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

Tue Nov 3 21:37:59 PST 2009

Matthew, 

I'm about in the same boat as you are, just on a smaller scale. We have a
ton of Nortel telephony gear, but it's time to move out of the 90's and
enter this millennium. My Cisco quote was in the same ballpark as yours. 

The Cisco stuff is mature, rock solid, meshes very well with their network
gear and is actually relatively easy to set up and maintain if you know your
way around IOS. I just refuse to pay that kind of money for yet another
semi-proprietary solution.

After looking at various asterisk distributions, SipX, 3CX and
what-have-you, I've come to the conclusion that FreeSWITCH is by far the
most advanced platform out there. Its architecture and performance is
literally light years ahead of the rest and I have yet to come up with
something that it can't do. But all that comes at a price: The learning
curve is like scaling a brick wall. The developers and the community are
great and available, but just starting out with SIP and voip in general,
this may not be the best platform. So let the blasphemy begin :)

SipX was a breeze to install (insert CD, boot, next next next...) and looks
pretty solid. I believe they actually use FreeSWITCH for their voicemail and
conferencing, internally. I just couldn't get my head around their GUI, ACD
was too basic and had all kinds of issues getting stuff to "just work".

3CX (Windows Only) was completely painless. It just worked. But I'm still
not convinced that I want to run all my voice on a single windows box. Plus
it's not free/open/etc and I don't want to lock myself in again.

Although it's an asterisk based solution, I found trixbox to be very easy.
Setup is automatic and everything "just worked". The GUI is simple and
logical enough that I can let somebody else handle the day-to-day phone
setup and basic admin. I have my doubts about it scaling to 250 users,
though.

This may be a completely flawed strategy and I may very well be shooting
myself in the foot by doing this, but I plan on piloting a trixbox install
with a dozen or so users and see how stable it is. I'll keep a FreeSWITCH
box next to it for the more advanced stuff. Once I get more comfortable with
the intricacies of SIP and get some time to code a basic GUI for FreeSWITCH,
I have a feeling that that trixbox is going to get phased out...

Peter

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
mkitchin.public at gmail.com
Sent: Tuesday, November 03, 2009 11:10 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Newbie trying to setup Cisco 7940 phones

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.

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