My company is adding VoIP to our existing satellite network system. We will be using AudioCodes FXS at the remote satellite sites and AudioCodes FXO or Mediant at the hub site (which connects to the PSTN). We’ve previously had the FXO/FXS network working properly, but now we need to expand to using Asterisk to handling the routing, because we have too many remote (FXS) sites for the FXO/Mediant to manage.
I’ve compiled and installed Asterisk 1.2 on a linux 2.6 machine, and have modified the FXO/FXS to use Asterisk as the Proxy, and indeed when I try to place an outgoing from an FXS, I see packets being routed to the Asterisk machine. So far all good!!
However, I’m overwhelmed when it comes to what I need to do to configure Asterisk properly… I’ve read the relevant chapters of “Asterisk - the Future of Telephony” (hereafter referred to as TFOT), looked at extensions.conf and sip.conf, and looked at the Starting Out page on voip-info, but I’m lost regarding the basic things I need to configure to just place an incoming and outgoing call.
Frankly, the Starting Out page just makes my eyes glaze over!! almost three screens of links to other pages, and none of the entries is clearly what I’m looking for. Some of the questions that I need to answer are:
How, specifically, do I route an incoming call to a remote FXS?? Presumably I want to use an ‘exten’ entry in extensions.conf, but there isn’t a route() function… so the obvious
exten => 703*,1,Route(172.18.10.23)
to get to the FXS (at 172.18.10.23) whose extensions are 70301-70324, doesn’t work like that - how does it work??
Similarly with outgoing calls; with the FXO/FXS environment, I told the FXS to Route 802xx to 172.18.100.23 (the FXO IP address), when when it called 80201, it indeed got a dialtone. How does that work via Asterisk??
What is Registering (from the FXO/FXS standpoint, and from Asterisk’s standpoint) ?? do I need to do that to make this work?? Does Asterisk register with the FXO/FXS, or they with it??
Relative to the Asterisk server, are the FXO and FXS(s) all peers?? users?? neither?? TFOT says “users place calls to us, while we place calls to our peers”… but I can look at this in different ways - to the user, they are never calling Asterisk, they’re calling the FXO/FXS. To the FXO/FXS, I guess it looks like they are always calling the server, so they are users… good… but wait!! The server will have to call the FXO to reach the PSTN, so it’s a peer!! or is it?? I’m very confused.
I’m sure these questions have been asked a thousand times before by other users, so if anyone wishes to just respond by posting links to other relevant conversations, that’s fine. Otherwise, any simple, clear clarification would be gratefully accepted!!! In the meantime, I’ll keep searching these forums, and I’ll try reading some of the voip-info tutorials to look for insight.