP client to access Asterisk, not the least because it appears that most of them don't cooperate with pulse audio (http://ultrahigh.org/2008/05/08/voice-over-ip-on-hardy/, I know it's about 8.04 and I have 8.10 on my desktops but still I get similar or worse results). Twinkle was recommended to me as a good client for SIP debugging. I'm no stranger to technical docs (e.g. learned the TCP/IP protocol stack from the RFC's long before Stevens cam out with the books which cover this) but I keep getting into loops of trying to understand where to begin and how to extrapolate from the examples I find to the specific information I got from grnvoip and my specific server. Would someone here be willing to sit with me for an hour and show me around how to get a basic Asterisk config working? Thanks very much in advance, --Amos -- SLUG - Sydney Linux User's Group Mailing List - http://slug.org.au/ Subscription info and FAQs: http://slug.org.au/faq/mailinglists.html
