Tim and Joseph make good points. You start with the default configuration and then remove what you do not need. Frankly, the default configuration is just fine for you. FS won't "hijack" your VoIP infrastructure so don't sweat that. It's simply a server sitting on your network waiting for calls.<br>
<br>My recommendation would be to do a basic install (see the wiki) and then make one addition to the public.xml dialplan file. Add this simple extension before the closing </context> tag:<br><br><span style="font-family: courier new,monospace;"> <extension name="Four-digit extensions"></span><br style="font-family: courier new,monospace;">
<span style="font-family: courier new,monospace;"> <condition field="destination_number" expression="^(\d\d\d\d)$"></span><br style="font-family: courier new,monospace;"><span style="font-family: courier new,monospace;"> <action application="transfer" data="$1 XML default"/></span><br style="font-family: courier new,monospace;">
<span style="font-family: courier new,monospace;"> </condition></span><br style="font-family: courier new,monospace;"><span style="font-family: courier new,monospace;"> </extension></span><br style="font-family: courier new,monospace;">
<br>Press F6 or type "reloadxml" at the FreeSWITCH command line. Now if you throw anonymous calls at this server it will handle four-digit extensions. For example, if you call sip:9999@x.x.x.x:5060 then you should hear music on hold. Call sip:5000@x.x.x.x:5060 and you should hear the demo IVR. (x.x.x.x = FS server IP address) This will let you test a lot of FS features without having to register phones to your FS server, although you are certainly free to do that.<br>
<br>Be sure to read the newbie article that I wrote for Linux-Pro magazine last year: <a href="http://bit.ly/EpVrv">http://bit.ly/EpVrv</a><br><br>It will help you get up and running quickly. (Note: we switched to git from svn, so we recommend you use git to download the source...)<br>
<br>Have fun!<br>-MC<br><br><div class="gmail_quote">On Wed, Jun 9, 2010 at 7:43 AM, Tim St. Pierre <span dir="ltr"><<a href="mailto:fs-list@communicatefreely.net">fs-list@communicatefreely.net</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">Hi Oliver,<br>
<br>
FreeSwitch will do whatever you tell it to do and no more.<br>
<br>
Here's a few suggestions though -<br>
<br>
Empty out the default dialplan directory. Don't throw those away, as you may want to reference them<br>
as examples, but move them somewhere else.<br>
<br>
Edit modules.conf.xml and comment or remove any modules that you don't need. This will also save<br>
memory and other resources.<br>
<br>
You can also disable all the SIP profiles except one. Pick one that makes sense (either Internal or<br>
External, it doesn't really matter that much), and edit it so that it makes sense with respect to<br>
your network. What is your topology? Will you just be setting freeswitch up with a static IP<br>
address and having calls sent to it by the main PBX? If that's the case, you can disable a lot of<br>
the STUN and uPNP functionality. Tell this profile to bind to the IP and port that the PBX will<br>
send the calls to.<br>
<br>
Then all you have to do, is create a very simple dialplan that will answer an incoming call and<br>
perform whatever task you want. You would essentially be starting with a blank sheet, adding just<br>
the functions that you want.<br>
<br>
Hope that makes sense.<br>
<font color="#888888"><br>
-Tim<br>
</font><div><div></div><div class="h5"><br>
Oliver Schenk wrote:<br>
> Hi All,<br>
><br>
> The company I work for currently has quite an extensive phone network<br>
> which gets carried between old analogue PABXes which also has an<br>
> interface to IP based phones. All the phones in our office are connected<br>
> via CAT5 cable using IP, however literally hundreds of phones out in the<br>
> field (we operate railway infrastructure) are on standard voice analogue<br>
> phones carried through fibreoptics.<br>
><br>
> Anyway, I would like to use Freeswitch purely for its IVR and TTS<br>
> abilities and nothing else. So basically I just need it to act like a<br>
> slave to whatever IP phone network is already out there. I'm a bit<br>
> worried if I fire up freeswitch it will hijack the phone network!<br>
><br>
> All our phones are accessible via a 5 digit extension. I would like<br>
> Freeswitch to be behind one of those ... say 12345. If anyone within our<br>
> phone network dials 12345 then Freeswitch should answer. I guess my<br>
> question is...how should I go about disabling most of FreeSwitch except<br>
> it's ability to pick up the phone and speak IVR/TTS and make an outgoing<br>
> call via the existing phone network?<br>
><br>
> Any general pointers appreciated.<br>
><br>
><br>
> Thanks,<br>
><br>
> Oliver Schenk<br>
><br>
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