ts of after-hours time working on setting this up since
then. :) I suspect you and the link on slashdot have bumped up the trend
in interest (not to mention sales of Digium developer kits) quite a bit over
the last few weeks.
Thanks for you, and to the Asterisk crew for an incredible resource.
Login: maint
Password: password
-Kerry
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Angus
ComberSent: Friday, March 25, 2005 7:08 AMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] What is web
login password for [EMAIL PROTECTED]
Hello
I have setup [EMA
We have some good walkthrus at http://www.geekgazette.com. These should
answer most of your questions all in one shot.
-Kerry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Trevor
Tregoweth
Sent: Friday, March 25, 2005 1:52 AM
To: 'Asterisk Users Mailin
Anyone
ever do any integration with a Nortel BCM? If so, are there any real
gotcha's?
Kerry GarrisonDirector of
Technical ServicesTech Data Pros - Orange County's Mobile IT Service
Provider(949) 502-7819 x200 - [EMAIL PROTECTED]http://www.techdatapros.com
Please post to the Users Groups forum at voipspeak.net with all the group
information.
Kerry Garrison
Publisher - http://GeekGazette.com - http://VOIPSpeak.net
(949) 502-7819 x200 - [EMAIL PROTECTED]
http://www.techdatapros.com
> -Original Message-
> From: [EMAIL PRO
Anyone have Mediatrix Unit Manager Express with the 1204 definitions handy?
Kerry Garrison
Director of Technical Services
Tech Data Pros - Orange County's Mobile IT Service Provider
(949) 502-7819 x200 - [EMAIL PROTECTED]
<mailto:[EMAIL PROTECTED]>
http://www.techd
Can anyone recommend a tool that can be used on Windows XP to configure the
Mediatrix 1204?
-Kerry
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.co
to:[EMAIL PROTECTED] On Behalf Of
> Kerry Garrison
> Sent: Saturday, January 14, 2006 12:58 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: [Asterisk-Users] Mediatrix windows-based setup?
>
> Can anyone recommend a tool that can be used on Win
x27;s through the unit manager
> that can fine tune settings, such as setting a dialmap that
> will disable call-waiting, etc. If you don't have the
> original disk, contact me off-list so I can send you a zip of the CD.
>
> -Jon
> I am in no way a mediatrix expert! I h
at
different locations based on the quality and performance of the phone lines.
That has certainly been my experience so far.
Kerry Garrison
Director of Technical Services
Tech Data Pros - Orange County's Mobile IT Service Provider
(949) 502-7819 x200 - [EMAIL PROTECTED]
http://www.techdatapro
Wish I could help, I can tell you I have never been disconnected from a
call on my system that has been running non-stop for months with an
SPA-3000.
All of my settings are documented here:
http://voipspeak.net/index.php?/content/view/24/27/
Kerry Garrison
Publisher - http://GeekGazette.com
What are you talking about? That is the address I used.
-Kerry
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Tomislav Parcina
> Sent: Monday, January 16, 2006 6:22 AM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] Re: Media
In their extension configuration make sure you have
NAT=YES.
Kerry
GarrisonPublisher - http://GeekGazette.com - http://VOIPSpeak.net
(949) 502-7819 x200 - [EMAIL PROTECTED]http://www.techdatapros.com
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ever
Zalaz
Answer is posted here:
http://voipspeak.net/forum/index.php?topic=66.0
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Manuel Casal
> Sent: Monday, January 16, 2006 5:00 AM
> To: asterisk Users Mailing List - Non-Commercial Discussion
> Subject
Go into the BIOS, disable all unneeded peripherals like floppy controller,
serial ports, parallel ports, etc. It should work fine, I have one at a
decent sized installation.
-Kerry
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Jennifer Hales
s not
using any real-time functions. They do use the meet-me rooms fairly heavily.
The system has a TDM400 with 4 FXO ports on it and the phone lines are in a
hunt group that does a rollover on the 5th call to Teliax on the pay as you
go plan which provides 10 additional channels.
Does that help?
The SoCal Asterisk Users Group will be meeting at
the Heritage Park Public Library on the corner of Walnut and Yale in Irvine on
the 3rd Thursday every month. The following dates are already
secured:Thurs Jan 19Thurs Feb 16Thurs Mar 17Meetings are
from 6pm - 9pmIrvine Heritage Park Library(
connector for TDM2400P and it doesn't
> share IRQ in two PCI Slots.
