None of those are open source that I recall.
-Kerry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Derek Whitten
Sent: Thursday, June 15, 2006 6:07 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
Is there any setting
in the voicemail that will send the voicemail file in a type that is recognized
on a Blackberry?
Kerry GarrisonDirector of
Technical ServicesTech Data Pros - Orange County's Mobile IT Service
Provider(949)502-7819 x200- [EMAIL PROTECTED]http://www.techdatapros.com
for better phones very quickly.
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
Polycom 501
Linksys spa-941
Polycom 301
Sipura/Linksys spa-841
Grandstream GXP-2000
If you call Digium they will help you get the card configured properly. You
get installation support with any of their hardware products.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Giorgio Incantalupo
Sent: Monday, May 29, 2006 12:33 AM
To:
There are several listed at http://voip-info.org. For Management check out
FreePBX, for recorded calls look for Asterisk Recording Interface.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Kanishka Somaratne
Sent: Saturday, May 27, 2006 9:55 AM
To enable call waitng on
extension 105
database put CW 105
ENABLE
to disable call walting
on extensions 105
database put CW 105
DISABLE
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Khaled
ChehabSent: Saturday, May 27, 2006 1:43 PMTo:
This is the same with VoipJet, some people have good luck but my lines have
been down for 3 months and all attempts at contacting them have gone
unanswered. Hard to believe people still rave about their service.
Here is a hint folks, if the company does not post a customer service PHONE
NUMBER
You can by creating different contexts and using the Administrators function
allow them to modify some of the settings themselves.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Daniel Salama
Sent: Thursday, May 25, 2006 10:42 AM
To:
You will have far better luck asking this in the AAH forum or the FreePBX
site.
-Kerry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Michael George
Sent: Wednesday, May 24, 2006 1:42 PM
To: asterisk-users@lists.digium.com
Subject:
1.2.8 would be the logical next version.
-Kerry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Obelix
Sent: Wednesday, May 24, 2006 1:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] What and When is
Could someone explain to a non-US dummy the following phrases I have
seen on the list.
I can provide you with tier 1 termination 6/6. I can blend or
NPANXX breakout.
We provide US48 termination, blended rate for 1 MOU and above is
.008 with 6/6.
Depends on your location and your requirements. A generic
post like this generally turns into a flame war. Please be MUCH more
specific.
-Kerry
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Crazy
BoySent: Tuesday, May 23, 2006 5:56 AMTo:
on.
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
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Bart Fisher
Sent
We are trying to
setup a sip connection behind a Watchguard Firebox 1000 and it is simply
put...not working. The ports are all forwarded but the packets are not going
out. It is as if the firewall simply ignores SIP packets. Has anyone seen this
or have any idea what the issue could be?
Please hook me up. I have customers dieing for something that works. All our
systems are Asterisk 1.2.7.1
-Kerry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Clint Sharp
Sent: Tuesday, May 16, 2006 11:58 AM
To: Asterisk Users Mailing List -
that at this time.
Otherwise it works great and we will continue to use it for future clients.
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
-Original Message-
From: [EMAIL
Our system is running all of the latest code and freepbx and would send the
attachment to my MDA just fine and I was able to play it without any
problem. My problem was that the MDA is a worthless turd and a complete joke
as a phone. I took it back and switched to the backberry 8700g which has its
How are you trying to do it? ChanSpy or ZapBarge?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Anthony Azzopardi
Sent: Wednesday, May 10, 2006 9:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
I have written up an
guide on how to do bulk provisioning of the Linksys phones and
ATAs.
http://voipspeak.net/index.php?option=com_contenttask=viewid=73
Kerry GarrisonDirector of
Technical ServicesTech Data Pros - Orange County's Mobile IT Service
Provider(949)502-7819 x200- [EMAIL
] On Behalf Of Kerry
Garrison
Sent: Tuesday, May 09, 2006 4:21 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] voipjet down?
I havebent been able to call out in weeks and nobody
returns emails to
[EMAIL PROTECTED
wholesale
termination, go check out voipjet.
On 5/10/06, Kerry Garrison [EMAIL PROTECTED] wrote:
I cant imagine anyone using voipjet as their only or
main provider.
And I'll
say again, 3.9 cents for an ITSP is the most expensive I
have found.
Business grade termination
You could install any number of interfaces but it does not
come with one.
