Hi all,
I'm new to Astrisk so bear with me.
I have just installed AsteriskNOW and am quite familiar with RH
Linux. I have configured it and am using Xlite to connect and learn to
move around the conf files. I have a problem, however. The client
connects and dials ok, but there is no audio. In
.
- Original Message -
From: Chris Nighswonger [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, March 09, 2007 1:16 AM
Subject: [asterisk-users] Newbie Question
Hi all,
I'm new to Astrisk so bear with me.
I have just installed AsteriskNOW and am quite
On 3/8/07, Chris Nighswonger [EMAIL PROTECTED] wrote:
Thanks for the responses.
iptables on the * box has no rules and all tables default to 'accept.'
I have not got to the point of placing calls out across the internet
yet. The issue here is no audio back from the * box when running
through
] Error 2
and dumps me back to the prompt.
I am working with a fresh install of fc6.
Any help is appriciated.
Chris
--
Chris Nighswonger
Network Systems Director
Foundations Bible College Seminary
www.foundations.edu
www.fbcradio.org
[EMAIL PROTECTED]
V:910-892-8761
C:919-820-5473
On 3/16/07, Kevin P. Fleming [EMAIL PROTECTED] wrote:
Chris Nighswonger wrote:
I am working with a fresh install of fc6.
Kernel 2.6.20 was released after Zaptel 1.4.0, so it will not build
against that kernel. Either use an older kernel, use the SVN version of
Zaptel branch-1.4, or wait
Hi all,
I have just successfully configured a Cisco 30VIP to work with my
Asterisk server. I have seven of these phones new and would like to
deploy them. I am wondering if anyone has this phone deployed with
Asterisk and can suggest configuration of the various buttons, etc.
(Bare with me as I
On 3/22/07, Bill Hackensack [EMAIL PROTECTED] wrote:
On 3/21/07, Chris Nighswonger [EMAIL PROTECTED] wrote:
I have just successfully configured a Cisco 30VIP to work with my
Asterisk server. I have seven of these phones new and would like to
deploy them. I am wondering if anyone has
On 3/22/07, Lacy Moore - Aspendora [EMAIL PROTECTED] wrote:
On 3/21/07, Chris Nighswonger [EMAIL PROTECTED] wrote:
Hi all,
I have just successfully configured a Cisco 30VIP to work with my
Asterisk server. I have seven of these phones new and would like to
deploy them. I am wondering
On 3/22/07, Lacy Moore - Aspendora [EMAIL PROTECTED] wrote:
On 3/22/07, Chris Nighswonger [EMAIL PROTECTED] wrote:
1.4.1
I've got one of those at home and a test system running 1.4.2. I'll
take a look tonight and see if there is anything obvious. I'm not a
developer, though. I know one
On 3/22/07, Lacy Moore - Aspendora [EMAIL PROTECTED] wrote:
On 3/22/07, Chris Nighswonger [EMAIL PROTECTED] wrote:
1.4.1
I've got one of those at home and a test system running 1.4.2. I'll
take a look tonight and see if there is anything obvious. I'm not a
developer, though. I know one
On 3/23/07, Chris Nighswonger [EMAIL PROTECTED] wrote:
I have three registering with * and having basic functionality. I am
at a loss to know how to program the buttons (other than dtmf, hold,
mute, spkr). Here is what the * console shows when one of the phones
registers:
-- Starting
On 3/24/07, Timothy Parez [EMAIL PROTECTED] wrote:
Hi,
I have an FWD account and it's configured in asterisk.
I can be called by people using FWD, but I cannot make FWD calls myself.
I have FWD and IAXTel configured as well. FWD has been having problems
with their IAX server for awhile
On 3/24/07, Timothy Parez [EMAIL PROTECTED] wrote:
Mar 24 15:28:55 ERROR[2873]: chan_sip.c:11076 handle_request_subscribe:
Got SUBSCRIBE for extension [EMAIL PROTECTED] from 172.17.249.253,
but there is no hint for that extension
I believe the subscribe error comes from not having a 'hint' in
not.
Thanks
Chris
On 3/23/07, Chris Nighswonger [EMAIL PROTECTED] wrote:
On 3/23/07, Chris Nighswonger [EMAIL PROTECTED] wrote:
I have three registering with * and having basic functionality. I am
at a loss to know how to program the buttons (other than dtmf, hold,
mute, spkr). Here is what
On 3/28/07, Jason Parker [EMAIL PROTECTED] wrote:
- Derek Whitten [EMAIL PROTECTED] wrote:
if i remember right, most of the buttons on those and the 12SP+ phones
don't really work
because there isn't a button template in *
There is a button template, the problem is that most of the
On 3/29/07, Jason Parker [EMAIL PROTECTED] wrote:
- Chris Nighswonger [EMAIL PROTECTED] wrote:
That is the conclusion I came to and was confirmed today in a very
brief chat with one of the individuals listed as a developer on the
chan_skinny module. He said that they could be implemented
Hi all,
Is there a speed dial type application in *? The NEC PBX we
currently use has a feature which allows any phone to access a
system-wide speed dail database simply by keying the speed-dial number
and pressing the 'redial' key from any extension. Even using a vinella
phone on an sli the
On 3/30/07, Jason Parker [EMAIL PROTECTED] wrote:
- Chris Nighswonger [EMAIL PROTECTED] wrote:
On 3/29/07, Jason Parker [EMAIL PROTECTED] wrote:
I really don't know what to say then.. It's a simple switch statement on the
phone model, with some assignments to set what the buttons do
Jason,
Ok, the 30VIP template seems to be working ok as far as button
assignment goes. I can define speeddial numbers to the speeddial
buttons. However, it appears that there is no code to support the
STIMULUS_SPEEDDIAL case. Is this correct?
