that's in your
location and dial settings in the phone somewhere.
Thanks,
Matt G
: http://www.voipphreak.ca
: http://www.ratemydialplan.com
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Is there any way you could get a cut-sheet from Verizon. I know they
are difficult to work with, but it would help to see for sure if your
circuit is indeed Loop-start. You could always try EM_wink or EM
immediate and see if there is any change.
MATT---
On 7/8/08, Daniel Hazelbaker [EMAIL
(Matthieu)
exten = s,n,ChanIsAvail(SIP/605,s)
Work, because line 7 exist
I use Asterisk 1.4.18
That's what it's supposed to do - maybe you are referring to the
deprecated +101 jumps?
- --
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Matt Riddell
Director
___
http
installation is doing
(or has been setup in the voip software to do)
Thanks,
Matt G
: http://www.voipphreak.ca
: http://www.ratemydialplan.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Remco Barendse
Sent: Monday, July 07, 2008 5:21 AM
To: Asterisk
yellow and one black
ethernet socket. (WAN/LAN)
- --
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Matt Riddell
Director
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http://www.venturevoip.com
the Authenticate
application?
- --
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Matt Riddell
Director
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Florian Hackenberger wrote:
Hi!
I'm using asterisk 1.4.17 with twinkle and a custom phone based on
iaxclient 2.0.2 and I'm struggling a bit with DTMF and features.conf.
Maybe the feature digit timeout?
- --
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Matt Riddell
acknowledge the call, and log that fact. (Basically an automated soft
phone). I found some info on how to do this here:
Please also attach the call file you are trying to use.
- --
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Matt Riddell
Director
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member won't just hit Ignore on their phone
and send it directly to voicemail.
You'd probably want to look at using the local channel and the followme
application + /etc/asterisk/followme.conf
- --
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Matt Riddell
Director
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Tilghman Lesher wrote:
On Thursday 03 July 2008 00:27:00 Matt Riddell wrote:
Tilghman Lesher wrote:
I find that a good number of people are using . in a pattern in
situations that are entirely unnecessary (such as local numbers). The
only place
/asterisk/outgoing
and
date
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Matt Riddell
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.
Bear in mind that if you want to do show queues as opposed to show queue
and you are using a realtime queue which has not been used, you will
need to type show queue my_queue_name first before it will show up in
show queues.
- --
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Matt Riddell
Director
on if they are
available for contracting work.
It is a work in progress and is being completed solely outside of office
hours, so the development pace is somewhat slow, but I'm attempting to
add new features every evening.
Enjoy! :-)
- --
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Matt Riddell
Director
was IAXtel
but that does not seem to be the case.
I think you're referring to FWDOUT.
- --
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Matt Riddell
Director
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is authentication-less, but its still possible...
and if security is in mind you could limit your TFTP server to specific
source IPs
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=mobile
call-limit=50
As for internet calling, once you have the feature installed, there is a new
option available in the connection settings use internet dialing when
available. I have this turned on, and calls route over wifi/voip when I am
registered instead of the cell network. Hth!
Thanks,
Matt G
really just making some logical
assumptions here.
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Hi Roland,
Did you try:
http://www.voipphreak.ca/2008/03/29/enable-the-hidden-voip-features-of-windo
ws-mobile-6x-for-free-voip-calls-using-asterisk/
We have this successfully working on a Touch (ELF), and a HTC Tilt (Tytn II)
Thanks,
Matt G
: http://www.voipphreak.ca
: http
to email me at
[EMAIL PROTECTED]
Thanks,
Matt G
: http://www.voipphreak.ca
: http://www.ratemydialplan.com
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of RoLaNd RoLaNd
Sent: Thursday, July 03, 2008 2:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
the
other 100%?
If so, what echo canceller are you using?
- --
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Matt Riddell
Director
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http
in sight.
- --
Kind Regards,
Matt Riddell
Director
___
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http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss
had the same problem on a 410p, and ended up fixing it by changing to
a lower suppression.
- --
Kind Regards,
Matt Riddell
Director
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-BEGIN PGP SIGNED MESSAGE-
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Tilghman Lesher wrote:
On Wednesday 02 July 2008 21:59:02 Matt Riddell wrote:
Tilghman Lesher wrote:
On Tuesday 01 July 2008 14:27:26 bilal ghayyad wrote:
Can anyone advise how to increase the waiting time to consider the
number is dialed
a period in my dialplan is for when I'm dialling an international number, when
I don't know in advance how long the number should be. A little bit of
planning goes a long way.
