Hi all,
I am using the Asterisk Manager API to originate calls and it is working well,
when a call is placed the local phone rings, once you pick it up you can here
the call ringing the other end. Now, I am using Polycom IP 300 and I have
them setup to auto-answer if I set the ALERT_INFO
Well I tried just about every combination that I can think of as well as every
combination mentioned and it still doesn't work. Not sure why, maybe it's
just not possible from the Manager API.
Pete
On Monday 15 November 2004 04:56, Peter Svensson wrote:
On Mon, 15 Nov 2004, Brian West wrote:
I had the same concenrs when my TDM400 showed up but the folks at Digium told
me to just plug the RJ11's into the RJ45 jacks, which I have done, and they
work fine. Our system went live 2 weeks ago.
Pete
On Tuesday 16 November 2004 05:17, you wrote:
OK I'm new to this. Just got the developers
Hi All,
I'm curious to know what software based SIP phones people are using under
Linux that work with Asterisk. I have tried several including kphone,
linphone, and SJPhone, I have the same problem with all of them, my voice
comes out quiet on the other end, and there is quite a bit of
On Wednesday 17 November 2004 12:15, Diego Aguirre wrote:
Hi,
I am using X-Lite with Wine!
wow! I triied to get it working under wine but it was a no go. I'm very
familiar with Wine, we run a few apps under wine here. Coudl you share your
config or somet tips to help me get it running?
Are
Asterisk.
Pete
On Wednesday 17 November 2004 12:23, Tom Ivar Helbekkmo wrote:
Peter Osborne [EMAIL PROTECTED] writes:
I would blame my onboard sound (I'm using a Toshiba M30 laptop)
except that I have had no problems using Skype on this machine.
I hear the Skype folks have done a very good job
We had the same with our old phone system, I replaced it be adding a little
panel in our web based support system that shows extension status, call
duration, etc. I used an iframe that refreshes every 5 seconds, I wrote a
script in python that generates the data by watching the Asterisk
I had the same problem on Debian, the mpg123 in Debian is really mpg321 which
is supposed to be a drop in replacement. Well, I don't think it is, I
compiled mpg123-0.59r from source and it works now. You may want to give that
a try.
Pete
On Thursday 18 November 2004 04:32, Joost Kraaijeveld
I've had this problem with the IP300's, we just reboot the phone.
Pete
On Monday 22 November 2004 04:31, Tim Jackson wrote:
We have Polycom IP500's, and just starting recently (after the
broadvoice patch I might add) after about 1-2 days these phones ring,
and answer, but we get no audio on
Hello,
With all the talk about Firefly, I decided to check it out, it seems to work
under wine (IAX only for some reason) so I'm thinking about using it on the
road. Now, my Asterisk box is behind a firewall, so I have set the firewall
to forward UDP port 4569 to my Asterisk box put I'm having
Yes,
I set constants for all of my extensions and extension groupings so I can use
logical names to ring a phone or a group of phones.
Pete
On Thursday 02 December 2004 12:40, Ed Greenberg wrote:
Can you put the SIP/3001SIP/3002SIP/3003...on and on in some
sort of variable, macro or
Hello all,
We've been using our Asterisk system live for about a month now and I'm
looking to tuning a few things. First, is echo, I receive a fair amount of
echo during the first 10-15 seconds of incoming calls.
Next is a very weird problem. We have serveral Polycom IP300's and one
Budgetone
On Tuesday 07 December 2004 12:34, Steven Critchfield wrote:
On Tue, 2004-12-07 at 11:00 -0500, Peter Osborne wrote:
Hello all,
We've been using our Asterisk system live for about a month now and I'm
looking to tuning a few things. First, is echo, I receive a fair amount
of echo during
Hi All,
I would like to add a panel to our in-house web based CRM software that
displays the status of people's extensions similar to the way the Asterisk
Flash Control Panel does but I would like to use either straight html or
dynamically generated GIF images.
First off, I assume I can
Hi All,
I'm using Asterisk for my PBX, I have a remote office that is connected by a
VPN link. I am using Openswan on my side and a Linksys box on the remote
side. I have a Polycom IP300 on the remote side configured with a static IP
address. When I call the phone on the remote side, it rings
Schindler [EMAIL PROTECTED]
To: Peter Osborne [EMAIL PROTECTED]
Cc: asterisk-users@lists.digium.com
Sent: Friday, July 15, 2005 8:35 PM
Subject: Re: [Asterisk-Users] VPN's
On Fri, 15 Jul 2005, Peter Osborne wrote:
Hi All,
I'm using Asterisk for my PBX, I have a remote office
You can alway use google's cache. Use site:www.voip-info.org when searching
or type the full URL into google and click on the cached version.
Pete
On 15 July 2005 4:36 pm, Damon Estep wrote:
Does anyone have a mirror of this running?
___
in the localnet directive
(otherwise, they'll use the wan ip address, and that may be the
problem...)
Julian.
On 7/15/05, Peter Osborne [EMAIL PROTECTED] wrote:
Hi All,
I'm using Asterisk for my PBX, I have a remote office that is connected
by a VPN link. I am using Openswan on my
Hi all,
I just upgraded from Asterisk 1.0RC1 to Asterisk 1.0.7 and our dtmf no longer
works with external phone systems. I have a Wildcard TDM400P with 4 FXO's?
(it connects to analog lines). No changes were made to the config files.
Here's my config:
/etc/zaptel.conf
fxsks=1-4
loadzone = us
Hi All,
I have some Polycom IP300's and I'm interested in increasing the max volume
for the headset (not handset), I'm wondering if anyone has experience
adjusting these values:
gains
voice.gain.rx.analog.handset=0 voice.gain.rx.analog.headset=0
voice.gain.rx.analog.chassis=3
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