>
> regards,
>
> tron
>
> -Mensaje original-----
> De: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] En nombre de
> Kerry Garrison Enviado el: miércoles, 18 de enero de 2006 5:38
>
No they didn't. They created a custom version of Asterisk that can act as
the voicemail server. That is totally separate than the firmware in the
9000.
Kerry Garrison
Publisher - http://GeekGazette.com - http://VOIPSpeak.net
(949) 502-7819 x200 - [EMAIL PROTECTED]
http://www.techdatapro
neous calls.
On the flip side, you could have 4 "lines" coming into your asterisk server
and have 100 phones with 4 call appearances each. By using Asterisk to
manage the lines, you don't need 400 phone lines to support 100 phones w/4
call appearances each.
Kerry Garrison
Publisher - ht
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of roswel
ajfSent: Tuesday, January 24, 2006 10:56 AMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] asterisk
newbie
I would like to the know following:
1. What is the latest greatest asterisk verision?
You only need to setup ONE account and all four call appearances will work.
-Kerry
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Michael Keyes
> Sent: Tuesday, January 24, 2006 3:41 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asteris
r.
>
> sangoma has a new line of FXO products. they're a bit spendy
> but look very nice.
I would highly recommend the SPA3000 for HOME use since that was what this
post was about. In a business environment I would move up to a Mediatrix
1204 or something like that.
Kerry Garrison
ith you on
what the actual issue is, I don't think it is 100% fair to write off the
SPA3000 in all cases.
Kerry Garrison
Director of Technical Services
Tech Data Pros - Orange County's Mobile IT Service Provider
(949) 502-7819 x200 - [EMAIL PROTECTED]
http://www.techdatapros.com
>
Accounts 3-4 are disabled, Account 1 is the only account.
Thats it. Nothing special. If you have problems, try doing *70.
-Kerry
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
tracinetSent: Wednesday, January 25, 2006 8:57 PMTo:
Asterisk Users Mailing List
How many free intreuppts do you have, its hard to get more than two to work
in most systems. How many PSTN lines are you trying to support?
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> [EMAIL PROTECTED]
> Sent: Monday, January 30, 2006 1:00 AM
I prefer IDEFISK.
-Kerry
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dave
MorrowSent: Monday, January 30, 2006 8:35 AMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject:
[Asterisk-Users] Most Popular FREE SoftPhone for Windows
Hi all
Listen to your voicemail.
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Jeremy Koski
> Sent: Friday, February 03, 2006 9:23 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] Sipura SPA-2002 rings ran
Not a chance, they sell SPA3000's by the truckload. If you only need one
line, then go with the SPA3000, if you need more, I would go with the
Mediatrix 1204.
Kerry Garrison
Director of Technical Services
Tech Data Pros - Orange County's Mobile IT Service Provider
(949) 502-7819 x20
This can easily be accomplished with AMP using the Users and Devices mode.
http://voipspeak.net/index.php?/content/view/49/28/
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Alex Ongena
> Sent: Tuesday, February 07, 2006 8:55 AM
> To: asteris
The best way I have found to use FOP is on a second monitor. That way you
don't need a second PC and it doesn't run behind the receptionists other
applications. Just a decent LCD monitor and a second video card or a dual
head video card, and you are all set.
Kerry Garrison
Director of
ave a dell
650 poweredge (single processory system), with a digium tdm400 card and 4 analog
lines plugged into it.
[Kerry Garrison
sayeth]
The 841 is fine for testing but I would never put
one on a clients desk. The sound quality is bottom of the barrel. Combine that
with the TDM400 card
Using some scripts
that have been posted, we have been able to get paging to phones working quite
nicely. However, with a few [EMAIL PROTECTED] 2.5 installs, (Aserisk 1.2.4) the
phones ring but never pick up. Any ideas on why or how to tweak the scripts to
get the phone paging working again?
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> [EMAIL PROTECTED]
> Sent: Tuesday, February 14, 2006 11:06 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Solution for 1 time blast of
> 200, 000
I dont recall the
SPA-941 playing a stutter tone in the previous firmware but it is driving me
nuts, anyone know where to turn it off?
