The Adit 600 is really good.
- Original Message -
From: Eran Gal [EMAIL PROTECTED]
Date: Wed, 25 Aug 2004 12:11:33 +0200
Subject: [Asterisk-Users] channel banks
To: [EMAIL PROTECTED]
Can anyone recommend a channel bank that works well with asterisk?
I'm looking for two types:
1.
Can you connect to mysql from the command line with the user/pass you
setup?
Also, when you test make sure you add: -h localhost to the flags you
pass 'mysql'. This will make sure it doesn't try connecting via the
unix domain socket. The permissions in the 'mysql' db may be set not to
allow
Yes too all, but the features you're talking about are more phone
related than asterisk related.
If your phone can log in multiple lines, then asterisk will send calls
destined for those lines to that phone.
I think most decent sip phones can do this.
On Wed, 2004-08-18 at 15:28, me peaceout
I'm not sure if this is your issue or not, but it looks like ext= 1,
starts over at the bottom of the 1's. You have 1,1-10 and then 1,1 and
2 after it. I can see how asterisk might get confused if you sent your
call back to ext 1 at starting point 1 or 2.
On Tue, 2004-08-17 at 15:03, defiance
I understand that you not supposed to have to reload anything for the
voicemail.conf to be updated on a running asterisk (similar to the
meetme.conf). However, if I add someone to voicemail.conf, asterisk
doesn't see them until I restart asterisk.
I thought maybe after adding someone, that if I
Could you post the part of your extensions.conf in question?
On Wed, 2004-08-04 at 11:47, [EMAIL PROTECTED] wrote:
Hi:
I am trying to setup a simple auto attendant with Asterisk using SIP extensions. I
have an IP trunk to voicepulse and my outgoing calls go over that.
I can also receive
Has anyone used this feature successfully? I 'think' I have a .wav file
that it wants.
Here is what 'file' says:
sf-george.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16
bit, mono 8000 Hz
I see the logs on my web server as it tries to access it, but all I get
is a screech out
You may need to use 'passive' ftp.
http://compnetworking.about.com/cs/novellgroupwise/ht/setpassiveftpie.htm
The above is a link for Internet Explorer.
Give it a try...it could be a firewall problem on your side.
On Thu, 2004-07-29 at 16:51, Lee Howard wrote:
And if you then do an 'ls'?
It
Depending on the context that your 'incoming' lines are on, you can do
something like this:
[incoming-lines]
exten = _1235551212,Macro(autoatt-company1)
exten = _1235551213,Macro(autoatt-company2)
[macro-autoatt-company1]
Do some junk, dial some peeps
[macro-autoatt-company2]
Do some junk,
You don't need the _ on the front of those extensions since those
particular examples aren't patterns. My mistake.
Check out www.voip-info.org for MANY good examples.
On Fri, 2004-07-23 at 15:21, Joshua McClintock wrote:
Depending on the context that your 'incoming' lines are on, you can do
Putting on Tin Foil Hat to pickup brain waves
Let's see here, from the information I'm receiving from my Brain Wave
Reader, it would seem that you aren't emitting enough activity for me to
determine much of anything.
I would suggest posting some of the errors you're getting.
On Wed, 2004-07-21
I think your kernel module isn't loaded for your card.
Once those get loaded, the stuff in /dev gets created.
Look in /lib/modules/kernel number/misc for the kernel modules
Do a 'demod -a' first and then you can do a blanket modprobe like this:
modprobe \*
It'll pretty much load all your
I would make a symlink in /usr/include that looks like this
linux - /usr/src/linux/include/linux
cd /usr/include
ln -s /usr/src/linux/include/linux linux
Sometimes you needs the asm as well in the same directory as above
asm - /usr/src/linux/include/asm
cd /usr/include
ln -s
Our production environment is using a 4 port 3ware 8500 series card with
2 drives (mirrored) on the pstn (2 t1 cards) machine and an 8 port 3ware
8500 series with 8 drives (raid5) on the pbx/vm machine.
Flawless so far.
On Wed, 2004-07-21 at 16:25, Kevin P. Fleming wrote:
Scott Laird wrote:
No good reason, except that the box may be used for something else in
the future..
On Wed, 2004-07-21 at 17:26, Scott Laird wrote:
On Jul 21, 2004, at 4:53 PM, Joshua McClintock wrote:
Our production environment is using a 4 port 3ware 8500 series card
with
2 drives (mirrored) on the pstn
I have an asterisk box setup with a T1 card (hooked to a pri from the
telco). Sometimes when people call a number that rings down that pri,
the first ring is really really long, like 3 normal rings put together.
My indications.conf is the default that comes out after you do a make
install. Is
We're using asterisk in production at my place. We have about 45 users
(33 snom 200 sip phones and the rest on soft phones). We're connected
to the telco via a pri (23 useable channels) using the digium t1 card.
We have another t1 card hooked to an adit 600 channel bank for analog
services.
Hello, I have an Adit 600 (3 FXS cards) hooked up to a digium T1 card in my
asterisk box. I 'connected' the slots to the a:1 T1 interfaces via the
command line. The slots (3 fxs) are configured with 'ls' signaling. I
configured the T1 card with the same line settings as the T1 interfaces on
the
:
On Monday 28 June 2004 23:04, Joshua McClintock wrote:
Hello, I have an Adit 600 (3 FXS cards) hooked up to a digium T1 card in
my
asterisk box. I 'connected' the slots to the a:1 T1 interfaces via the
command line. The slots (3 fxs) are configured with 'ls' signaling. I
configured the T1 card
-16 2:1-8
connect a:1:17-24 3:1-8
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joshua
McClintock
Sent: Tuesday, June 29, 2004 12:25 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Adit 600 - Getting Dial Tone
Ok so to be clear about what I need to have
I believe libpq is a postgres dev library. You probally need a psql-dev
package of some sort.
- Original Message -
From: Freddy Setiawan [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, June 28, 2004 9:34 PM
Subject: [Asterisk-Users] cannot make app_prepaid
hai there, today i
- Original Message -
From: Andrew Kohlsmith [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, June 28, 2004 9:39 PM
Subject: Re: [Asterisk-Users] Adit 600 - Getting Dial Tone
On Tuesday 29 June 2004 00:24, Joshua McClintock wrote:
Asterisk side: Configure T1 card like this (it's
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