at support{at}weavver.com
for help.
If you want to keep track of the new company you can follow our
twitter feed at www.twitter.com/weavver.
Here's to a new year, new ventures, and a new president!
Thank you for using Snap and supporting it over the years,
Mitchel Constantin
p.s. Hope to see some
Hey Chris, You could try our Firefox plugin to acheive the same thing in an easier way (with no programming), it'll just work with whatever website you throw at it. -- www.snapanumber.com
On 8/6/06, Christopher Aloi [EMAIL PROTECTED] wrote:
Hello List -Asterisk Version: Asterisk
that way.
Good luck,
Mitchel
On 10/12/05, Samy Antoun [EMAIL PROTECTED] wrote:
Hi,
We have a Data/Voice service supplied through an
integrated T1.
Does anyone know if Digium T1 card will support the
splitting of the Voice and Data?
Regards.
__
Yahoo
feature rich list planned!
Thank you,
Mitchel
On 5/25/05, admin [EMAIL PROTECTED] wrote:
Here are a couple of items I hear people asking for regularly.
- Multi-tenant functionality
- Allow users to change their own preferences via web (call forwarding, MoH,
etc...)
We are two
third parties. Hopefully, by asking up front we
won't be wasting our time on something nobody wants or needs.
Specifically I am asking in the way of GUI's (web-based or not), not
in backend programming as Mark and others have that well under
control!
Thank you for your suggestions,
Mitchel Tom
Matt,
This isn't meant as a flame, rather I'm curious about what other
people think about the following situation...maybe it's just the
philosopher in me, what happens when the load balancer fails?
Thanks,
Mitchel
On Tue, 29 Mar 2005 13:47:58 -0800, Matt [EMAIL PROTECTED] wrote:
you can use
I think from what I remember you have to use agi to do this, so you
can send the command once the call is bridged. I don't know how off
the top of my head though but I do think this is the route to look at.
mitchel
On Wed, 30 Mar 2005 01:48:14 -0600, Paul [EMAIL PROTECTED] wrote:
I have setup
I just thought this link might be interesting to some of you. I know
it's m$ware but please hold back the flames.
http://msdn.microsoft.com/library/en-us/wcetarget5/html/wce50oriDevelopingVoIPPhone.asp
mitchel
___
Asterisk-Users mailing list
Asterisk
This doesn't solve the clear text issue, but how about an access list
based on the mac addresses? That'll secure tftpd a little more.
mitchel
On Mon, 31 Jan 2005 14:04:55 -0800 (PST), David Newman
[EMAIL PROTECTED] wrote:
On Mon, 31 Jan 2005, Gregory Junker wrote:
There should not be any
I've tried it and didn't get it working, I also asked when calling
their support and they said chances are relatively high that they
won't let you do it if you ask nicely (to management) and that it is
for sure disabled by default.
mitchel
On Tue, 11 Jan 2005 18:50:41 -0500, digium-list [EMAIL
.
Mitchel
On Tue, 04 Jan 2005 14:30:43 +1100, Howard Lowndes [EMAIL PROTECTED] wrote:
On Tue, 2005-01-04 at 10:08, Race Vanderdecken wrote:
Each 4-pair wire has 8 wires in the blue wrapper cable.
You only need 2 pairs, 4 of the wires, for 100MB Ethernet.
You could split the wire
I have the same issue, but I've been running this for several months
on a faster (2.8ghz) computer with no loss to data or harm of any
kind, so I would have to say it's probably safe to ignore.
mitchel
On Tue, 21 Dec 2004 16:37:17 -0600, Mark Farver [EMAIL PROTECTED] wrote:
I have a pair
You have to use tftp to do this, use www.voip-info.org it should have
everything you need.
