On Sun, May 18, 2008 at 12:32 AM, Jay R. Ashworth [EMAIL PROTECTED] wrote:
On Sat, May 17, 2008 at 06:43:51PM -0400, Steve Totaro wrote:
Maybe next they will charge $250 for conference bridge capabilities.
It's a joke to cripple things that can be enabled by flicking a
switch. Your system
On Fri, May 16, 2008 at 7:50 PM, Nicolás Gudiño [EMAIL PROTECTED] wrote:
Hello,
I have finally released the queue stats package to the public.. please go to:
http://www.asternic.org/stats
To get it or see the online demo.
--
Nicolás Gudiño
Buenos Aires - Argentina
Nico, you are the
Glad I was able to foster some good open discussion.
Hopefully DIGIUM will take to heart some of the thoughts expressed here
and end up with a BETTER SOLUTION for ALL.
Steve Totaro wrote:
Inline
On Fri, May 16, 2008 at 11:44 AM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
On Friday 16 May
Hi folks,
Because of our shared interests on this list, I suggest some of you
consider using Twitter.com to follow the people whose comments you
enjoy on the list or those you know by name and reputation. I was
reminded of my own interest in this idea by the fact that a few people
from the
In article [EMAIL PROTECTED],
Mohammad A. Navid [EMAIL PROTECTED] wrote:
I'm implementing a simple calling card feature for testing purpose. I have a
DID number, when I called my DID number and enter the phone number to call,
Asterisk would dial the number for me but the sound was only one
Thanks Tony for you reply.
Do you have any idea why Asterisk require t in Dial command?
Cheers,
Moe
On Sun, May 18, 2008 at 1:14 AM, Tony Mountifield [EMAIL PROTECTED]
wrote:
In article [EMAIL PROTECTED],
Mohammad A. Navid [EMAIL PROTECTED] wrote:
I'm implementing a simple calling card
Dear All
I want to use asterisk for the following Senario and Need help to find a
SAMPLE extension.conf
Incoming call Asterisk GSM Termination Gw
first leg second
leg
What I want to do is putting first call leg
On Sat, 2008-05-17 at 03:37 -0700, bilal ghayyad wrote:
Well, why Digium is still using this kind of power
connector while all new machines does not come with
these types?
As I explained before, Digium recognizes that not all machines have a
spare Molex power connector. For that reason, we
At this stage, I've only seen one machine that didn't come with the old
style power connectors. SATA power connectors may be a standard, but
they haven't (yet?) supplanted the older power connectors.
In fact, most power supplies I've bought recently have had more molex
style connectors than
On Sun, 18 May 2008 00:34:30 -0400, Jay R. Ashworth wrote:
On Sat, May 17, 2008 at 07:07:51PM -0500, Michael Graves wrote:
I work in the broadcast and TV production business. Some time ago a
major company called Quantel created a hardware system called Edit
Box. It was wickedly fast and could
[EMAIL PROTECTED] wrote:
I am new to asterisk and am looking to setup a small office with 4-6 IP
phones and 4 analog lines from the local telco (primary line with HUNT
to the other lines). I am considering purchase of a Digium AEX800.
One of the features that will be important (particularly
On Sun, May 18, 2008 at 07:47:02AM +0300, Tzafrir Cohen wrote:
Is that true for *all* makes of card? I know the Sangomas put it in
ifconfig, but for pick-it-off-with-SNMP-for-Nagios purposes, that will
make my life a lot easier...
I don't really think Sangoma can put this in ifconfig for
On Sun, May 18, 2008 at 02:45:09AM -0400, Steve Totaro wrote:
Let me clarify this a bit with some ramblings if anyone cares to read,
if not move along.
Yes, I was whining a bit.
I always recommend what I think fits best for a customer. I am not in
the Digium Foodchain because of the
On Sun, May 18, 2008 at 11:44:48AM -0400, Jay R. Ashworth wrote:
On Sun, May 18, 2008 at 07:47:02AM +0300, Tzafrir Cohen wrote:
Is that true for *all* makes of card? I know the Sangomas put it in
ifconfig, but for pick-it-off-with-SNMP-for-Nagios purposes, that will
make my life a lot
On Sun, May 18, 2008 at 11:44:48AM -0400, Jay R. Ashworth wrote:
On Sun, May 18, 2008 at 07:47:02AM +0300, Tzafrir Cohen wrote:
Is that true for *all* makes of card? I know the Sangomas put it in
ifconfig, but for pick-it-off-with-SNMP-for-Nagios purposes, that will
make my life a lot
On Sunday 18 May 2008 10:56:00 Jay R. Ashworth wrote:
On Sun, May 18, 2008 at 02:45:09AM -0400, Steve Totaro wrote:
I do however, think that Digium should provide some rough concurrent
call figures and I guess that is how I got off topic on this SwitchVox
tangent. There are some common
Try this...
