[Freeswitch-users] Newbie trying to setup Cisco 7940 phones

[mkitchin.public at gmail.com]

Michael Collins wrote:
>
>
> On Tue, Nov 3, 2009 at 2:19 PM, mkitchin.public at gmail.com 
> <mailto:mkitchin.public at gmail.com> <mkitchin.public at gmail.com 
> <mailto:mkitchin.public at 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.