On Thu, Oct 30, 2008 at 1:40 PM, Juan RodrÃguez [EMAIL PROTECTED] wrote:
With a script connecting to a DB server and looking for the prefix, is a
good solution. This way you don't need to force the user to dial the the
leading 1 (or not to do it), you just look on the DB server and if it does
Here are my numbers, with CentOS 4.4
processor : 0
vendor_id : CentaurHauls
cpu family : 6
model : 9
model name : VIA Nehemiah
stepping: 10
cpu MHz : 533.573
cache size : 64 KB
fdiv_bug: no
hlt_bug : no
f00f_bug: no
On 12/28/06, Bryan M. Johns [EMAIL PROTECTED] wrote:
I recommend the hitachi wifi phones for use with asterisk.
Bryan M. Johns
Partner
Shelton Johns Technology Group
Office: (678) 248-2637 X: 1500
Direct: (678) 229-1809
http://www.sheltonjohns.com
**Sent from my mobile phone**
-Original
Or you could AGI a PHP script that runs before each caller enters the
conference room that sets the AbsoluteTimeout on the channel to midnight
that night (calculating the proper seconds, of course).
The conference would of course die at midnight after all its participants
left.
Nathan
On
On 7/24/06, Douglas Garstang [EMAIL PROTECTED] wrote:
Douglas Garstang wrote:
-Original Message-
When the extension on your desk is ringing, after you have
pressed transfer key a second time(soft or hard key), does
the original caller still hear music on hold, or ringback or
Mac Ethernet ports are auto-switching. Don't need a cross-cable :)
On 4/18/06, Mark Phillips [EMAIL PROTECTED] wrote:
Just for shits and giggles, have you tried using a cross over cable? I'mnot saying it's gonna work because everything I read says you're doing
the right thing but it's worth a
On 4/3/06, Roger McCoy [EMAIL PROTECTED] wrote:
Scratch that!Apparently asterisk just needed to be restarted.. wonder why a reload didn't work?
Reloads don't update music on hold settings.
moh reload seems to work just fine for me.
Nathan
___
On 4/4/06, asterisk user [EMAIL PROTECTED] wrote:
hi all,I can not get voicemail working in realtime withasterisk-1.2.6. extconfig.conf is correct
voicemail = odbc,asterisk,voicemail_usersi am getting the fallowing errorExecuting Answer(SIP/xx.xx.xx.xxx-0a02e1c0, ) innew stack -- Executing
On 3/29/06, Brian Roy [EMAIL PROTECTED] wrote:
I'm currently using Asterisk running version 1.2.5 and trying to use
cdr_odbc to connect to a Microsoft SQL database. I have everything running,
but the insert statement being sent to database doesn't appear to have the
start, answer, end
Alright, I've come across a really strange issue and I've been banging
my head trying to figure it out.
I have 3 machines. 1 Dell Dimension 4100, Pentium 3. 1 Dell 400SC,
Pentium 4. 1 Dell 1600SC, Xeon. I run mpg123 0.59r on each machine.
Using RH9 with a 2.4.20-8 kernel, each machine plays
On 3/16/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Q: What the deal with the limit on the number of people you can monitor
for presence?
A: There is no limit in the phone. This is an Asterisk limitation.
BS. Polycom even stated it's a polycom limitation which will be fixed in
On 3/15/06, Giorgio Incantalupo [EMAIL PROTECTED] wrote:
Hi,
we have an asterisk 1.2.1 box and 2 polycom SIP phones. We'd like to
show the called name on the calling polycom display instead of his /her
extensions as I do with the caller name on the called polycom.
Is it possible? If yes, how?
On 3/15/06, Alexander Lopez [EMAIL PROTECTED] wrote:
This is a function of the Phone itself. Asterisk has nothing to do with
it as it does not know anything about the call until after the SIP
device 'sends' it.
To my knowledge it is not posible. I don't even think a SIP standard is
On 3/13/06, Chris Stenton [EMAIL PROTECTED] wrote:
I have had no issues with 8.2 so far!
Chris
Except the Caller ID issue reported in another thread?
