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 the office.
We have made no hardware
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
of
hardware are you using for FXO?
Jason Stewart
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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
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
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:
. For huge enterprise databases I use PostgreSQL.
Regards,
--
Jason Stewart | Tel: 616-532-2300
Systems Administrator/ | Fax: 616-532-3461
Programmer | Email: [EMAIL PROTECTED]
Right to Life of Michigan | Web: http://www.rtl.org
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 log
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).
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
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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
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 with FWD. I've tried this myself and it does
work even using FWD's IAX services.
Cheers,
Jason Stewart
On 05/08/04 15:24 +0100, Tom Lawrence wrote:
snip
0Kernel 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 channel,
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
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 FALLING!! NOT!!!
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 necessarily.
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 may
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