ecide what fits best
for both your budget and your growth.
Best Regards,
Jason Stewart
On 11/01/06 15:06 -0600, Jim Freeze wrote:
> Hi
>
> I am setting up a phone system for a small office.
> The office will have 5-8 phones and a fax line.
> There are 4 hunt lines coming into
HTTP uses TCP. Too much overhead. Add SSL on to that and you have a
huge amount of overhead. The end result would be poor and choppy sound
quality.
Jason
On 21/07/05 21:58 +0200, Rob Scott wrote:
> For work environments where you only get HTTP or HTTPS access, what is
> the feasibility of doing
the problem. What kind of
hardware are you using for FXO?
Jason Stewart
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On 22/07/05 02:49 +0900, Kuniyoshi Murata wrote:
> Hi,
>
> Now, I think I want to disable Asterisk's access to console audio device
> based on the logic above. How can I do that?
>
Make sure the following is in your modules.conf file:
noload => chan_alsa.so
noload => chan_oss.so
On 18/07/05 17:06 -0700, Michael D Schelin wrote:
>I was waiting for everyone to reply so here is mine.. Check out the
>Mediatrix web site. There are no downloads or lists of resellers who might
>have this provisioning software that is normally included with purchase.
>You may be r
On 07/06/05 11:30 -0400, Matt wrote:
> Hi,
> Has anyone used the SS7 link from Digium? If so, how did it work for
> you? Any issues? Anything to be aware of? Do I just need a T1 card
> like the PRI card I have now from Digium?
Hi Matt,
There are some links to user reports on the wiki:
http:/
h more quickly with
Postgres.
My mantra is "pick the right tool for the job". For smaller webapps I
use MySQL. For huge enterprise databases I use PostgreSQL.
Regards,
--
Jason Stewart | Tel: 616-532-2300
Systems Administrator/ | Fax: 616-532-3461
Progr
On 10/02/05 15:10 +0100, Jean-Louis curty wrote:
> so I stopped asterisk, type mail and got a strange mail saying that
> user [EMAIL PROTECTED] could not be reached and body was like if it was
> the result of commands ifconfig etc
>
> unfortunally I don't have the message anymore but I went to the
On 15/12/04 22:53 -0600, Kevin Curtis wrote:
>I would recommend Uniden UIP200 phones. Great sound quality with inbuilt
>phone book, call logs etc works great with asterisk. I recently purchased
>from [1]www.qualvoip.com (they also provided me sample configuration files
>for asterisk
ut a
config file from the bank so in the event of a power and battery
failure I don't have to type in the configuration commands, just load
a file.
Is there a way to get a config from the Adit 600 and load it back in
again?
Thanks,
Jason Stewart
_
On 09/11/04 16:13 -0500, Matt Gibson wrote:
> Hi Everybody,
>
> I have a quick question regarding some old Dialogic hardware. We have an
> old Artisoft PBX (http://www.artisoft.com/PBXPhoneSystems.html). In this
> box are some older ISA Dialogic cards.
> My question is, does anyone know if the f
an
Hi Dan,
I had the same problem when I tried calling myslef via IAX also. I
contacted telesthetic and they said they would look into it (they
thought that it was iaxtel's fault). I waited a couple of weeks and
tried again to no avail. I gave up trying telesthetic/Iaxtel.
Telesthetic does work w
On 05/08/04 15:24 +0100, Tom Lawrence wrote:
> <0>Kernel panic: fatal exception in interrupt
>
> i have had to rebuild the kernel to get the modules in but they seemed to go
> in ok after that. If I run ztcfg I can see both of the cards working. Could
> it be something to do with the IRQ numbers
On 08/07/04 19:04 +0500, Nauman Farooq wrote:
> wondering if anybody knows this..does shady dial work only with a zap
> interface or can it be configured to be used with SIP or IAX.
>
> Nauman
--- Unecessary reply to asterisk-users digest snipped out---
It should work with any type of chann
On 06/07/04 15:17 -0400, Joe Baptista wrote:
>
> -- Forwarded message --
> Date: Wed, 07 Jul 2004 00:31:21 -0400
> From: Declan McCullagh <[EMAIL PROTECTED]>
> To: [EMAIL PROTECTED]
> Subject: [Politech] House bill exports analog phone regs to VoIP
>
>
>
> http://www.politechbot
On 14/05/04 10:36 -0400, Joseph Finley wrote:
>
> >-Original Message-
> >From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of >>>brian k. west
> >Sent: Friday, May 14, 2004 11:24 AM
> >To: [EMAIL PROTECTED]
> >Cc: [EMAIL PROTECTED]
> >Subject: [Asterisk-Users] OMG THE SKY IS FA
On 21/04/04 08:37 -0500, Sean Bruton wrote:
> I am having some difficulty getting a T100P card to work with my PRI.
> When I attempt to make an outbound call via:
>
> exten => 1004,1,Dial(Zap/g1/NPANXX)
>
> I see the following on the asterisk console:
>
> -- Executing Dial("SIP/sbruton-b8ce"
On 22/03/04 17:58 +0100, randulo wrote:
> For info,
>
> I receive the mailing list on a brand new account that is not used for
> anything else.
>
> Just received, a virus (*apparently*) From: [EMAIL PROTECTED]
>
> I suppose there may be 8,000 people getting it but just in case.
No, not necess
On 19/03/04 14:11 +1100, Master Abi wrote:
> Hi,
>
> G726-32 codec from beta firmware 1.0.4.54 now works fine with *. Tested
> on BT101 and HT286 over a 64K DSL line. Some progress but iLBC still has
> not surfaced.
Great news. This fw update breaks NTP sync in the phone for me, but
your milage
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