Just recently a client of mine took a lightning hit, which in turn blew
out their Digium TE411P board. This just so happened to be their main
office where their call center was located. We had a backup card on
hand, but this still meant downtime for the client until we got out
there to
the app to exit with a non-zero status,
Because: [EMAIL PROTECTED]|g(2) causes asterisk to hard crash.
And trying to use g2 in either case doesn't work either.
Cullin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf Of Dustin Wildes
Sent: Wednesday, June 28
Okay, that would make sense if you wanted 2 different volume levels for
the messages.
Just typically if the email attachment has low volume, usually the
message on the phone is low too.
In any case - you have 2 options now for adjusting volume. :-)
Aaron Daniel wrote:
The other problem is
Why use an application like sox - when you can make the voicemail
application do it natively:
exten = s,1,Dial(SIP/100,10)
exten = s,2,Voicemail([EMAIL PROTECTED]|ug(10))
The key is the g(10) parameter:
From the 'show application voicemail':
g(#) - Use the specified amount of gain when
/valet/
Enjoy!
Dustin Wildes
VecSector, LLC
1.912.422.7082 x101
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to be any progress.
http://forums.digium.com/viewtopic.php?p=23974#23974
I will be posting the code later today.
--Dustin Wildes
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taken care
of) - so all you would have to bring to the table is criticism! ;-) *JK*
Dustin Wildes
VecSector, LLC
email: [EMAIL PROTECTED]
1.912.422.7082 x101
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.
Along with your salary requirements.
You'll be working with PhoneCALL, so be sure to look over the code first
before applying.
http://www.vecsector.com/phonecall
Thanks everyone!
---
Dustin Wildes
President
VecSector, LLC
1.912.422.7082 x101
email: [EMAIL PROTECTED
start shameless plug
I invite you to look at our interface PhoneCALL. I designed it from
the ground up, all 100% php. If anything, just to learn how it's done.
http://www.vecsector.com/phonecall
end shameless plug
Thanks!
--Dustin
moona ather wrote:
As I know only php and no
) to build an
interface - and is why alot of projects on here come/go. We want to
work together with other businesses so we ALL make money!
Thanks, and sorry for the marketing lecture. ;-)
Dustin Wildes
Kerry Garrison wrote:
Those are reasons for WANTING to create your own, he specifically
If anyone would be interested in a 1099, work from anywhere job with
PHP/MySQL programming - please contact me.
Most jobs would be centered around Asterisk and PhoneCALL GUI, so
in-depth knowledge in both is desired.
Send Resumes and pay requirements to:
[EMAIL PROTECTED]
Thanks!
Dustin
Dinesh Nair wrote:
On 02/01/06 09:29 Damon Estep said the following:
Ok, now lets go for 5000 of them. 160kbps*5000=80kbps or 800mbps -
full duplex.
Have you ever seen a NIC or switch that can run GigE full duplex at 80%
utilization and not at least start to fall apart?
and versatile
GPL GUI out there.
We're almost done with the Tenant User portals as well, to make
setup even easier for the junior admin for the regular user to
administer their phone services.
I'd love any feedback/suggestions you'd have on it!
Dustin Wildes
VecSector, LLC
Steve Totaro wrote:
I don't see how any of these are better than AMP or [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] . What features do any of these have that [EMAIL
PROTECTED]
mailto:[EMAIL PROTECTED] doesn't? The only problem with [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] is the
name.
Doug Lytle wrote:
They must have fixed it, because I just logged in. Looks nice, will
have to give it a try this long holiday weekend.
Doug
Hey Doug - yes, it was fixed this morning - we'd purged all the old demo
data forgot to re-create the demo account.
We've already gotten quite a few
=Downloads
Thanks!!
Dustin Wildes
INFO on 2.7-RC1
New System Features include:
-
-- Better script handling of Arguments
-- New Queue Configuration
-- New Conference(MeetMe) configuration
-- Defaults configuration for SIP/IAX/Voicemail
-- Easy to use Installation
Stay tuned for PhoneCALL's 2.7-RC1 release scheduled soon. We're adding
a new Security Manager that allows you to set the levels of editing for
your users/admins.
Chris Bagnall wrote:
Hello all,
I'm trying to find an Asterisk web interface (or windows gui interface) to
asterisk that won't
contact me I'll get you started on a
language file.
