Hello,
I am using an incoming iax provider to bring calls to my server. When
an incoming call comes in, the calling party does not hear a ring
tone. I figured that this was no big deal, and that I just needed to
enable the "r" flag to dial. This has not fixed the situation though.
Just to try
You need to make sure that you install the asterisk-config package as well.
--Brian
On 11/16/06, blackwater dev <[EMAIL PROTECTED]> wrote:
I have an Ubuntu system and went into Synaptic and checked asterisk for
installation. Once installed, I started it with /usr/sbin/asterisk -vvvgc
and got
You need to make sure that you install the asterisk-config package as well.
--Brian
On 11/16/06, blackwater dev <[EMAIL PROTECTED]> wrote:
I have an Ubuntu system and went into Synaptic and checked asterisk for
installation. Once installed, I started it with /usr/sbin/asterisk -vvvgc
and got
On Mon, Nov 13, 2006 at 12:46:14PM +0100, nik600 wrote:
> Hi
>
> i have an application developed with bayonne.
>
> Recentely i'm experiencing some problems and i am planning to migrate
> to asterisk.
>
> I would like to know if i can do these things whit asterisk:
>
> - IVR integration with dat
You can just seperate multiple phones with "&" in the Dial command,
as the voip-info wiki page shows:
http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial
On Tue, Oct 31, 2006 at 10:28:32AM -0700, Stephen Bosch wrote:
> Hi, folks:
>
> I need to be able to have a single DID ring multiple remote (
I have seen this if you do not include -c1 for stereo audio files.
--Brian
On Tue, Oct 10, 2006 at 02:31:59PM -0400, Bill Merriam wrote:
> I am testing 1.4 and am having trouble with the sound files. The gsm
> files are much larger than they used to be. Sox (12.18.2) plays them
> back really ss
Absolutely, the MeetMe command just takes a conference number. You
could have as many extensions invoke it as you would like.
--Brian
On Tue, Oct 10, 2006 at 11:37:49AM +0100, Mike Williams wrote:
> Hi,
>
> Is it within the realms of possibility to have a single conference with
> multiple num
I know that this is a silly suggestion but you should check to make sure
that you actually have the cdr_mysql module, because at some point (I
believe at the 1.2 release or shortly thereafter), it was moved into
asterisk-addons.
--Brian
On Tue, Oct 10, 2006 at 08:31:43AM +0200, Garth van Sittert
I don't know if there is a better way to do this with meetme itself, but
you could use the manager interface (or even the file method described
in http://www.voip-info.org/wiki/index.php?page=Asterisk+auto-dial+out).
You can pass a Data argument with the filename, to an extension that
simply plays
As I recall, you nee to make sure you run this script with the DeadAGI
command, not just AGI. This will make sure that the dial command will
return to your script only after it is done.
--Brian
On Sat, Oct 07, 2006 at 10:45:10PM -0700, Ali wrote:
> So what should I do?
>
>
>
> On 10/7/06, Tza
Check out the MySQL realtime module. It is in asterisk-addons. You can
read more about this at:
http://www.voip-info.org/wiki/view/Asterisk+RealTime
You will need to compile the add-ons yourself though (unless your
distribution includes a package for them).
--Brian
On Wed, Oct 04, 2006 at
On Mon, Sep 25, 2006 at 12:52:43PM -0600, Douglas Garstang wrote:
> > -Original Message-
> > From: Brian Rogan [mailto:[EMAIL PROTECTED]
> > Sent: Monday, September 25, 2006 12:40 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject
Doug,
Why do you want to do this to begin with? I think the best solution is
to use the realtime stuff, and build your own management tools, which
would allow you to do this (you could drastically cut the complexity
with the right tools). Even if you could run them together, how
would you put ev
I don't have experience using the 480i CT, only using the 9112i, so you
should take what I say with a grain of salt.
I have been nothing but impressed with this phone. In terms of being
friendly with *, they dedicate a section of their manual to asterisk
configuration, which makes things go quite
ing if
there is anything that I can do to debug this problem.
Thanks a lot,
--Brian
On Fri, Sep 15, 2006 at 09:51:01AM -0400, Brian Rogan wrote:
> Hello,
>
> I am trying to write an AGI application that will transfer the caller to
> a phone number on certain conditions. From what I u
On Sun, Sep 17, 2006 at 01:25:10PM +0200, Timothy Parez wrote:
> We'll have about 10 internal phones.
> One of the phones should be like a central station, where all other
> calls can be monitored (if possible)
> and from that phone the user should be able to press a button to take
> over a call
Hello,
I am trying to write an AGI application that will transfer the caller to
a phone number on certain conditions. From what I understand (from the
astcc application and voip-info wiki), I should just be able to EXEC the
dial command. I'm having problems with this though. I send asterisk
the
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