I purchased a Grandstream 4 FXS Gateway and my * extensions are
not working. I disable the special features and changed the DIAL to
{X*#+} but not luck.
I can dial any other number, receive calls and so on. This is the
only thing that seems to be an issue.
Has anybody found a way around th
On 11/13/06, mitcheloc <[EMAIL PROTECTED]> wrote:
For about a year and a half now I've had Asterisk set up to unlock my
front door at my house when calling a certain number. I locked it down
by using caller id (not the most secure, but hey nobody knows the
phone number to my door). Speed dialing
On 11/13/06, Steven <[EMAIL PROTECTED]> wrote:
I have been using http://www.snapanumber.com/ 's Windows tray utility, and it
works great.
--
--
Steven
http://www.glimasoutheast.org
"Ondrej Valousek" <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECTED]
Hi all,
I am interested in inte
On 11/12/06, Yu Safin <[EMAIL PROTECTED]> wrote:
I am running the most recent asterisk 1.2.13 on a Fedora 3.0.
When I go into asterisk (asterisk -r), defaults to verbose 3 and I get
a stream of messages:
Remote Unix connection
Remote Unix connection disconnected
...
...
(keeps on repeati
I am running the most recent asterisk 1.2.13 on a Fedora 3.0.
When I go into asterisk (asterisk -r), defaults to verbose 3 and I get
a stream of messages:
Remote Unix connection
Remote Unix connection disconnected
...
...
(keeps on repeating).
I went to google and searched on "asterisk Remote Unix
On 9/30/06, Tim Panton <[EMAIL PROTECTED]> wrote:
On 29 Sep 2006, at 19:20, Yu Safin wrote:
> Hi, I am a salesman currently using asterisk to contact my customers.
> So far, I have asterisk connected to two PSTN analog lines where I
> only receive phones calls.
> The
Hi, I am a salesman currently using asterisk to contact my customers.
So far, I have asterisk connected to two PSTN analog lines where I
only receive phones calls.
Then, I have asterisk connected to a VoIP service company for
terminating my phone calls.
I also kept one PTSN phone line to place cal
On 1/24/06, Lee Archer <[EMAIL PROTECTED]> wrote:
Thanks, I've got it running on my test box but didn't know if there was
any global objection to using it. I've had a few funnies with it but
that might be down to Supermicro and P4's with the EM64T thing.
Regards
Lee
-Original Message-
On 10/20/05, Dave Cotton <[EMAIL PROTECTED]> wrote:
> On Thu, 2005-10-20 at 03:31 -0400, [EMAIL PROTECTED] wrote:
> > SuSE Linux Enterprise Server 9
> > Asterisk 1.2.0 beta1
> >
> > I am trying to build and install Asterisk on SuSE. I started with a
> > fresh full installation of SuSE.
> >
> > The
am I correct in believing that only goiax.com offers free DID's?
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On 10/19/05, Steve Totaro <[EMAIL PROTECTED]> wrote:
> YES
>
> - Original Message -
> From: "Frank Kostin" <[EMAIL PROTECTED]>
> To:
> Sent: Wednesday, October 19, 2005 8:58 AM
> Subject: [Asterisk-Users] SIP to IAX
>
>
> Hello everybody,
> Is it possible to route "any" incoming SIP call
>
has anybody tried to register with Sixtel to obtain a DID?
I signed up 9/27 and I am yet to receive my DID.
Also, how do I change my aix.conf to connect to Sixtel?
I have a userid and password but I don't have details about all the parameters.
___
--Bandw
On 10/5/05, Don Pobanz <[EMAIL PROTECTED]> wrote:
> Stephen Bosch wrote:
> > Since Colin Anderson -- in a previous thread -- asked the question about
> > whether ADSI was dead, I thought it was worth discussing.
> >
> >
> > Does anybody else have anything to add?
> >
> > -Stephen-
>
> I hope ADSI i
On 10/4/05, Mojo with Horan & Company, LLC <[EMAIL PROTECTED]> wrote:
> from the CLI, show g729 -- In my situation, with polycom 501s, if a
> phone calls another internal phone and canreinvite is set to yes, this
> does not count against your licenses 'cause the phones are now the only
> devices in
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