Setup a music on hold class called myivrhold .
then
Exten = s,1,Dial(Zap/g1/{NUMBEROFGSM}|20|m(myivrhold)
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mohammad
Mirzaee
Sent: Sunday, May 18, 2008 6:57 AM
To:
On Sun, May 18, 2008 at 1:24 PM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
On Sunday 18 May 2008 10:56:00 Jay R. Ashworth wrote:
On Sun, May 18, 2008 at 02:45:09AM -0400, Steve Totaro wrote:
I do however, think that Digium should provide some rough concurrent
call figures and I guess that is
On Sun, May 18, 2008 at 02:02:18PM -0400, Steve Totaro wrote:
Asterisk benchmarking is one of the topics that comes up on the list
frequently and consistently for what, like the last six years (that I
have been involved with Asterisk)? I would call that a Salt
Boulder.
On the wiki there
At 02:25 PM 5/18/2008, Tzafrir Cohen wrote:
Please suggest a test environment
IMHO, it is definitely NOT EASY to come up with a standardized test
without some standardized network configurations and standardized
load generation tools. It is even harder when a non-standard or
niche application
On Sun, 18 May 2008, Steve Totaro wrote:
On Sun, May 18, 2008 at 12:32 AM, Jay R. Ashworth [EMAIL PROTECTED] wrote:
On Sat, May 17, 2008 at 06:43:51PM -0400, Steve Totaro wrote:
Maybe next they will charge $250 for conference bridge capabilities.
It's a joke to cripple things that can be
I noticed that the string alt.asterisk.canary.tweet.tweet.tweet does not
appear in search results.
Hopefully this will help seed it.
See also:
http://www.venturevoip.com/news.php?rssid=1910
http://svn.digium.com/view/asterisk?view=revrev=93805
--
Tzafrir Cohen
icq#16849755
Hi folks,
Would anyone here happen to know of the existence of a Dutch Asterisk
mailing list? If so, where can it be found?
It's not that I'm unable to pose my questions here in English, but I'm
hoping that I may sooner find an answer there to the following question:
What is the most
On 22:46, Sun 18 May 08, Jaap Winius wrote:
Hi folks,
Would anyone here happen to know of the existence of a Dutch Asterisk
mailing list? If so, where can it be found?
Not that I know off.
I can start it if you want.
It's not that I'm unable to pose my questions here in English, but
Hi,
I am just testing Asterisk with a softphone on a fedora box until my ip phones
arrive
and have a basic config so far. I am a bit confused over how to setup the
inbound.conf
file now. It appears as if outbound and demo works so far.
Any hints would be greatly appreciated!
jlc
My sip.conf is
In article [EMAIL PROTECTED],
Moe Navid [EMAIL PROTECTED] wrote:
Thanks Tony for you reply.
Did my suggestion fix the problem?
Ah yes, I just noticed you said it in the subject line.
Do you have any idea why Asterisk require t in Dial command?
Yes, t specifies that the called party may
On Sun, May 18, 2008 at 10:33:28PM +0300, Tzafrir Cohen wrote:
I noticed that the string alt.asterisk.canary.tweet.tweet.tweet does not
appear in search results.
Hopefully this will help seed it.
See also:
http://www.venturevoip.com/news.php?rssid=1910
On Sunday 18 May 2008 16:34:30 Tzafrir Cohen wrote:
On Sun, May 18, 2008 at 10:33:28PM +0300, Tzafrir Cohen wrote:
I noticed that the string alt.asterisk.canary.tweet.tweet.tweet does not
appear in search results.
Hopefully this will help seed it.
See also:
Hi
I just saw this now !
does the microphone and speaker works ?
Can you use it like softphone for recive calls ?
Regards Andrea
C F ha scritto:
TODAY I have managed to hack the iPhone and install Asterisk on it.
Detailed instructions to follow.