Here we have variable length cellphones as well as international.
- --
Kind Regards,
Matt Riddell
Director
-the-hidden-voip-features-of-windo
ws-mobile-6x-for-free-voip-calls-using-asterisk/
Thanks,
Matt G
: http://www.voipphreak.ca
: http://www.ratemydialplan.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Remco Barendse
Sent: Monday, June 30, 2008 3:52 PM
actually have an AstriBank then there is
no sense in even compiling/installing the drivers for it.
I;m guessing you haven't run a make menuselect to select only the drivers you
need?
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: Pick whatever distro you are comfortable with
Distro is more of a personal choice than anything... ultimatly they all have
the same software available to them (for the most part), they all just do it
a little bit differently.
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Matt Watson
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, but it may be. There's more information on this forum posting
over here that may help debug the faulty code in the GUI.
http://forum.utorrent.com/viewtopic.php?id=32335
Thanks,
Matt G
: http://www.voipphreak.ca
: http://www.ratemydialplan.com
From: [EMAIL PROTECTED]
[mailto:[EMAIL
... if
you compiled from source maybe you just forgot to 'make install' for
asterisk?
--
Matt Watson
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anything useful.
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Matt Watson
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asterisk-users mailing list
monitoring server can still page me in the event of an internet outage).
Anyhow, sorry that message got a bit lengthy pretty fast!
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OrecX that are extremely
flexible and offer a user interface for file access and management as
well as live monitoring.
All of the high-volume recording solutions we have installed use
separate archive servers to store the recordings.
MATT---
On 6/18/08, Mark Hamilton [EMAIL PROTECTED] wrote
Hello,
If you have a PRI-T1 in the USA, then you can set outgoing CallerID
with just about any carrier.
MATT---
On 6/17/08, Mark Hamilton [EMAIL PROTECTED] wrote:
How can they even set such 1234567890 callerIDs anyway?
For example, our inter/intra state calling depends a lot on the callerIDs
IP670 was just released...about 30% more than the IP650.
http://polycom.com/usa/en/products/voice/desktop/soundpoint_ip/soundpoint_ip670.html
-Matt
On Tue, Apr 29, 2008 at 1:02 AM, Patrick
[EMAIL PROTECTED] wrote:
On Mon, 2008-04-28 at 14:49 -1000, Matt Darnell wrote:
Anyone seen anything
How come he has it, and he's in Paris! I'm in Toronto, and I don't have
it?
:(
I was thinking the same thing, Ottawa here.. :(
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a small 7-seat inbound call center like this for a client on a
P4 1.6GHz PC and it has worked great for the last 3 years.
Thanks,
MATT---
On 6/16/08, Sherwood McGowan [EMAIL PROTECTED] wrote:
broadband Voice wrote:
Is anyone using Asterisk as a call center. I want to be able to set it
up
was also in my pencil cup) saved my bacon as
well that night :)
Any chance of more of these being handed out at Astricon this year?
Thanks,
MATT---
On 6/16/08, Mark Hamilton [EMAIL PROTECTED] wrote:
Now you're just trying to get us all jealous, Steve. No good.
But I'd like that screwdriver
they were literally trying to offload their job to this list
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.
All in all it looks like a decent product... i'd be interested in hearing from
anybody that might of been using them for a long period of time (1-2yrs+).
I'm pretty picky about power distribution, i've seen bad power cause too many
problems in my computing history.
--
Matt Watson
http
last post will do load metering...
it'll cost you about twice as much as the one Steve posted however. The sale
price that is, couple hundred more than the regular price.
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Matt Watson
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to its device
classification it could be a web interface like this device, a telnet/ssh
interface, it might not even have any remote capability and just have
physically switches for each port.
--
Matt Watson
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on it. However I think
that the APC logo on something means alot more than the Sony logo :)
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, it is illegal to scrub leads for a company against
the USA FTC DNC lists unless those companies have paid the FTC and
registered to have access to those leads, do you verify FTC
registration before offering this service?