Kerry GarrisonDirector of
Technical ServicesTech Data Pros - Orange County's Mobile IT Service
Provider(949) 502-7819 x200 - [EMAIL PROTECTED]http://www.tec
Yes you can but it is generally not a good idea nor is it simple to resolve
the additional IRQ conflicts on some machines.
-Kerry
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Thomas Artner
> Sent: Monday, February 20, 2006 4:38 PM
> To: Aste
ach person comes into work, they log into their own call queue. The first
approach is easier to implement.
Kerry Garrison
http://voipspeak.net
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Fraser
Sent: Thursday, November 17, 2005 3:44 AM
To
Does anyone know of an Asterisk Users Group in the Orange County California
area or is there enough interest in starting one up?
Kerry Garrison
Director of Technical Services
Tech Data Pros - Orange County's Mobile IT Service Provider
(949) 502-7819 x200 - [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] suggestions for hard phones?
Does the SPA-941 support stun?
Kerry Garrison wrote:
> My two favorite phones (in order) are:
>
> Linksys SPA-941
> http://voipspeak.net/index.php?option=com_content&task=view&id=41
>
> Grandstream GXP-2000
> h
IAX.cc is what I use for my DID numbers.
-Kerry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeff Ramsey
Sent: Thursday, November 17, 2005 1:23 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] VoIP Gateway Providers
Hi,
Can anyone r
VoicePulse
IAX.cc
BroadVoice
Teliax
Kerry Garrison
Publisher - GeekGazette.com - VOIPSpek.net
(949) 502-7819 x200 - [EMAIL PROTECTED]
http://www.techdatapros.com
Hi,
Can anyone recommend a good reputable VoIP gateway service provider that I
can use with my Asterisk server in wa.us? All I
We have a review of it at http://voipspeak.net, I personally really like it.
Kerry Garrison
Publisher - GeekGazette.com - VOIPSpek.net
(949) 502-7819 x200 - [EMAIL PROTECTED]
http://www.techdatapros.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Whenever I restart Asterisk, I then have to go to each
phone and dial *70 to turn call waiting back on so that the multiple lines on
the phones will ring through instead of getting a busy when the user is only on
a single call. Is there a simple way to have call waiting be On by
default?
-
I have a fairly simple menu structure, three options branch to submenus.
There is a long (several seconds) delay between pressing a key and getting
the next menu. This happens on 2 out of 3 of my menus for no apparent
reason. I am kind of at a loss as to what to look at. Any suggestions would
be ap
oblem?
-Kerry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Don Pobanz
Sent: Tuesday, November 22, 2005 6:01 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Menu Tree Delay
Kerry Garrison wrote:
> I have
5:16 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Call waiting issue
A simple sql command will do this.
> -Original Message-
> From: Kerry Garrison [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, November 22, 2005 1:10 AM
> To: '
I personally prefer a single-box solution with a TDM400
(although I am one of the rare people who haven't had problems with the X100Ps I
have put in). My office uses an SPA-3000 on a phone line that has call forward
on busy to an iax.cc DID line, therefore I speak from experience with both ty
Don't you actually want to do a move instead of a copy? During a copy
Asterisk might actually pull a partial file but a move will not be detected
until the file is 100% in place. Probably not a problem unless you were
writing a very busy call center app.
Kerry Garrison
Publ
Don't you love clients that keep asking for features after an install?
I have a client that is asking about doing distinctive rings for external vs
internal calls. They are using Grandstream GXP-2000 phones which (although a
pain to configure) have 4 ring types. I am guessing that I would need to
http://asteriskathome.sourceforge.net
There are several setup guides at: http://voipspeak.net
The Wiki: http://www.voip-info.org/tiki-index.php?page=Asterisk%20at%20%20Home
The Handbook: http://www.voip-info.org/wiki/view/Asterisk%40home+Handbook+Wiki
Kerry GarrisonPublisher - GeekGaze
[EMAIL PROTECTED] is just
Asterisk with the AMP web GUI and a few other tools. Any size company that
Asterisk can handle would be indentical to what [EMAIL PROTECTED] can handle.
