Michael Collins wrote: > > > On Tue, Nov 3, 2009 at 2:19 PM, mkitchin.pub...@gmail.com > <mailto:mkitchin.pub...@gmail.com> <mkitchin.pub...@gmail.com > <mailto:mkitchin.pub...@gmail.com>> wrote: > > I'm working on an alternative to a $120,000 Cisco phone system that my > > company is looking at. I got Freeswitch installed on CentOS last week > using the Quick and Dirty instructions. That part was painless. We > had a > few 7940s laying around. After some wrestling with it, I got the > latest > SIP firmware installed and what I hoped was a functional config > (attached). X-Lite phones can call each other no problem. 7940s > can call > X-Lite no problem. Anytime I try and call a 7940, it goes straight to > voicemail. I attached a log file that shows the activity when > trying to > call a7940 from X-Lite. > X-Lite is at 10.86.10.58. 7940 is at 10.86.11.50. Freeswitch is > nshplpbx1.unix/10.85.0.53 <http://10.85.0.53>. Everything is on > the same LAN. Different > subnets, but no firewalls. > I didn't see anything that said posting attachments was frowned > upon. I > apologize if it isn't appropriate. I'm guessing this is something > simple > and I'm just clueless on how to diagnose the issue. > I'm not tied to using this model for good, but it is what we had > laying > around. Any help would be greatly appreciated. Next step is > configuring > it to talk to Verizon VOIP over a DS3. > > Thanks, > Matthew Kitchin > > > Matthew, > 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. > > Here's a handy wiki page that will help you get the diagnosing skills > you need: > http://wiki.freeswitch.org/wiki/Reporting_Bugs > > 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. > (http://pastebin.freeswitch.org) > > You can also join the IRC channel #freeswitch on irc.freenode.net > <http://irc.freenode.net> and get some real-time help. There are some > sharp folks in there, not the least of which are the three main > FreeSWITCH developers. > > -MC Thank you. I think I did what you are looking for. I stopped FS and launched this command. TPORT_LOG=1 /usr/local/freeswitch/bin/freeswitch and captured all output to http://pastebin.freeswitch.org/10965 Does this tell you anything? I'm definitely new to SIP and phone system admin in general. I have plenty of network and Linux experience. With that in mind, someone on this mailing list emailed me directly and said SipX would be a better fit for me. Is that blasphemy for me to even mention? I went through the documentation and the provisioning aspect and web interface do look tempting to a novice. I apologize if this is like trying to buy a chevy at a ford dealership. I'm looking to deploy about 150 handsets at a corporate office and then 10 to 12 handsets at 120 remote locations. We are moving from an old key system, so our current features are very limited. We just need a few ACD groups, call history, and the other general basics. I first found Asterisk and read about some of the shortcomings. FS looks like the most robust solution. I have no idea where SipX would fit in. The people here are obviously a very knowledgeable group and I would gladly accept any thoughts, comments, etc.
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