MATT---
On 6/13/08, Muhammad Zulqarnain [EMAIL PROTECTED] wrote:
Dear User
FTC DNC lists.
A large portion of these companies are doing lead-generation for
USA-based companies, and over the years a lot of those USA-based
companies have been shut down for the activities of their lead
suppliers.
MATT---
On 6/13/08, Dean Collins [EMAIL PROTECTED] wrote:
Yep it's funny how
Not sure who complains, but it has happened before. the first case was
in 2006 when Phase One Marketing who was fined by the FTC for
indirectly acquiring the FTC DNC list from another entity.
MATT---
On 6/13/08, Steve Totaro [EMAIL PROTECTED] wrote:
I suppose if they are properly scrubbing
was fined $1 million for violation
of the DNC through it's affiliates, some of which were off-shore lead
generation companies. The company shut down because of this fine.
MATT---
On 6/13/08, Dean Collins [EMAIL PROTECTED] wrote:
A large portion of these companies are doing lead-generation
a few dozen people involved in
each one, it would be great if this could be done on a larger scale
and officially organized.
Thanks,
MATT---
On 6/12/08, John Todd [EMAIL PROTECTED] wrote:
We're busily churning away at creating the Astricon
(http://www.astricon.net/) talk track this year
Ah, you got me there! Could start throwing in a lot of Over's going down
that road :)
--
Matt
http://www.mattgwatson.ca
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen
Sent: Tuesday, June 10, 2008 4:10 AM
To: asterisk-users
don;t need the duplicate group=,
signalling=, switchtype= in zapata.conf
4. you can ditch rxwink= that setting is for non-PRI T1s
try that and see if that helps... I suspect the span not being used as primary
timing source is whats causing your greif.
good luck!
--
Matt Watson
http
. I've never tried faxing between ATA's so I
don;t know if they can actually negotiate T.38 support between each other,
but I don't really see a reason why they couldn't.
--
Matt Watson
http://www.mattgwatson.ca
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, there is the T.38 protocol which is designed to solve this exact
problem, Asterisk support for it is just rather limited currently (pass-thru
only).
T.38 often gets referred to as FoIP (Fax over Ineternet Protocol)
--
Matt Watson
http://www.mattgwatson.ca
talked about DNS as a possible cause and see if there are any
similarities.
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Tzafrir Cohen wrote:
On Wed, Jun 04, 2008 at 04:06:28PM +1200, Matt Riddell wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Tilghman Lesher wrote:
On Tuesday 03 June 2008 10:12:58 Todd Reese wrote:
Hi All,
I'm stumped on this and I
Hello,
We routinely run meetme with over 140 ULAW channels connected to 70
meetme rooms with no issues on an Intel Core 2 Quad core CPU.
The major factor in capacity would be your CPU and RAM capacity. If
you have at least a base-level P4 you don't need to worry about 12
participants.
MATT
-core Intel core 2 Quad processor with 4GB RAM. I
have three systems like this in place at different call centers and
the load is consistent for all three of them. Usually we put less load
on a single server, but these were inbound-only scenarios which is
less load than outbound.
MATT---
On 6/8/08
, and another has SIP agents with calls
coming in over Zap T1 channels.
MATT---
On 6/8/08, Matt Florell [EMAIL PROTECTED] wrote:
Hello,
The load is usually quite high because this is VICIDIAL inbound call
center traffic with full Asterisk-based recording. On a system with
70-80 Meetme rooms
on the IRC channel for Asterisk and asking
around there.
Also if you can somehow get a hold of Matt Fredrickson(who is a very
busy guy) at Digium, he could probably figure this out in a matter of
minutes.
MATT---
On 6/6/08, Remi Quezada [EMAIL PROTECTED] wrote:
Hey,
Is there a way I can
to attach a fax machine to it...
keep in mind that faxing over VoIP is extremely tricky at best, but if your
entire call path is TDM then you shouldn;t have much of a problem.
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Matt Watson
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In short, fxotune adjusts line impedance, where as adjusting gains I believe
is essentially adjusting the amplification / deamplification of the signal.
http://www.voip-info.org/wiki/view/Asterisk+fxotune
--
Matt Watson
http://www.mattgwatson.ca
On June 6, 2008 12:43:51 am Noah Miller wrote
them?
export CFLAGS=
or will they get overwritten?