-Kerry
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
ramSent: Friday, November 25, 2005 9:0
any one point me to the some docs
ram
On 11/26/05, Kerry
Garrison <[EMAIL PROTECTED]>
wrote:
[EMAIL PROTECTED] is just Asterisk
with the AMP web GUI and a few other tools. Any size company that Asterisk can
handle would be indentical to what [EMAIL PROTECTED] can handle.
to Ast Server,help need for the
configuration
Hi
yes i have downloaded now compiled
its went to successfully
i would like to suggest people
what are need to done later
to achice lan extentions
and out dialing using IAX provider
ram
On 11/26/05, Kerry
Garrison <[EMAIL PROTECTED]>
using existing FC4
installation
and test
with 4 or 5 clients
by
downloading Source of Ast
thanks
any one point me to the some docs
ram
On
11/26/05, Kerry Garrison <
[EMAIL PROTECTED]> wrote:
[EMAIL PROTECTED] is just Asterisk
Ust go to eBay and find an X100P card. I usually pay $6.95 and usually buy
two at a time if they will ship them together so with shipping it will run
about $10-$12 each. If that will break the budget, I don't know if these is
a better solution.
-Original Message-
From: [EMAIL PROTECTED]
[
isited the Site you have mentioned
in that site, they have mentioned how to config [EMAIL PROTECTED]
but i dont see any config installed with Source of Ast
does any one have step by step config
to configure IAX provider confg
and as well local extensions
ram
On 11/26/05, Kerry
Garris
Depends on the type of headset. The Grandstream GXP-2000 and Liksys SPA-941
have headphone jacks on them, most phones are compatible with Plantronics
univeral headsets.
Kerry Garrison
Publisher - GeekGazette.com - VOIPSpek.net
(949) 502-7819 x200 - [EMAIL PROTECTED]
http://www.techdatapros.com
Client wants to use a *67 feature to block caller id on next call. In the
Wiki I have seen references to this being available but I haven't see any
code to actually make it work. Does anyone have a quick solution for
implementing this type of function?
-Kerry
_
S:
[from-internal]
exten => *67,1,Dial(ZAP/1/*67 )
exten => *67,2,Wait(3)
exten => *67,3,SoftHangup(ZAP/1)
exten => *67,4,Playback(YourCustomPromptStar67IsEnabled)
exten => *67,5,DISA(no-password|from-internal)
Untested, but don't see why it shouldn't work hth
-Origina
You have to have a login to the Cisco site to download the firmware.
-Kerry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Riek
Sent: Tuesday, November 29, 2005 2:02 PM
To: asterisk user list
Subject: [Asterisk-Users] Cisco 7970
I have the same pro
There is a review on the homepage at http://voipspeak.net
It has been available for a few weeks, it is much nicer than the 841!
Kerry Garrison
Publisher - GeekGazette.com - VOIPSpek.net
(949) 502-7819 x200 - [EMAIL PROTECTED]
http://www.techdatapros.com
-Original Message-
From
same time.
Kerry Garrison
Publisher - GeekGazette.com - VOIPSpek.net
(949) 502-7819 x200 - [EMAIL PROTECTED]
http://www.techdatapros.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dakota
Sent: Sunday, December 04, 2005 12:06 PM
To: Asterisk
No it does not user Asterisk. It is a proprietary system based around the
Call Manager products. Linksys sells the system to a service provider who
then offers the service to end users. Basically, LinksysOne is a means by
which service providers can offer a hosted PBX solution.
Kerry Garrison
Asterisk.Net, there is a page on the Wiki.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Vladimir
Montealegre
Sent: Tuesday, December 06, 2005 8:50 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] activex
i have
Title: Can Asterisk accept and relay calls
The simple answer is yes, this can be done. Is there anyone
in Sydney? I dont know.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Skeeve
StevensSent: Thursday, December 08, 2005 8:51 PMTo:
asterisk-users@lists.digium.comSubject: [
Overhead paging is totally possible, there are several articles available on
how to do it. But you cannot have multiple zones today unless you use a sip
device that has autoanswer.
Easiet way to remove that message is to replace the file with one that only
has a split second of silence.
-Or
That article is about shared call appearance. I have this working using
Grandstream GXP-2000's. It's a great new feature.
-Kerry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stas Khromoy
Sent: Friday, December 09, 2005 8:21 AM
To: asterisk-users@lists.