Kerry
GarrisonDirector of Technical ServicesTech Data
Pros - Orange County's Mobile IT Service
Provider(949)502-7819 x200- [EMAIL PROTECTED]http://www.techdatapros.com
From: [EMAIL PROTECTED]
Go into the BIOS and disable every possible device like
USB, COM, Serial, etc. But odds are, you are screwed with that
motherboard.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Antonio
AlmodóvarSent: Wednesday, May 10, 2006 8:01 AMTo:
Asterisk Users
Have you looked at CBeyond? I like their T1 SIPConnect
product.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nitin
GuptaSent: Wednesday, May 10, 2006 7:04 PMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject:
[Asterisk-Users] VOIP provider
[EMAIL PROTECTED] is an ISO image that installed CentOS, Asterisk, FreePBX, and
some other tools. FreePBX is just the web interface.
-Kerry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Zach A
Sent: Tuesday, May 09, 2006 12:38 PM
To: 'Asterisk
I havebent been able to call out in weeks and nobody
returns emails to [EMAIL PROTECTED]
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Julius
Barber :: GringoTel.comSent: Tuesday, May 09, 2006 12:40
PMTo: asterisk-users@lists.digium.comSubject:
Why make a brand new?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
moona ather
Sent: Sunday, May 07, 2006 11:36 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] asterisk management interface
Hi,
I have to make a web-based
and serving my pupose?
thanx!
From: Kerry Garrison [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To: 'Asterisk Users Mailing List - Non-Commercial
Discussion'asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users
Those are reasons for WANTING to create your own, he specifically said I
HAVE to make my own and I wanted to know why he HAS TO create his own when
there are fantastics products already available. There is a huge difference
in saying I would like to create my own and I have to create my own. I
He asked about hard phones not soft phones.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Steve Totaro
Sent: Sunday, May 07, 2006 12:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] asterisk
Go into the BIOS, disable every possible device such as floppy controller,
usb, serial, parallel, etc. If that doesn't work, move card to another slot.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Giorgio Incantalupo
Sent: Friday, May 05, 2006
I have a weird issue
with a system, running freepbx in devices and users mode (same as dozens of
systems) but this one when you hit *70 and get "call waiting activated" it is
not storing the setting in the database. I can manually set CW 900 ENABLED but
then *71 does not disable it. Any
Hard to believe you arent associated with calleveryone.com as I find it hard
to believe that you would be extolling the virtues on one of, if not the
most expensive companies around. $7 a month plus 3.9 cents a minute
domestic, that's pretty much double the cost of anyone else. Customer
service
that the
card will only function properly in one of the three PCI slots. If you get a
motherboard that has more than 3 PCI slots your chances of success are
dramatically higher.
Kerry Garrison
Publisher - http://VOIPSpeak.net
(949) 502-7819 x200 - [EMAIL PROTECTED]
http://www.techdatapros.com
that you
would fork for a PRI card but use cheap winmodems for analog lines. You will
have much better luck tossing the x100p cards and using either SPA-3000's,
a TDM400, or a Mediatrix 1204.
Kerry Garrison
Publisher - http://VOIPSpeak.net
(949) 502-7819 x200 - [EMAIL PROTECTED]
http
You can already do that. You ca specify different access to different users
with the Administrators module.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Rich Adamson
Sent: Monday, May 01, 2006 6:33 AM
To: Asterisk Users Mailing List -
The current versions of IDEFISK use a Windows installer,
wether it is required or not now I dont know.
-Kerry
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bruce
ReevesSent: Monday, May 01, 2006 9:13 AMTo: Asterisk
Users Mailing List - Non-Commercial
some type of functionality, no system is perfect, but it does cover far more
than just a few small businesses.
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
Are you dialing 9 first? It is showing that the digits you
dialed are:
9-770-719-0239
Using your dialplan you should be dialing
1-770-719-0239
Kerry
GarrisonPublisher - http://VOIPSpeak.net
(949)502-7819 x200- [EMAIL PROTECTED]http://www.techdatapros.com
From: [EMAIL
han_zap.c: Got event Hook Transition Complete(12) on channel 1
(index 0) Apr 30 13:29:07 DEBUG[4242] chan_zap.c: Exception on 19, channel
1So what should I have in my dialing plan to let me dial 7707190239
and have it use that exact number? Or do I have to dial 9
first?Thanks, Jim.
On 4/
some of the features they wanted this way
than dealing with the config files.
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
-Original Message-
From: [EMAIL
This is not the right place for help with AAH. Use the AAH forum at sf.net.
If it is just hanging up on users, it is not configured properly.