- --
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Matt Riddell
Director
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http://www.venturevoip.com
Asterisk 1.6 probably isn't going to give you any benefit.
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Matt Watson
http://www.mattgwatson.ca
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brent Davidson
Sent: Wednesday, June 04, 2008 11:52 AM
To: Asterisk Users Mailing List - Non-Commercial
of this, at the cost of a a little
speed due to re-transmissions)
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to contact whomever you bought your Polycom's from to
obtain the most recent versions.
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will do this. I suppose even if its on the same subnet, you could have
something running on either your * box or mysql box that will blow away idle
connections... but that would probably be a little more obvious and you'd
know about it.
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Matt
http://www.mattgwatson.ca
On May 22, 2008 04:42:27 pm Steve Totaro wrote:
PS. Figured I would start with DHADI now.
psst. its DAHDI ;)
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Does your extensions.conf have any more configuration than what you've shown?
If not, then you are lacking dialplan for anything but internal calls.
--
Matt
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of RoLaNd RoLaNd
Sent: Wednesday, May 21, 2008 9:01 AM
To: asterisk-users
circuits.
MATT---
On 5/20/08, Joe Pukepail [EMAIL PROTECTED] wrote:
Is there a way to see error counts on the T1 of a PRI? Hooked up to
asterisk via a digium TE122. Looking for something to make sure I'm not
getting any CRC, framing or other errors on the T1.
Using asterisk 1.4.19 and zaptel
DID from a another SIP provider and re-print all of your materials
(probably incredibly expensive)
--
Matt
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andreas van dem
Helge
Sent: Monday, May 19, 2008 11:49 PM
To: Asterisk Users Mailing List - Non
of
what i think without any real evidence behind my thoughts.
One day i'll have to read up on it.
--
Matt
From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Steve Underwood [EMAIL
PROTECTED]
Sent: Tuesday, May 20, 2008 9:42 AM
To: Asterisk Users Mailing
You might want to see if you can change the IRQ assignments in your servers
bios (might have to turn off the PNP OS Installed option if you have one)
--
Matt
From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Cavanna, Richard [EMAIL
PROTECTED]
Sent
as speed-dial keys.
I don't really see how either of those is not 'human friendly'? Or maybe I'm
just completely off on what you mean.
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.
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-only box and its running bug-free, why screw with it? If
you can get it to over the internet however... security becomes a big
concern.
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http://www.mattgwatson.ca
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were designed for SATA specifically.
Personally, I prefer molex connectors for most things simply because they are
far more secure than SATA connectors (at least the ones i've used).
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Matt
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to
take Asterisk out of the call loop?
The list goes on and on... and every single one of those answers is going to
influence that number for How many calls can my system handle?
--
Matt
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Backeberg
Sent
You'd probably want to run something else to handle your registrations like
OpenSER with that many phones.
--
Matt
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bhrugu Mehta
Sent: Thursday, May 15, 2008 8:31 AM
To: Asterisk Users Mailing List - Non
the
problem after it rained, and the other case was bad carrier equipment
at their shelf, once they moved it to another port on another shelf
the problem disappeared.
Good luck,
MATT---
On 5/15/08, Sherwood McGowan [EMAIL PROTECTED] wrote:
Alright guys and gals,
I'm a little lost, I'm primarily
provide a method of telling Asterisk to wait
for a specific period of time or rings
--
Matt
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeremy Mann
Sent: Thursday, May 15, 2008 12:07 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] *72 Telco
to over 100 conferences on a
similarly equipped server with a very rapid call turnover rate.
MATT---
On 5/15/08, Wai Wu [EMAIL PROTECTED] wrote:
Hi all,
What is maximum number of three party conferences can a quadcore 3GHz
system can handle? All the parties a setup with G.711 codec
Do you mean
What do I need to configure on my * installation so that only registered sip
users can make calls? ?
If so, you are going to need to give a lot more details regarding your current
configuration for you to get any answers.
--
Matt
From: [EMAIL PROTECTED] [mailto:[EMAIL
you could probably get away with doing it directly on the 57i
with no 560M's (or 536M's) too many more phones and you'd need the sidecars
just for the extra buttons I think.
--
Matt
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent
of the chassis.