We'd love to get as many languages as possible! :-)
Thank you for your time!
Dustin Wildes
VecSector, LLC
[EMAIL PROTECTED]
http://www.vecsector.com/phonecall
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Hey Arne!
My project 'PhoneCALL' http://www.vecsector.com/phonecall does pretty
much the same thing as you are describing - stores the configs in mysql
then submits the changes to flat files reloads asterisk on
completion. For me my clients - there hasn't been any noticeable
difference, as
Angus Comber wrote:
Hello
I am using a VIA Epia ME6000 with a 600MHz Eden Fanless CPU. Is this
likely to be enough power for a 8 extension system with 6 external
pstn lines?
How important is cpu? Is there some measure, eg xMHz CPU per
extension or something benchmark?
I have installed
will be ready.
Please feel free to contact me if you have any questions.
Thanks!!
Dustin Wildes
VecSector, LLC
Matthew Crocker wrote:
Are there any switchvox/fonality type Asterisk based PBXs where I can
buy just the software? I don't want to buy their 'bundles' that come
with junky PC
,
meetme control a queue manager for the next version 2.7.
I'm sure it's not perfect, but I think with more feedback like you are
requesting, we can really iron this out to be a nice complimentary GUI
for Asterisk.
Feedback is most welcome, either onlist or offlist.
Thanks!
--Dustin Wildes
Angus - I have several mini-itx systems based on the Epia MII6000
(fanless) system.
They all run great, and I have no problems. I also run 'mpg123' with
several mp3s.
I run it in an embedded configuration (in house).
However, I do remember one board that I got where the heatsink on the
CPU
!
Dustin Wildes
VecSector, LLC
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feature requests, and all feedback is welcome.
Thanks!
Dustin Wildes
VecSector, LLC
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as simple as AMP but it does seem to be more powerfull.
Keep up the good work and write a manual!
Mark
Dustin Wildes wrote:
Hello All!
Just a notice that our PHP/Smarty-based GPL version of PhoneCALL
version 2.6.1 has been released, and is the current stable release.
http://www.vecsector.com
Michiel van Baak wrote:
On 11:21, Tue 16 Aug 05, Dustin Wildes wrote:
Thanks Mark!
You're right - this version is intended for the 'advanced' admin, one
who is very knowledgable with Asterisk, but we are working on
simplifying the interface in the next revisions that will make
.
have fun,
Klaus
--
Klaus-Peter Junghanns
On Mon, 2005-08-08 at 21:55 -0400, Dustin Wildes wrote:
Has anyone written a php/perl or a hack to the 'hint' function in
Asterisk that will let you monitor a MeetMe conference?
So if anyone was in a conference, I could have a button light up on my
, Dustin Wildes wrote:
This would be absolutely perfect!
I found the app_devstate.so in the 'bristuff' package. Has anyone
ported over the app_devstate.c to work with HEAD? Or do you have to use
this with bristuff's patched version of asterisk?
Klaus-Peter Junghanns wrote:
Hi,
take
for the info Klaus!
--Dustin
Klaus-Peter Junghanns wrote:
hmm..extracting it from:
http://www.junghanns.net/asterisk/downloads/bristuff-0.2.0-RC8f-CVS.tar.gz
shouldnt be rocket science. ;-)
good luck,
Klaus
On Tue, 2005-08-09 at 09:36 -0400, Dustin Wildes wrote:
I had noticed the 'devicestate.c
Colin E. McDonald wrote:
The new update seems to have cured my issue with calls intersecting and
Zap lines not being hung up after the user terminates the session but
now I am having sound issues with all of my phones. The sounds seems to
be very low on all of them and there is a definite
Has anyone written a php/perl or a hack to the 'hint' function in
Asterisk that will let you monitor a MeetMe conference?
So if anyone was in a conference, I could have a button light up on my
Snom 360?
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Adam Vocks wrote:
In an order to save money, I would like to use a PRI that we have
going to one of our dial-up modem banks (We are an ISP.) During
business hours these channels are idle and during our peak internet
times, we are closed. Sounds too good to be true, but I thought I
would
[EMAIL PROTECTED] wrote:
Hallo,
we have started playing with asterisk about one month ago, and we do like
very much what we are experiencing.