This issue has been fixed in SIP firmware 7.5
Omar A. Sabek
Yes, and I read that SIP 7.5 firmware have some other issues. They
On 3/9/06, Greg Oliver [EMAIL PROTECTED] wrote:
On Thu, 2006-03-09 at 10:36 -0500, Nabeel Jafferali wrote: Is there a way to display the time of the 7960 running firmware
7.4? Im unable to find any information. Add the following to SIPDefault.cnf or SIPMAC.cnf: sntp_server: time.nrc.ca
On 2/23/06, Darrick Hartman [EMAIL PROTECTED] wrote:
Dinesh Nair wrote: On 02/22/06 11:08 C F said the following:
http://bugs.digium.com/view.php?id=5574 That is a patch that will do just that. while an app is nice, followme could have been done thru some nifty dialplan work as well.True, but
On 2/23/06, Olle E Johansson [EMAIL PROTECTED] wrote:
Isaac Xiao wrote: We have the same issue happened to all Asterisk versions of 1.2.X (I tried all). In CLI, it shows "-- Incoming call: Got SIP response 500
Internal Server Error back from 192.168.2.104". Once you see this msg, the buddy watch
On 10/21/05, Sean Cook [EMAIL PROTECTED] wrote:
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Sherwood McGowan
Sent: Friday, October 21, 2005 3:45 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE:
On 1/31/06, Damon Estep [EMAIL PROTECTED] wrote:
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Jerry Glomph Black
Sent: Monday, January 30, 2006 11:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
On 1/27/06, Kevin P. Fleming [EMAIL PROTECTED] wrote:
Sean Cook wrote:
Is there an implementation for shared line support in asterisk? I know
that with hint I can watch line status... I just want to be able to
pick up on an extension when ringing or jumping in on a call by punching
the
Greetings,
I have two machines. One is a P3 Dell Dimension 4100, the other is a
PowerEdge SC420. Both are running Asterisk 1.0.7, the PowerEdge has a
TE405P card in it, the Dimension has a Digium X100P present (although
not modprobed). Each machine has mpg123 0.59r loaded, and is using
the
On Fri, 11 Mar 2005 16:38:38 -0600, Nathan Bowyer [EMAIL PROTECTED] wrote:
On Fri, 11 Mar 2005 15:04:03 -0600, Matthew Boehm [EMAIL PROTECTED] wrote:
You can't have this:
[from-sip]
switch = Realtime/[EMAIL PROTECTED]
The context in your extensions.conf must be different from your
Greetings,
I'm having some trouble with the realtime engines. When asterisk
loads, everything looks fine, there don't seem to be any problems via
notices or anything. Furthermore, cdr_odbc is working, and actively
logging my failed call attempts to db through ODBC using the same DSN.
unixODBC
doesn't
work. The extension to access the voicemail is static in
extensions.conf.
-Matthew
Nathan Bowyer wrote:
Greetings,
I'm having some trouble with the realtime engines. When asterisk
loads, everything looks fine, there don't seem to be any problems via
notices or anything
On Wed, 19 Jan 2005 12:56:59 -0500, Paul Rodan [EMAIL PROTECTED] wrote:
[snip]
3. Create historical report to pull agent activity. Should display
login/logout activity. Be able to pull information by rep and timeframe.
This could probably be done with the CDRs and queue_log.
4. Create
On Mon, 29 Nov 2004 17:08:23 -0800, Brian Wright [EMAIL PROTECTED] wrote:
I'm trying to get zaptel 1.0.2 compiled on FC2 or FC3 and I'm getting
compile time errors. Systems include:
FC2: Linux 2.6.9-1.3_FC2 #1 Mon Nov 15 14:46:43 EST 2004
i686 i686 i386 GNU/Linux
FC3: Linux
On Sun, 28 Nov 2004 16:32:10 -0600, Rich Adamson [EMAIL PROTECTED] wrote:
I'm looking at the Sayson 480i or the Cisco CP-7960. Which one would you
suggest and why please ?
I briefly tested the 480i a couple of weeks ago. Had a problem in that it
would not use the tftp server address
Hello,
I have a problem which I've found quite strange, to say the least. I
have a client who uses long distance access codes from their LD
provider. The codes are 4-digits, nothing extraordinary there. The
problem is, if you dial the digits quickly, without pauses inbetween
them, the LD
On Mon, 08 Nov 2004 15:05:33 +0800, el Flynn [EMAIL PROTECTED] wrote:
Nathan Bowyer wrote:
Doesn't seem to work for me that way. Anyone else got any ideas?
When I look at the code, it looks like copying what roundrobin does,
then simply removing the pos whenever you complete a call
-Original Message-
From: Nathan Bowyer [mailto:[EMAIL PROTECTED]
Sent: Wednesday, October 27, 2004 9:40 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] New Strategy in App_queue
Hello,
I've been looking at and working on a new queue strategy for
about a week now, off
Hello,
I've been looking at and working on a new queue strategy for about a
week now, off and on. However, being that I'm not really a C
programmer (yet, anyway) I have not made much progress.
The concept is rather simple, probably the easiest of all the queue
strategies. I simply want to it
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