Justin Tunney wrote:
Lobster Technologies has just anounced the release of the most
annoying open source IVR application ever devised by lobsters called
PhoneParrot. PhoneParrot is an app that uses silence detection to
Hahahaha, too funny.
Doug
-- Ben Franklin quote: Those who would give up
Hi Iain,
Thank you for that. That should work well for me.
On 8/29/06, Iain Young [EMAIL PROTECTED] wrote:
On Tue, Aug 29, 2006 at 02:18:32PM +1000, Devraj Mukherjee wrote:
The simplest way I can think of solving this is using prefixes, so
someone appends a 0 or 1 and the dialplan puts the
I'm wondering if anyone can tell me what the following output,
repeated about once per minute on my verbose=5 CLI , means.
-- Contact header: transport
-- Contact header: q
-- Contact header: transport
-- Contact header: q
I'm on the latest version, 1.2.11, and am recovering
Hi list,
On my asterisk based home pbx system, i have 1 zap interface(wildcard x100p)
and 1 sipura 2000(which receiver is connected to) and some sip account for my
long distances and incoming calls.
When my call comes from SIP i've got no problem. However when calls comes from
zap, the zap
Hi all,
I have three Asterisk servers behind a SER. I want to know how to
configure the Dispatcher module of SER to achieve load balance for the
Asterisk servers. I have visited
http://www.openser.org/docs/modules/1.1.x/dispatcher.html, is there any
web sites have more detail information on
Stefan-Michael. Guenther (in-put GbR) wrote:
Hi,
a few weeks ago someone mentioned a menu point 2 in the advanced options of
the voicemail menu, which allows a call back to the caller who left the
message.
Feature needs to be enabled in the voicemail.conf
callback=context
I've
Christian Schlatter wrote:
I ran in the same issues as John Todd did some while ago:
http://lists.digium.com/pipermail/asterisk-users/2005-November/129541.html
I use qualify=yes to ping our internal SIP proxies for failover and
therefore I have very low delays, e.g.
Name/usernameHost
Greetings, I finally
got my Asterisk server up and running and now am in the process of looking for a
provider to use as a SIP trunk. Unfortunately, I'm realizing that
unlimited really is in fact limited -- Galaxy Voice's unlimited plan, for
example, translates to a mere 2500 minutes/month.
Assuming that the patch was applied correctly and that each context of sip.conf
and also [general] context have all those 3 parameters you mentioned above, try
to activate also the accept reinvites option in those contexts of sip.conf. Use
ethereal to try to see if communication fails in that
That might not be a good idea. If you transfer or forward calls on your phones,
you MUST make sure the transferred or forwarded call goes back to the same
Asterisk box it was handled on. If you use the dispatcher, and load balance,
there is no guarantee of that, and transfers and forwarding
Hi Douglas,
Thanks for your advice. So is there any alternatives?
Thanks!
Andy
Douglas Garstang wrote:
That might not be a good idea. If you transfer or forward calls on your phones,
you MUST make sure the transferred or forwarded call goes back to the same
Asterisk box it was handled on.
*dunks email in bucket*
Heheh...Gee, ya think, Dean? Pardon my possession of an opinion.
*Cautiously waits for next flame*
SKM
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Dean Collins
Sent: Monday, August 28, 2006 9:38 PM
To: Asterisk
Not really. You need to make sure that a phone always uses the same primary
asterisk system under normal circumstances. You can simulate load balancing my
staggering your phones to use different asterisk systems.
-Original Message-
From: Andy Chung (Power-All)
Steven M. Sawczyn wrote:
Greetings, I finally got my Asterisk server up and running and now am in
the process of looking for a provider to use as a SIP trunk.
Unfortunately, I'm realizing that unlimited really is in fact limited --
Galaxy Voice's unlimited plan, for example, translates to a
Well, it really depends on what he's using the asterisk servers for. If
it's for voicemail or apps, he'll just have to make sure that certain
apps land on certain servers, and voicemail can be distributed for
various things. If ser can do what I've heard/read it can do, it can
handle all the
Hi,
I recently bought a handful of g729 licenses and moved all my
equipment over to use it. We terminate most of our calls with a
provider that supports g729, so it's g729 all the way through from the
phone on the desk to the provider. Asterisk works very well in
passthrough mode, simply moving
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