Thanks,
Mitchel
On Tue, 30 Nov 2004 10:20:47 -0800, Michael Levenson
[EMAIL PROTECTED] wrote:
Does anyone have any simple documentation on converting a 7940 to SIP and
making it function with *? I have
This is a great idea, I've got that phone and it really would be an
amazing feature, I have a really nice asterisk system set up in my
house and a $10/month broadvoice line, but linking everything together
would definately be a really nice touch. I've got $15 towards a bounty
;).
mitchel
On Thu
Our prayersanswered? (http://www.phonelabs.com/prd_blue01.asp)
mitchel
On Thu, 23 Sep 2004 12:10:11 -0500, Jay Milk [EMAIL PROTECTED] wrote:
When I installed my first home-PBX three years ago, I was looking at
cellsockets -- devices which will accept certain cellular phones and
provide
Yes you can do it, I've done it with a T100P and an Adtran 612, if you
need specific help let me know, look up adtran on the wiki for a
similar example.
Mitchel
On Fri, 13 Aug 2004 20:16:20 -0300, Daniel Bichara
[EMAIL PROTECTED] wrote:
You can use VoiceTronix boards.
Joe Pukepail
Which hurdles are you talking about specifically? These phones work
great with asterisk (as long as you install the SIP image on them).
mitchel
On Wed, 4 Aug 2004 15:57:11 -0400, Gary Carr [EMAIL PROTECTED] wrote:
Are they still hurdles using Cisco phones with asterisk as mentioned at
http
Scott,
This may not be the point of your message, but just a side note, I
believe that echocancel and echotraining are turned off when a fax is
detected automatically regardless of whether or not they are enabled.
mitchel
On Wed, 04 Aug 2004 14:59:22 -0700, Scott Petersen [EMAIL PROTECTED
This sounds great in my opinion, I am looking forward to hearing more
about it, as far as list etiquette it may be off topic but many of us
do use the ciscos and it could be a great feature we can offer our
clients.
mitchel
On Thu, 29 Jul 2004 09:28:09 +0200, Holger Schurig
[EMAIL PROTECTED
I still don't see why you can't use a script and an array to simplify
this, that way you don't have to work with extensions.conf, just work
on your file, possible php and an array with a loop to check
everything.
-mitchel
On Wed, 7 Jul 2004 20:49:48 -0400, William Suffill
[EMAIL PROTECTED] wrote
Just an idea, I'm sure Asterisk supports an optimized version of what
you want to do, I haven't delved into that area yet, but why not use
an AGI script and some arrays to simplify everything?
-mitchel
On Tue, 06 Jul 2004 18:03:20 -0300, Gelson Dias Santos
[EMAIL PROTECTED] wrote:
brian wrote
Your probably going to get this url (www.voip-info.org) thrown at you
by a few other people too...check there if you haven't already for
more information.
-mitchel
On Wed, 7 Jul 2004 00:45:53 -0400 (EDT), kaiduan xie [EMAIL PROTECTED] wrote:
Hi,
I have some doubts on sip.conf.
1) Can I
I believe this has already been done, it's the o extension. Check
out the samples with the newest cvs of asterisk. That should answer
any questions you have.
-mitchel
On Tue, 06 Jul 2004 08:27:42 -0500, Daniel Jimenez [EMAIL PROTECTED] wrote:
Anyway to make hitting `0` during a voice mail dial
number followed by their passwordone less
step.
On the cisco phone you need to set in the tftp configuration the
messages_uri setting to the number you want dialed..i.e. *123 for
extension 123.
messages_uri: *123
Mitchel
On Mon, 5 Jul 2004 18:46:49 -0700, Randy Bush [EMAIL PROTECTED] wrote
Robb,
I wrote up a small tutorial on setting up the standard tftp server for linux check it out on my site.
http://asterisk.titaniumsoft.net/
Mitchel
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Robert Boardman
Sent: Thursday, May 13, 2004 2:44 PM
I don't know specifically about your question, however you can do a MATCH="*" for all matches that don't match anything (no pun intended).
Mitchel
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Roger
Sent: Thursday, May 13, 2004 4:38 PM
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