--
Matt
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sherwood McGowan
Sent: Monday, May 12, 2008 11:29 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] 3U server chassis Digium TE405P?
Gentlemen,
First
FreePBX has this functionality... they call it Confirm Calls
I;m not sure if you can set it on actual extensions, but I know you can set it
on ring groups.
I don't imagine the dialplan for doing it is very complicated if you wanted to
do it by hand.
--
Matt
I just took a quick look at the dialplan that freepbx uses for doing call
confirmation... the dialplan part of it is actually quite simple... its just a
matter of setting the USE_CONFIRMATION varialbe =TRUE.
However, the actual magic looks like it happenes through its dialparties.agi...
which
Poking around the zaptel SVN earlier today i see support was added for an
AEX410 card recently...
I'm going to go out on a limb and assume this is the PCI-Express version of the
TDM410?
Any hints on a general availability date?
--
Matt
portgage (Gentoo's package system).
--
Matt
From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Marco [EMAIL PROTECTED]
Sent: Saturday, May 10, 2008 4:25 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Best Linux
Hello,
You can cluster queues across several servers with VICIDIAL. We have
clients with hundreds of seats taking in hundreds of lines across
multiple Asterisk servers, and the calls are distributed to agents on
all systems.
MATT---
On 5/9/08, Vieri [EMAIL PROTECTED] wrote:
--- Vieri
on many different version of
Asterisk. There is a slight extra load cost for these advantages, but
the ability to cluster many Asterisk servers together greatly
overrides that problem in my opinion.
MATT---
On 5/9/08, Steve Totaro [EMAIL PROTECTED] wrote:
Matt,
Is there any module or code
On 5/9/08, Philipp Kempgen [EMAIL PROTECTED] wrote:
Matt Florell schrieb:
I built VICIDIAL around AGIs, manager interface daemons and
agents in meetme rooms.
Sounds a bit scary. Doing everything in MeetMe rooms just doesn't
feel right IMO.
the ability to cluster many Asterisk
On 5/9/08, Steve Totaro [EMAIL PROTECTED] wrote:
On Fri, May 9, 2008 at 10:25 AM, Matt Florell [EMAIL PROTECTED] wrote:
On 5/9/08, Philipp Kempgen [EMAIL PROTECTED] wrote:
Matt Florell schrieb:
I built VICIDIAL around AGIs, manager interface daemons and
agents in meetme rooms
. I know it will successfully
connect to systems that give multiple results, i;m just not sure if it does
infact failover if the first one doesn;t work.
--
Matt
From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Brian J. Murrell [EMAIL
PROTECTED]
Sent
phones have QoS/ToS settings
under Options - Network - Type of Service
--
Matt
From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Vikas [EMAIL PROTECTED]
Sent: Wednesday, May 07, 2008 7:59 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
I don;t have any answers for you...
But I would love to hear about the results after you get this working and what
road blocks you hit and how you overcame them.
--
Matt
From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Ex Vito [EMAIL PROTECTED
://lists.digium.com/pipermail/asterisk-users/2008-May/211000.html
--
Matt
From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Russell Bryant [EMAIL
PROTECTED]
Sent: Wednesday, May 07, 2008 6:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re
?
Thanks,
MATT---
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will have retired the box being used as a PSTN gateway and
won’t need the IAX2 trunk anymore.
--
Matt
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vinícius Fontes
Sent: Tuesday, May 06, 2008 8:11 AM
To: Asterisk Users Mailing List - Non-Commercial
Google is awesome
http://www.voip-info.org/wiki-Asterisk+AGI
--
Matt
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of chetherston miles
Sent: Tuesday, May 06, 2008 10:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Performance issues
] chan_zap.c: The PRI Call have not been
destroyed
Those are they only 3 relevant lines in the log file.
--
Matt
Disclaimer Statement: This e-mail is confidential and is intended for the
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My bad, I also should of mentioned...
That was on Asterisk 1.4.18 and Zaptel 1.4.10
Using a TE220B
--
Matt
From: Matt Watson
Sent: Tuesday, May 06, 2008 12:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: PRI D-Channel reconfiguration = crash asterisk?
Hello,
I just
mailing
list. By my action of sending a message to a public mailing list, one can say
there is implied consent that it gets distributed to whomever the mailing list
chooses on my behalf.
Thanks,
--
Matt
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf
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