Anthony Francis wrote:
Steve Underwood wrote:
C. Savinovich wrote:
I am puzzled by the quality of magicjack. I keep trying to figure out how
they can the quality be that adequate. Since Skype also has an excellent
quality, that leaves me to believe that software based calls
On Sat, Jul 12, 2008 at 01:14:36AM +0100, Grey Man wrote:
On Fri, Jul 11, 2008 at 7:50 PM, Ronald Lewis [EMAIL PROTECTED] wrote:
I've just added a PREVIEW release of my upcoming how-to guide for Asterisk
PBX on EC2. It is based on months of testing and evaluating Asterisk on EC2.
It
On Sat, Jul 12, 2008 at 10:26:24AM +0800, Steve Underwood wrote:
C. Savinovich wrote:
I am puzzled by the quality of magicjack. I keep trying to figure out how
they can the quality be that adequate. Since Skype also has an excellent
quality, that leaves me to believe that software based
Tzafrir Cohen wrote:
On Sat, Jul 12, 2008 at 10:26:24AM +0800, Steve Underwood wrote:
C. Savinovich wrote:
I am puzzled by the quality of magicjack. I keep trying to figure out how
they can the quality be that adequate. Since Skype also has an excellent
quality, that leaves me to
On Fri, 2008-07-11 at 11:22 -0500, Tilghman Lesher wrote:
On Friday 11 July 2008 09:17:37 Artie Gold wrote:
In updating to 1.4.21 recently, we've encountered a problem, when running
over a satellite connection (where the latency is considerable; a regular
internet connection did not exhibit
On Fri, 2008-07-11 at 18:37 +0200, Dave Cotton wrote:
SIP wrote:
Joseph wrote:
I need another Sipura 3K and the replacement I think is Linksys SPA3102.
Any input on how reliable is it?
We have a few dozen subscribers using them at any given point in time. I
and my wife even use
On Fri, Jul 11, 2008 at 10:52:53AM -0700, Douglas Garstang wrote:
Wanting to provide a real time call timer on a web page.
Can't you get information about other channels through the manager
interface without this special AGI?
Maybe you just need to somehow mark those channels as interesting
Hans Witvliet wrote:
There's not much that can stand lightning (not just a direct hit), so
you cant't blame the sipura box for that.
Even when it was build, using a Faraday-cage with double insulation with
optocouplers, the amount of energy picked up by a 3 km line is beyond
commercial
On Fri, 11 Jul 2008 19:26:06 -0400, Steve Totaro wrote:
As Michael Graves points out, people will hack it to run on thin
clients and why not virtual machines with very limited access? Maybe
an AP with a USB port and OpenWRT or something?
Since it needs to run their app it's probablly limited to
On Sat, Jul 12, 2008 at 9:32 AM, Michael Graves [EMAIL PROTECTED] wrote:
On Fri, 11 Jul 2008 19:26:06 -0400, Steve Totaro wrote:
As Michael Graves points out, people will hack it to run on thin
clients and why not virtual machines with very limited access? Maybe
an AP with a USB port and
I can not seem to get AsteriskNow to register my SIP provider correctly?
I can do this manually when compiling Asterisk and installing it w/o a
GUI, but not with this. I just get the following message.
-- Registration for '[EMAIL PROTECTED]' timed out, trying again (Attempt #22)
The register
My understanding is Skype's secret is using the iLBC codec, which Cisco
has also licensed for their 79X2 models as well. I travel and lot and
in places where Yahoo Phone Out or MSN Phone or Cisco IP Communicator
will fail the Skype client will work. The iLBC codec can really handle
packet loss.
Jason Aarons (US) wrote:
My understanding is Skype's secret is using the iLBC codec, which Cisco
has also licensed for their 79X2 models as well. I travel and lot and
in places where Yahoo Phone Out or MSN Phone or Cisco IP Communicator
will fail the Skype client will work. The iLBC codec
On Sat, Jul 12, 2008 at 09:54:37AM -0400, Steve Totaro wrote:
On Sat, Jul 12, 2008 at 9:32 AM, Michael Graves [EMAIL PROTECTED] wrote:
On Fri, 11 Jul 2008 19:26:06 -0400, Steve Totaro wrote:
As Michael Graves points out, people will hack it to run on thin
clients and why not virtual
On Sat, Jul 12, 2008 at 11:18 AM, Tzafrir Cohen
[EMAIL PROTECTED] wrote:
On Sat, Jul 12, 2008 at 09:54:37AM -0400, Steve Totaro wrote:
On Sat, Jul 12, 2008 at 9:32 AM, Michael Graves [EMAIL PROTECTED] wrote:
On Fri, 11 Jul 2008 19:26:06 -0400, Steve Totaro wrote:
As Michael Graves points
On Sat, Jul 12, 2008 at 11:52:21AM -0400, Steve Totaro wrote:
On Sat, Jul 12, 2008 at 11:18 AM, Tzafrir Cohen
[EMAIL PROTECTED] wrote:
On Sat, Jul 12, 2008 at 09:54:37AM -0400, Steve Totaro wrote:
On Sat, Jul 12, 2008 at 9:32 AM, Michael Graves [EMAIL PROTECTED] wrote:
On Fri, 11 Jul 2008
Hello,
I am running Asterisk 1.4.20-1 and having the exact same problem. It
looks like others are having issues as well according to this thread:
http://www.trixbox.org/forums/trixbox-forums/help/recordings-out-sync-using-mixmonitor
Anyone have any idea's?
On Wed, Apr 9, 2008 at 7:16 AM,
The person I am working is building a calling card. They want to allow the user
to recharge their account when their time runs out (without hanging up the
current call). I got no idea how to implement that. In addition, they don't
want to charge the user for the time they spend recharging their
If anyone on the list has a Polycom 601 + sidecar expansion they want to
sell for less than $250 including shipping to New York 10027 then email
me details.
Cheers,
Dean
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
All,
I was able to use the Redirect AMI command to take two bridged channels and
send them elsewhere in the dial plan. Great.
Now... how can I bridge them back together again? Looks like Asterisk 1.6 might
have a bridge command. What about Asterisk 1.4?
Doug.
Hi, this is my first post to the list, but I have tried to search
elsewhere for a solution, and have had a read of 'Asterisk - The
Future of Telephony'. So you could say that I have at least tried to
RTFM as it were!
I've configured a couple of Asterisk instances on both Debian and
CentOS based
On Sat, 12 Jul 2008, Douglas Garstang wrote:
The person I am working is building a calling card. They want to allow
the user to recharge their account when their time runs out (without
hanging up the current call). I got no idea how to implement that. In
addition, they don't want to charge
Hello,
From the netstat output my initial *guess* is that asterisk is listening
(udp/5060, udp/2727, among others). One way to tell for sure would be
to run 'lsof -i' which would show you the process associated with the
port.
As far as the call not reaching asterisk or being a firewall
On Sun, 13 Jul 2008, Chris Rowson wrote:
Hi, this is my first post to the list, but I have tried to search
elsewhere for a solution, and have had a read of 'Asterisk - The
Future of Telephony'. So you could say that I have at least tried to
RTFM as it were!
I've configured a couple of
On Sat, 12 Jul 2008 10:54:07 -0400, Julio Arruda wrote:
Jason Aarons (US) wrote:
My understanding is Skype's secret is using the iLBC codec, which Cisco
has also licensed for their 79X2 models as well. I travel and lot and
in places where Yahoo Phone Out or MSN Phone or Cisco IP Communicator
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