-Kerry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Patrick Siglin
Sent: Sunday, April 30, 2006 8:25
The client's needs are the mother of invention. We
have a client that currently uses a Cisco Call Manager and one of the features
they love was the Locate-Me function (or follow-me, or find-me, whatever you
want to call it) which basically rings their desk phone a few times then plays a
upgrading from what version?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dave
MorrowSent: Saturday, April 29, 2006 6:11 PMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject:
[Asterisk-Users] RE: Install/Upgrade
Hi all, I was
--
--
*From:* [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
*On Behalf Of *Kerry Garrison *Sent:*
Wednesday, 26 April 2006 10:36 AM *To:* 'Asterisk
Users Mailing List - Non-Commercial Discussion'
*Subject:* RE
AMEN!!
Any consultant that DOESNT take this into consideration
should stick to installing Windows and calling themselves an IT
"Expert".
You can screw up someone's network, mess upa
workstation, hose their email, but you break someone's telephone service there
will be hell to pay.
Kerry
I am ready to pull
my hair out. I cannot seem to get the Polycoms to read the time properly.
Regardless of the server they are pointed to our the offset, i am getting the
correct time, but 24 hours ahead. So for today it is showing Friday April 28 but
with the correct time. Any clues?
Kerry
] Polycom NTP issue
What Polycom phone model?
What firmware version?
What bootROM version?
Older versions of Polycom phones only worked with SNTP time
servers not NTP.
MATT---
On 4/27/06, Kerry Garrison [EMAIL PROTECTED] wrote:
I am ready to pull my hair out. I cannot seem
-Users] Polycom NTP issue
What dns server are you running?
On Thu, 27 Apr 2006, Kerry Garrison wrote:
I am ready to pull my hair out. I cannot seem to get the
Polycoms to
read the time properly. Regardless of the server they are
pointed to
our the offset, i am getting the correct time
Kerry Garrison wrote:
I am ready to pull my hair out. I cannot seem to get the
Polycoms to
read the time properly. Regardless of the server they are
pointed to
our the offset, i am getting the correct time, but 24 hours
ahead. So
for today it is showing Friday April 28
to a
Rhino R1T1 card in the asterisk server. This bundle will hit you for about
$2,000 but will only be half full.
However, you say you have 12 extensions and 12 lines? It is very rare to
have a 1:1 ratio. Usually for 12 people you will see 5-8 lines needed.
Kerry Garrison
Director of Technical
be to use ethereal
to capture one of the ntp request/response pkts and analyze
the content. If that looks okay, then something in the phone
isn't right.
Kerry Garrison wrote:
Here is the sip.cfg file
SNTP tcpIpApp.sntp.resyncPeriod=86400
tcpIpApp.sntp.address=192.168.10.50
PROTECTED] On Behalf Of
Jean-Michel Hiver
Sent: Thursday, April 27, 2006 1:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Looking for input on which way
to gowith smallbusiness setup
Kerry Garrison a écrit :
You will kick yourself up
This is an excellent USB speakerphone
http://voipspeak.net/index.php?option=com_contenttask=viewid=39Itemid=27
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jim
HouserSent: Wednesday, April 26, 2006 6:26 AMTo:
'Asterisk Users Mailing List - Non-Commercial
Yes, I have that device, I wrote the review of it and have
used it regularly ever since. I use it with IDEFISK softphone for the most part
but have tested it with Skype, X-Lite, and SJPhone. I have had it since November
and just love it.
Kerry
GarrisonPublisher - http://GeekGazette.com -
it was a revewiers sample that I begged them to not make me
send it back and they let me keep it.
Kerry
GarrisonPublisher - http://GeekGazette.com - http://VOIPSpeak.net
(949)502-7819 x200- [EMAIL PROTECTED]http://www.techdatapros.com
From: [EMAIL PROTECTED]
[mailto:[EMAIL
When used in this mode they can only detect a ring. Your
best bet would be to put in some overhead paging.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Richard
SchroederSent: Wednesday, April 26, 2006 6:51 PMTo:
asterisk-users@lists.digium.comSubject:
Asterisk is simply a telephony toolkit, so the simple
answer is yes, Asterisk can do this. Also, being a toolkit means there are a
number of ways to accomplish it. You could right PERL, Python, TCL, C, PHP or
numerous other types of scripts that can manage this for you. To see how to do
GPRS or WiFi. That would have been a great benefit to me but its just not
going to happen on a device that barely runs Windows Mobile as it is.