Create a ring group, put all the extensions into the ring group, set your
dialplan to go to the ring group first and then failover to a voicemail
extension.
-Kerry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Derek Whitten
Sent: Friday, December 09, 2
On 12/9/05, Kerry Garrison <[EMAIL PROTECTED]> wrote:
> Overhead paging is totally possible, there are several articles
> available on how to do it. But you cannot have multiple zones today
> unless you use a sip device that has autoanswer.
>
>Why can mutil
The current Teliax pay as you go plan is 10 concurrent channels.
-Kerry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Reynolds
Sent: Friday, December 09, 2005 6:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asteri
With some of the newer versions of Asterisk and AMP, many people have been
having problems getting the Linksys/Sipura SPA-3000 working properly. We
have just posted an all-new guide to getting the SPA-3000 up and running.
http://voipspeak.net/index.php?option=com_content&task=view&id=51&Itemid=28
Dammit, I better go pull my servers out of all of my client's locations
because their production servers have just been rendered unusable. Time to
take their old Toshiba system out of mothballs an.wait a sec, it didn't
do it either, what now? I guess IP Telephony has just died today. Sad, and
i
nquire as to
just how you got it working when it isn't supposed to? This limitation is
proving to be a real thorn in our side and I would just die to get it to
work.
Doug.
-Original Message-
From: Kerry Garrison [mailto:[EMAIL PROTECTED]
Sent: Sun 12/11/2005 1:41 PM
To:
In your sip.conf
externip=10.0.0.1 ; set to your public facing ip
address
localnet=192.168.1.0/255.255.255.0 ; set your network
info
In the extensions make
sure you have NAT=yes
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason
BrashearSent: Monday, December 12, 200
So every person needs to be able to have 7 or more calls on hold at one
time? Usually 2-4 lines is normal as the architecture of an Asterisk system
is generally different from a simple multi-line phone system. You need to
think through the actual usage of the system and how internally Asterisk
mana
We just posted an updated guide to the SPA-3000 a few days ago. The example
uses AMP but all the settings are there:
http://voipspeak.net
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson
Sent: Tuesday, December 13, 2005 5:04 AM
To: Asterisk U
But can they run Asterisk and create an IAX trunk back to your PBX while
running a softphone?
I thought not.
-Kerry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of C F
Sent: Wednesday, December 14, 2005 8:44 AM
To: Asterisk Users Mailing List - Non-Com
Cant
seem to get an answer anywhere else so hopefully somewhere here will have a
clue. With Flash Operator panel, if you have too many extensions or trunks, the
last ones wrap back to the beginning and cover the first ones. I have played
with rectangle sizes with no luck. Anyone have a clue
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Terry H.
Gilsenan
Sent: Friday, December 16, 2005 12:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] FOP button limit?
Kerry Garrison wrote:
> Cant seem
: [Asterisk-Users] FOP button limit?
So how many are you trying to display? A hundred is pretty easily done.
On Dec 16, 2005, at 3:03 PM, Terry H. Gilsenan wrote:
> Kerry Garrison wrote:
>> -Original Message-
>> From: [EMAIL PROTECTED]
>> [mailto:[EMAIL PROTECTED] On
I just converted 5 7960's to the latest SIP firmware, used the Cisco example
configuration files, and nothing custom within Asterisk and my message
lights work fine.
Kerry Garrison
Publisher - GeekGazette.com - VOIPSpek.net
(949) 502-7819 x200 - [EMAIL PROTECTED]
http://www.techdatapro
Everyone should simply uninstall Skype and switch to the Gizmo project
because it interfaces quite nicely with Asterisk.
Kerry Garrison
Publisher - http://GeekGazette.com - http://VOIPSpeak.net
(949) 502-7819 x200 - [EMAIL PROTECTED]
http://www.techdatapros.com
-Original Message-
From
Have it send an email to everyone that there is a voicemail in the box, then
anyone can log into the voicemail system and retreive it
Kerry Garrison
Director of Technical Services
Tech Data Pros - Orange County's Mobile IT Service Provider
(949) 502-7819 x200 - [EMAIL PROTECTED]
Yes you can send and receive calls via Asterisk.
http://voipspeak.net/index.php?/content/view/19/28/
Kerry Garrison
Publisher - http://GeekGazette.com - http://VOIPSpeak.net
(949) 502-7819 x200 - [EMAIL PROTECTED]
http://www.techdatapros.com
-Original Message-
From: [EMAIL PROTECTED
TECTED] On Behalf Of Luigi Rizzo
Sent: Monday, December 19, 2005 3:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk <-> Skype anywhere/anyhow?
