[Freeswitch-users] Newbie trying to setup Cisco 7940 phones
Shelby Ramsey
sicfslist at gmail.com
Wed Nov 4 04:08:06 PST 2009
Peter,
Did you look at ? Probably just what you are
looking for. GUI goodness based on FS.
SDR
Peter J. Zandvoort wrote:
> 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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