Now we would like to take some step further towards standardizing
installed modules, functionalities, tools etc.
The wall we are facing now is:
[EMAIL PROTECTED] wrote:
I've google'd this to death, is there a simple way to make MWI work from *
for my Cisco phone ??? Examples ???
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Dan Perik wrote:
Dustin Wildes wrote:
I feel there is nothing wrong with having a web-based configuration
utility, if set up correctly. Look at the WRT54G Linksys router, plus
other countless devices that use an embedded browser for configurations.
Just a nitpick
We are working on finalizing a production release of our PhoneCALL
product, a GPL php/smarty configuration GUI for Asterisk:
http://www.vecsector.com/phonecall
I feel there is nothing wrong with having a web-based configuration
utility, if set up correctly. Look at the WRT54G Linksys
Chris Mason (Lists) wrote:
I thought I saw a Soekris embedded in the Digium booth photos, can you run
Asterisk on one of these? How? I'd be interested in it for a back pbx, given
the reliability. In fact, might want to move my home pbx to this also.
Chris Mason
-in-one piece of
equipment.
Kristian Kielhofner wrote:
Dustin Wildes wrote:
Chris Mason (Lists) wrote:
I thought I saw a Soekris embedded in the Digium booth photos, can
you run
Asterisk on one of these? How? I'd be interested in it for a back
pbx, given
the reliability. In fact, might
Not a problem Kristian! :-)
Same here!
Comments below:
Kristian Kielhofner wrote:
Dustin Wildes wrote:
Maybe my point was missed.
Hardware wise - a VIA MII EDEN based board will greatly outperform a
Soekris system, which is why my embedded platform is based on the VIA
hardware instead
I found it was worse when using the G726 or G723 codecs, but if you used the G711
codec, the DTMF echo was hardly noticable. I was using the latest image: 2.0.9d
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Rich Adamson
Sent: Wednesday, July 28, 2004
I currently received two of the sayson 390 480 phones. I like the style of the
phones, but was wanting some feedback from other users.
My phones seem incredibly slow whenever connecting to voicemail. I've added the
security settings to my adsi.conf file re-downloaded the script to the phone.
] Behalf Of DUSTIN WILDES
Sent: Friday, March 19, 2004 6:26 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] ADSI slow?
I currently received two of the sayson 390 480 phones. I like the style
of the phones, but was wanting some feedback from other users.
My phones seem incredibly slow whenever
I have a 3802 and I was told by their support the SIP version doesn't support CallerID
from the PSTN side.
Also - mine was freezing occasionally on calls. I sent several debugs to technical
support, but didn't get any response.
My experience has not been that pleasant - please let me know what
This
has been an occasional problem with us as well (around 45
users).
If
anyone has a fix - please share! :-)
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Darren
NickersonSent: Friday, February 27, 2004 10:36 PMTo:
[EMAIL
Cool - thanks Florian. I'll give that a try.
I guess there isn't a away to just pass the native flash via SIP yet?
-Original Message-
From: Florian Overkamp [mailto:[EMAIL PROTECTED]
Sent: Monday, January 19, 2004 2:30 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] ATA-186
Hello all!
I have an FXO port on a cisco router that is directly connected to our PBX.
Our ATA-186 (firmware version 3) registers with asterisk, which connects to our cisco
router's fxo port to give me a dialtone on our PBX from the ATA.
How do I pass the flash button to the PBX? It seems the
I think this is a great addition!!!
Thanks for the app!
-Original Message-
From: Steven Sokol [mailto:[EMAIL PROTECTED]
Sent: Friday, November 21, 2003 3:32 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Asterisk Call Manager for Windows 0.0.1
(Alpha)
If anybody is interested, I
Your inheritied context is including the exten = h,... for dial-out internal
because your sip.conf is pulling both via your local context.
Something like this should fix it:
[local]
include = extensions
exten = _9,,1,Goto(dial-out,${EXTEN},1)
That will only execute the exten = h,... entry
Title: Message
I feel
this needs to be a separate application in Asterisk, like
app_sipintercom
The
application would connect to all available auto-answer SIP phones, play a short
frequency tone for the intercom alert, only allow one-way streaming to the
phones, then disconnect all phones
but if you did the software wuld need to be smart enough to
know which groups of extensions could be in a multicast and
whci need to be bridged. Basically check to see if the SIP phone
are on the same subnet.
