On 01/05/2008 00:27, George Pajari [EMAIL PROTECTED] wrote:
On Wed, 2008-04-30 at 13:11 +0100, Dee Lowndes wrote:
...Question is do I still need to worry about timing and if so can this be
resolved in a Xen enviroment?...
We're an ITSP and use OpenVZ to offer customers Virtual
Hi All,
I am trying to decide weather to move my asterisk setup on to a Xen
setup or not. I do use transcoding, meetme and music on hold although in a
purely sip scenario real lines are handled via cisco kit. Currently its a
dedicated box with X100P card for timing handling it however it's
Hi All,
I am trying to setup my snom 190 so that the LED's light up when
one of my shared lines are in use.
e.g.
Extension 2 should ring on the snom and the phone associated with
extension 2 and I should be able to see if the phone associated with
extension 2 is making a call on the
with no signal if I
dial 2 from another phone. Also if I push the orange lit button on the
Snom it just rings extension 2 ideally I would like to be able to dial
from that extension but on the snom like a shared line.
Dee
Netherlands.
On Dec 17, 2004, at 8:37 PM, Dee Lowndes wrote:
Hi
Hi all,
I am noticing echo/jitter problems when going sip - asterisk
iax (ALAW)- asterisk pstn depending on the codec I use. Both ULAW/ALAW
works fine on the budgetone and ata286 but g726 only works well on the
budgetone.
Ilbc just doesn't work well with broken speech and echo issues.
Hi all,
Does anyone know if its possible to get an ATA 286 to make the
actual phone to ring instead of just the ATA ringing?
Cheers,
Dee
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-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Juergen K. Zick
Sent: 12 October 2004 18:44
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Slackware 10.0/Asterisk 1.0 compile error
Hi Folks,
Hi All,
I am testing out Asterisk with IAX between 2 machines on local
IP addresses and I want one machine to act as an IAX gateway with the
other connecting to it. Anyone know of or can supply me an example of
how to do this?
Cheers,
Dee
___
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Chris Shaw
Sent: 17 August 2004 21:12
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] New $89 VOIP phone
A solution to this very problem has already been discussed... in
fact...
Hi all,
Anyone know where/how I can setup my own menu to work like the
voicemailmain menu.
e.g.
extension.conf
exten = 888,1,mymenusystem
exten = 888,2,Goto(s,6)
then somewhere mymenusystem plays message and give options to goto exten
1, 2, 3 etc
Thanks in advance.
Dee
Hey All,
I am using an x100p on a UK Telewest phone line and appear to be having
problems with end user hang ups.
If I call my * from and phone line and let * pick it up when I hang up the
mobile or whatever I am calling from * continues with the call as if I
haven't hung up.
Was wondering if
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