Good morning all, I'm setting up Asterisk for the first time with no prior PBX experience. I'm following Andy Powell's 'Getting Started with Asterisk' (http://www.automated.it/guidetoasterisk.htm). This is my second time through that document - as I did something weird the first time and really upset it somehow - and I wanted to ask a few general questions of the list.
First, a little on what I'm trying to do: I need to setup the PBX to answer on multiple 'lines' (I use that word with trepidation as I'm not sure if it's the right term in the absence of modems & actual lines) and play a brief message identifying itself as the 'line' connected to. The originator of that call will be a softphone. Before rolling this out to my lab, I'm trying to work out the proper config on my laptop. Therein I have Windows XP w/ VMware - Red Hat 7.3 is running in a VMware session. My connection to the net is NAT'd. My internal IPs for both XP & Linux are of the 192.168.x.x private network variety. I have the Xten X-Lite softphone on XP to test with. (I also have another called SJPhone, but haven't done much with that past installing it.) I've configured a number for that through freeworlddialup.com. X-Lite appears to be working fine. At least I can dial their echo & test numbers without a problem and get the expected responses. So the questions: 1. A general "will this work?" (vmware linux, same pc as phone, NAT'd addresses,etc) 2. Has anyone done it this way before and/or followed Andy Powell's doc, and have any suggestions or things to watch out for? 3. Reading the various published SIP documentation (Ubiquity's 'Understanding SIP' for instance), it seems like freeworlddialup is acting as Registrar, Proxy & Redirect server. Is that accurate? 4. How do I tell the freeworlddialup registrar 'where' to find my PBX? Should I setup an account from it - like I did with the softphone on XP - so it will have a 'phone number' of its own? Or is the proxy/redirect server expecting to talk to the Asterisk PBX in some other way? I appreciate any and all responses. Please cc my email address directly on replies as I have the list configured in digest mode to stem the flow a bit and don't want to miss any of them in the mix. Thanks to one and all Fletcher Bonds Operations Software Tester TeleCommunication Systems, Inc. (TCS) Enabling Convergent Technologies www.telecomsys.com [EMAIL PROTECTED] _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