--- DUSTIN WILDES [EMAIL PROTECTED] wrote:
I feel this needs to be a separate application
Hey guys - hate to beg, but my Cisco ID has expired (yes - I'm renewing) and I can't
get the latest ver 6.0 image for my SIP Phones - could anyone send me the .scp .bin?
Of course this email never happened! :-)
Thanks!!
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by a beep and
silence).
So, basically you need to decide 1) what is audio and what is background
noise and 2) how long should there be audio followed by silence.
On Tue, 2003-10-28 at 19:25, Alastair Maw wrote:
On 27/10/03 21:57, DUSTIN WILDES wrote:
Does anyone have any recommendations
Why not just ask them to press-any-key ?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
DUSTIN WILDES
Sent: Thursday, 30 October 2003 12:30 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Answering Machine Detection
Thanks for all
Title: FW: Voice/Data mixed routing over Digium E1/T1 Card
I'm
currently using this setup for a channelized T1 for voice and
data.
First
9 channels of the T1 are voice - the rest are data for
internet.
Works
extremely well!
This
is being used for a production server that receives/places
Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of DUSTIN WILDES
Sent: Wednesday, October 29, 2003 2:23 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] FW: Voice/Data mixed routing over Digium E1/T1
Card
I'm currently using this setup for a channelized T1 for voice
Does anyone have any recommendations on implementing Answering Machine detection for
call generation programs?
What I would like is * to determine what picks up the other line (Answering Machine,
Voicemail, or Human) to determine which action to take. For example:
If * detects Answering
I've been digging around with some cisco engineers for about a week I finally got an
encouraging response to the Auto-Answer issue with the SIP Phones.
Here is their reply:
===
== FROM CISCO ==
===
Auto-Answer feature is introduced in SIP IP Phone 6.0 version. This
Yes
-Original Message-From: Bartosz Jozwiak
[mailto:[EMAIL PROTECTED]Sent: Thursday, October 16, 2003 2:31
PMTo: ASTERISK USERSSubject: [Asterisk-Users]
consultative transfer cisco
Hello,
Is it possible to
makeconsultative transfer on Cisco 7940 and 7960 phones?
Does anyone have the .bin file(s) to convert a Cisco 7940/7960 back to the CallManager
image?
I want to start playing around with the chan_skinny addition, but it seems the .exe's
from cisco want to open a connection to a SQL server or CallManager (which I don't
have).
with some tweaking to get a phone back to
Skinny without having a CallManager.
Good luck. If you need a pointer or two, drop me a line at
[EMAIL PROTECTED]
Matt Hardeman
PaperSoft
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of DUSTIN
WILDES
Sent: Friday
This is Off-Topic for Asterisk, but I wanted to get some feedback on headsets for
Cisco 7940/7960 phones.
We have about 10-20 people who wants/needs a headset for their phone was hoping to
collect some real-world input.
Thanks!!
___
Asterisk-Users
with a soldering iron.
- Justin
On Wed, 3 Sep 2003, DUSTIN WILDES wrote:
This is Off-Topic for Asterisk, but I wanted to get some feedback on headsets for
Cisco 7940/7960 phones.
We have about 10-20 people who wants/needs a headset for their phone was hoping to
collect some real-world input
.
(this was cvs as of last friday)
DUSTIN WILDES wrote:
Hi all!!
Does anyone have a short example or even better - a working AGI script that uses
GET VARIABLE' from a /var/spool/asterisk/outgoing call that uses SetVar?
Here's what I've tried with no luck so far:
sample.call
Did anyone have a way to make codec negotiation work with Asterisk?
This is something I would love to have working as well.
I won't need PSTN - G729 mixing. Just SIP - SIP using G729 for calling remote
offices via VPN, but everything else use G711.
-Original Message-
From: Brancaleoni
If this is through your Telco, they may have turned on the Callerid-Name field along
with your number.
I had mine turn on the Callerid-Name field for us.
-Original Message-
From: Andy Powell [mailto:[EMAIL PROTECTED]
Sent: Sunday, June 15, 2003 3:25 PM
To: [EMAIL PROTECTED]
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