Kerry Garrison
Director of Technical Services
Tech Data Pros - Orange County's Mobile IT Service Provider
(949) 502-7819 x200 - [EMAIL PROTECTED]
http
i have a client that
wants a function that will automatically logout an agent from a queue if they do
not answer a call. This would prevent future calls from being sent to that phone
if the agent forgot to logout. Any ideas?
Kerry GarrisonDirector of
Technical ServicesTech Data Pros -
On a TDM2400 with 3
FXO modules, is there a way to split each line into basically being its own
trunk or another way to pull off the following scenerio:
PBX has 12 inbound
PSTN lines
1,3,5,7 are the 714
phone number hunt group
2,4,6,8 are the 888
phone number hunt group
9-12 are fax
: [Asterisk-Users] Auto Logout from queue
Use the local channel to call the agent first, and if there is no answer,
log them out.
_
From: [EMAIL PROTECTED] on behalf of Kerry Garrison
Sent: Tue 4/25/2006 2:38 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject
-Commercial Discussion
Subject: Re: [Asterisk-Users] Polycom MWI
Kerry Garrison wrote:
Didn't help. Could I be missing something else?
My phone.cfg looks like this:
mwi
msg.mwi.1.subscribe=300
msg.mwi.1.callBackMode=contact
msg.mwi.1.callBack=*97/
And sip.conf for extension
NerdVittles.com has a dialout announcement system article.
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 PROTECTED] On Behalf
I have tried
everything from voip-info and I still cant get the Polycom 501/601 to display
the MWI indicator light. Everything else works just fine. I am using FreePBX set
to users and devices mode. Here is the MWI section of the phonexxx.cfg
file:
mwi
msg.mwi.1.subscribe=""
voicemailbox number (usually extension) in the 1.subscribe field.
Bill
_
From: [EMAIL PROTECTED] on behalf of Kerry Garrison
Sent: Thu 4/20/2006 7:32 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Polycom MWI
I have tried everything from voip-info and I
Linksys SPA-3000 Single Port $90
Mediatrix 1204 4 port Gateway $580
Rhino CB24 24 Port Channel Bank + Rhino R1T1 Card $2000
Kerry Garrison
Publisher - http://GeekGazette.com - http://VOIPSpeak.net
(949) 502-7819 x200 - [EMAIL PROTECTED]
http://www.techdatapros.com
-Original Message
I don't know that distro but with CentOS you have to run alsamixer to turn
on the output and turn up the volume, it is off by default.
Kerry Garrison
Publisher - http://GeekGazette.com - http://VOIPSpeak.net
(949) 502-7819 x200 - [EMAIL PROTECTED]
http://www.techdatapros.com
-Original
Submit a bug report to the FreePBX
team?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sean
GarlandSent: Wednesday, April 12, 2006 8:46 PMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] freepbx
dialing prefix
I need to put a w
in
Has anyone got any
information on bulk provisioning of Linksys SPA-941/94s? There is an overview in
the admin guide but it refers to a different provisioning guide that I haven't
found anywhere.
Kerry GarrisonDirector of
Technical ServicesTech Data Pros - Orange County's Mobile IT Service
Disallow=all
allow=ulaw
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael
StrelnikovSent: Saturday, April 08, 2006 7:25 PMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] Force
codec
Hi, Is it possible to force using codec depends
on
I tried the latest version of Jive over the weekend and I have to say it is
a giant pile of crap. I did this on multiple machines on both Linux and
Windows, and after setting everything up, the moment you add the asterisk
module, all authentication and user setup is lost and there is no way to log
Welcome to the painful world of analog phone lines. Unless
you are using a digital line, there really is no true call progress detection
available. In many situations this is not a problem, where we see this the most
is when you are trying to ring a zip device and a zap channel at the same
Yes.
In Sip.conf you need the following lines:
externip=xxx.xxx.xxx.xxx ; put public ip address here
localnet=192.168.10.0/255.255.255.0 ; edit as appropriate
In your firewall, add the following mappings to your server:
5060-5061 UDP
10,000 - 20,000 UDP
Kerry Garrison
Director of Technical
Use an amplifier off the headphone jack.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Jerry Geis
Sent: Thursday, April 06, 2006 8:30 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] increasing volume level to console/dsp
I
Currently Asterisk will not integrate with Skype. You would
need a provider such as Teliax, Broadvoice, IAX.cc or many others or you can use
hardware devices to connect to traditional phone lines. You didn't say what
broadband phone you have but if its Vonage, there are also issues with
"or you can use hardware devices to connect to
traditional phone lines"
That can be a Digium card, Sangoma card, Linsys
SPA3000, Mediatrix 1204, and several other devices.