On Mon, Dec 19, 2005 at 02:56:30PM -0800, Kerry Garrison wrote:
> Yes you can send an
Can someone describe a simple means of doing this upgrade?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Malcolm
Davenport
Sent: Friday, December 23, 2005 11:24 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] TDM2400P driver change
He
Possible polarity problem with the
jack.
-Kerry
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Filippo
CaroneSent: Friday, December 23, 2005 2:16 PMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] tdm400 fxo
problem
Hi,I'm experiencing a very weird behaviour
ly
seen them in the BES setup. If there is something different that I don't
know about, please let me know.
Kerry Garrison
Publisher - http://GeekGazette.com - http://VOIPSpeak.net
(949) 502-7819 x200 - [EMAIL PROTECTED]
http://www.techdatapros.com
-Original Message-
From: [EMAI
Asterisk
server. If I can borrow a Blackberry from my client for a few days, I will
try it.
Kerry Garrison
Publisher - http://GeekGazette.com - http://VOIPSpeak.net
(949) 502-7819 x200 - [EMAIL PROTECTED]
http://www.techdatapros.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
How would you play a file to a line that hasn't been picked up? You have to
pick up the line in order to do anything with it.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Arik Funke
Sent: Tuesday, December 27, 2005 1:33 PM
To: asterisk-users@lists.digi
Some users with Blackberry's cant play .wav files, is there a way to save
the voicemail to save as another format like mp3?
-Kerry
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The SC430 will experience unusable call quality with a
TDM400P due to IRQ Sharing problems. If you have some magic to get around this,
please share because everyone I know that has tried using an SC430 has given up
and switched to other platforms.
-Kerry
From: [EMAIL PROTECTED]
[mailto:[
Kielhofner
Sent: Wednesday, December 28, 2005 3:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] What is the best Dell Machine for Asterisk?
Kerry Garrison wrote:
> The SC430 will experience unusable call quality with a TDM400P due to
> IRQ Sharing pr
ation with the TDM2400 so I am
kind of at a loss.
Kerry Garrison
Director of Technical Services
Tech Data Pros - Orange County's Mobile IT Service Provider
(949) 502-7819 x200 - [EMAIL PROTECTED]
http://www.techdatapros.com
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scussion
Subject: Re: [Asterisk-Users] What is the best Dell Machine for Asterisk?
Kerry Garrison wrote:
> The SC430 will experience unusable call quality with a TDM400P due to
> IRQ Sharing problems. If you have some magic to get around this,
> please share because everyone I know that
about how we can work together.
Kerry Garrison
Director of Technical Services
Tech Data Pros - Orange County's Mobile IT Service Provider
(949) 502-7819 x200 - [EMAIL PROTECTED]
http://www.techdatapros.com
___
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ation with the TDM2400 so I am
kind of at a loss.
Kerry Garrison
Director of Technical Services
Tech Data Pros - Orange County's Mobile IT Service Provider
(949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com
___
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hope this helps.
I had EXACTLY the same symptom when I was trying to get an X100P clone to
work yesterday. Bumping the toneduration parameter in zapata.conf to 200
milliseconds cured the problem.
Roger
Kerry Garrison wrote:
>Asterisk 1.2.1
>Updated the TDM2400 driver over the weekend
>
You REALLY don't want to have call waiting on a line going into any PBX. You
are only asking for problems. My basic home setup is an SPA-3000 but the
PSTN line only has call forward on busy, when busy, the number is forwarded
to a DID at iax.cc.
Kerry Garrison
Publisher - http://GeekGazett
ng a lot to the table, with sales
> approaching $50MM and aggregating, on average, around 2200 new
> customers per month over the past year.
>
> Thanks for the email!
>
> Regards,
>
> Cory Andrews
> Senior Partner
> +++
> VOIPSupply.com
> 454 Sonwil
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