Problem solved!
msmtp from the command line was using the config file ~/.msmtprc (the
one I configured) when called from asterisk msmtp uses /opt/local/etc/
msmtprc
so I copied the config in there and voila the emails worked as a champ.
Ciao
Roberto
On May 14, 2008, at 9:14 PM, Roberto
On Sun, May 18, 2008 at 02:02:18PM -0400, Steve Totaro wrote:
If Digium won't supply benchmarking, then let's have a 3rd party throw
down the gauntlet.
Digium vs Sangoma on stock kernels and stock machines with identical
call load, apps, transcoding. No optimization, just stock, and see at
On Sun, May 18, 2008 at 12:32:01PM -0700, Steve Edwards wrote:
I guess I hate to see something I have viewed as such a huge paradigm
shift and disruptive force from selling boxes to selling knowledge.
If I buy a bare DL380 from you, will it cost the same as a fully loaded
DL380
On Sun, May 18, 2008 at 11:57 PM, Jay R. Ashworth [EMAIL PROTECTED] wrote:
On Sun, May 18, 2008 at 12:32:01PM -0700, Steve Edwards wrote:
I guess I hate to see something I have viewed as such a huge paradigm
shift and disruptive force from selling boxes to selling knowledge.
If I buy a bare
On Sun, May 18, 2008 at 3:32 PM, Steve Edwards
[EMAIL PROTECTED] wrote:
On Sun, 18 May 2008, Steve Totaro wrote:
On Sun, May 18, 2008 at 12:32 AM, Jay R. Ashworth [EMAIL PROTECTED] wrote:
On Sat, May 17, 2008 at 06:43:51PM -0400, Steve Totaro wrote:
Maybe next they will charge $250 for
On Sun, May 18, 2008 at 5:17 PM, Joseph L. Casale
[EMAIL PROTECTED] wrote:
Hi,
I am just testing Asterisk with a softphone on a fedora box until my ip
phones arrive
and have a basic config so far. I am a bit confused over how to setup the
inbound.conf
file now. It appears as if outbound
You should probably clean it up and put it up on the wiki. I don't
think
anyone has put up a step-by-step like you did before.
There might be much easier additions/modifications done to it, and it
will
be available to everybody.
Done. No problem - glad to be of service to the open-source
Lee, John (Sydney) would like to recall the message, [asterisk-users] Newbie
Asterisk: Install Asterisk as non-root.
___
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asterisk-users mailing list
To UNSUBSCRIBE or update options
You should probably clean it up and put it up on the wiki. I don't
think
anyone has put up a step-by-step like you did before.
There might be much easier additions/modifications done to it, and it
will
be available to everybody.
Done. No problem - glad to be of service to the open-source
Hi Kashif,
It is possible to do conferenecing using Asterisk.
For this you need to add a patch for conferencing.
You can download this path from the following link:
http://sourceforge.net/projects/appconference
With Regards,
Sukhbir Singh
- Original
my new asterisk server 1.4.19, disconnected the
established calls after the 6 times, retries, when the
quality of Bandwidth between cisco(2600) and
server(asterisk) is not well.
but there is no problem, with asterisk 1.2.7
please help me
___
How can I put a time limit on calls for specific extensions?
For extension 123, for example, I want the phone to disconnect after 5
minutes.
Thank you
Sim
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing
Hi Sim,
You can set the time limit of a call by Adding S(times in seconds)
option in Dial Application.
For Example if u want to disconnect call After 5 mintue for extension
123 then use Dial Application in extensions.conf file as follow:
exten = 123,1,Dial(SIP/abc|10| S(400))
With
Sorry, Do like that
exten = 123,1,Dial(SIP/abc|10| S(300))
In Last mail time was wrong.
- Original Message -
From: Sukhbir Singh [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-CommercialDiscussion
asterisk-users@lists.digium.com
Sent: Monday, May 19, 2008 10:45 AM
Subject: Re:
According to
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Dial#Parameters
the timeout parameter is only in effect if the call is not answered.
The timeout parameter is optional. If not specifed, the Dial command will
wait indefinitely, exiting only when the originating channel
Oops. My bad. I realized that it was the S option that you were talking
about, not the timeout parameter.
Thank you
Sim
Sukhbir Singh wrote:
Hi Sim,
You can set the time limit of a call by Adding S(times in seconds)
option in Dial Application.
For Example if u want to disconnect
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