Kerry GarrisonDirector of
Technical ServicesTech Data Pros - Orange County's Mobile IT Service
Provider(949)502-7819
Think people will fall for it again next year too?
Hello All
I read in www.sineapps.com have Asterisk 2.0 rewritten C#
and run on
windows, any body could be mail or send to me URL to download.
Thanks
Tin Trung Nguyen
Technical Team
Mobile:
When it works it works great. We have had a few issues
lately but they were resolved fairly quickly.
-Kerry
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Giridhar
Reddy BandiSent: Thursday, March 30, 2006 8:14 AMTo:
asterisk-users@lists.digium.comSubject:
You made some change to something using AMP and it overwrote the
extensions_additional.conf file as it was designed to do. The only safe
place to put customizations is in extensions_custom.conf.
Kerry Garrison
Publisher - http://GeekGazette.com - http://VOIPSpeak.net
(949) 502-7819 x200 - [EMAIL
With Asterisk you can use Analog lines (PSTN) , Digital lines (PRI), or
Internet Telephone Service Providers (ITSP) such as Broadvoice, Teliax,
IAX.cc, and many more.
Kerry Garrison
Director of Technical Services
Tech Data Pros - Orange County's Mobile IT Service Provider
(949) 502-7819 x200
http://www.asterisk.org/features
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Rich Adamson
Sent: Wednesday, March 29, 2006 3:05 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Marketing Materials
FreePBX allows you to set up multiple companies as well as determine what
level of access each user has.
Kerry Garrison
Publisher - http://GeekGazette.com - http://VOIPSpeak.net
(949) 502-7819 x200 - [EMAIL PROTECTED]
http://www.techdatapros.com
-Original Message-
From: [EMAIL
FreePBX is a configuration manager for Asterisk. It is NOT its own version
of Asterisk, it is simply a GUI to manage the config files.
Kerry Garrison
Publisher - http://GeekGazette.com - http://VOIPSpeak.net
(949) 502-7819 x200 - [EMAIL PROTECTED]
http://www.techdatapros.com
-Original
Does anyone know if
a TDM2400 will fit into a Dell 2850?
Kerry GarrisonDirector of
Technical ServicesTech Data Pros - Orange County's Mobile IT Service
Provider(949)502-7819 x200- [EMAIL PROTECTED]http://www.techdatapros.com
___
--Bandwidth and
Look at the Linksys SPA942, it's a great phone for the price.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Radcliffe
Sent: Sunday, March 26, 2006 10:21 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] 3Com Phones
Hi
Its all about how you configure your dialplan. Asterisk doesn't know what a
PSTN or VOIP phone number is. If you want all 08444 numbers to go through a
certain trunk, then you set your dialplan up accordingly.
-Kerry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
If you have Line 1 on hold, and you on a call on Line 2, then hitting TRNF
and hitting Line 1 will transfer Line 2 to Line 1. Same concept as
Conference.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mimmus
Sent: Thursday, March 16, 2006 7:30 AM
If you are in Southern California and would like to attend the Asterisk Users
Group Meeting, it is tonight from 6-9pm at the Heritage Park Library.
Irvine Heritage Park Library(949) 936-404014361 Yale AveIrvine,
CA 92604
Tonight we will be having a demo of SIPX, a review of the SNOM 320
If you go into the BIOS and disable all unneeded devices (serial, parallel,
USB, floppy, etc) then you shouldn't have any problem. I have one in a 15
user setup that is working fine.
-Kerry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Leo Ann
Irvine California, Heritage Park Library on the
corner of Yale and Walnut. The Walnut is just south of the 5 fwy and Yale is
between the Culver and Jeffery offramps. Meeting will run from 6 - 9pm. This
week will feature a review of the SNOM 320, a demo of SIPX, some book giveaways
courtesy
.
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
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Thczv F. Thczv
Sent
On a 55 station
install onto a Cox PRI with a TE110P (Polycom 501 phones) a few users are
complaiining about echo. According to the users, the echo seems to be phone
number dependant. They claim that certain phone numbers have echo while others
dont. Are there any tuning parametes like
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: Asterisk
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
-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
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
650 poweredge (single processory system), with a digium tdm400 card and 4 analog
lines plugged into it.
[Kerry Garrison
sayeth]
The 841 isfine 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 and itsa wonder
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 Technical
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:
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