<br><br><div class="gmail_quote">On Tue, Nov 3, 2009 at 2:19 PM, <a href="mailto:mkitchin.public@gmail.com">mkitchin.public@gmail.com</a> <span dir="ltr"><<a href="mailto:mkitchin.public@gmail.com">mkitchin.public@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">I'm working on an alternative to a $120,000 Cisco phone system that my<br>
<br>
company is looking at. I got Freeswitch installed on CentOS last week<br>
using the Quick and Dirty instructions. That part was painless. We had a<br>
few 7940s laying around. After some wrestling with it, I got the latest<br>
SIP firmware installed and what I hoped was a functional config<br>
(attached). X-Lite phones can call each other no problem. 7940s can call<br>
X-Lite no problem. Anytime I try and call a 7940, it goes straight to<br>
voicemail. I attached a log file that shows the activity when trying to<br>
call a7940 from X-Lite.<br>
X-Lite is at 10.86.10.58. 7940 is at 10.86.11.50. Freeswitch is<br>
nshplpbx1.unix/<a href="http://10.85.0.53" target="_blank">10.85.0.53</a>. Everything is on the same LAN. Different<br>
subnets, but no firewalls.<br>
I didn't see anything that said posting attachments was frowned upon. I<br>
apologize if it isn't appropriate. I'm guessing this is something simple<br>
and I'm just clueless on how to diagnose the issue.<br>
I'm not tied to using this model for good, but it is what we had laying<br>
around. Any help would be greatly appreciated. Next step is configuring<br>
it to talk to Verizon VOIP over a DS3.<br>
<br>
Thanks,<br>
Matthew Kitchin<br>
<br></blockquote><div><br>Matthew,<br>Welcome to FreeSWITCH! We're glad you're ditching a $120K system. We think you'll find FS is as powerful as any software out there right now.<br><br>Here's a handy wiki page that will help you get the diagnosing skills you need:<br>
<a href="http://wiki.freeswitch.org/wiki/Reporting_Bugs">http://wiki.freeswitch.org/wiki/Reporting_Bugs</a><br><br>I'd say first thing to do is capture the SIP traffic to see if there are any clues. A "normal temporary failure" doesn't give you a lot of detail. :) If you're new to SIP debugging then the best thing to do is to capture the SIP trace and put it in the pastebin. (<a href="http://pastebin.freeswitch.org">http://pastebin.freeswitch.org</a>)<br>
<br>You can also join the IRC channel #freeswitch on <a href="http://irc.freenode.net">irc.freenode.net</a> and get some real-time help. There are some sharp folks in there, not the least of which are the three main FreeSWITCH developers. <br>
<br>-MC<br></div></div><br>