Debug output from chan_skinny shows the following
On 4/2/07, Jason Parker [EMAIL PROTECTED] wrote:
Yes, you are (mostly) correct. Speeddials can be added to the phone, but they
can't actually be used.. There is code there that is #ifdef'd out, because it
(mostly) does not work.
If this thread should be moved to the -dev list, just let me
I'm looking for any pros/cons of running an Asterisk based PBX over a
metro ethernet pipe. The system will have about 40 handsets and 6
DIDs.
Kind Regards,
Chris
--
_
-- Bandwidth and Colocation Provided by
Are there deb packages available for Asterisk 10 or for 11 beta?
Kind Regards,
Chris
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
Is it possible to indicate multiple incoming calls from a multi-channel DID
on a single phone? The phone in question is a Polycom 550.
I've googled this with little to no success.
Thanks,
Chris
--
_
-- Bandwidth and Colocation
On Mon, Sep 3, 2012 at 8:25 PM, Chris Nighswonger
cnighswon...@foundations.edu wrote:
Is it possible to indicate multiple incoming calls from a multi-channel
DID on a single phone? The phone in question is a Polycom 550.
I think I may have it, but would like some feedback so I won't chase
I have some Polycom 351 on Asterisk 10. On the same box as * I have a tftp
server running to handle configs, etc. The Polycom phones have no problem
grabbing config foo from the tftp server as well as writing log files back
to the server. However, when I use the web-if on a phone to set a custom
What are best practices for allowing connection by remote SIP
extensions over the internet? I'm thinking of putting the SIP inside a
VPN connection.
Kind Regards,
Chris
--
_
-- Bandwidth and Colocation Provided by
On Sat, Sep 29, 2012 at 12:41 PM, Carlos Rojas crt.ro...@gmail.com wrote:
Hello.
Vpn is good idea, is more secure, you can use tls with srtp as well.
Are you using asterisk 1.8? Right?
Asterisk 10.7.0
Kind Regards,
Chris
--
If this is the wrong place to post this I'm sure someone will let me know. :-)
I'm looking for a reliable, inexpensive call termination service
(SIP). The one I am presently with does not seem to know what IPs they
send inbound calls from, and it is maddening to keep up with the FW
changes
On Tue, Oct 2, 2012 at 5:30 PM, Chris Bagnall
aster...@lists.minotaur.cc wrote:
On 2/10/12 6:51 pm, Carlos Alvarez wrote:
Your traffic level, number of concurrent calls, etc would help us know
what
sort of carrier you should be talking to.
Equally important, your geographic location, and
On Wed, Oct 3, 2012 at 4:37 AM, Michel Verbraak mic...@verbraak.org wrote:
Have a look at your /etc/asterisk/rtp.conf file. In it you specify the UDP
portrange your asterisk will use for RTP traffic. change the rtpstart and
rtpend to your needs and set them open in your FW. Do not make the
On Wed, Oct 3, 2012 at 10:45 AM, Carlos Alvarez car...@televolve.com wrote:
On Wed, Oct 3, 2012 at 7:35 AM, Chris Nighswonger
cnighswon...@foundations.edu wrote:
At this point I only have ~40 extensions, so I took Michel's advise
and set my RTP range to 1-10100. The default 1 ports
On Wed, Oct 3, 2012 at 12:09 PM, Carlos Alvarez car...@televolve.com wrote:
And people, please stop trying to use human security to IP port analogies,
they make you look foolish.
--
Carlos Alvarez
TelEvolve
602-889-3003
I stand corrected, Carlos. And thank-you for taking time to tell me
I'm working on setting up incoming fax reception on our * server. The
majority of faxes come through fine. However each timed a fax comes
in, I get a bunch of this:
WARNING[6616]: res_fax_spandsp.c:416 spandsp_log: WARNING T.30 ECM
carrier not found
Should this be of concern to me? A snip of the
I'm running Asterisk 10.7.0 with three sip trunks to my call
termination provider. For the most part everything works great.
However, at apparently random times and usually about 20 mins or so
into the call, the outbound audio stream dies. The call stays
connected and the inbound audio works fine.
I'm running Asterisk 10.7.0 with three sip trunks to my call termination
provider. For the most part everything works great.
However, at apparently random times and usually about 20 mins or so into
the call, the outbound audio stream dies.
The call stays connected and the inbound audio works
On Mon, Nov 5, 2012 at 10:43 PM, Vladimir Mikhelson v...@mikhelson.comwrote:
My practical experience shows otherwise. I am able to receive faxes on
SIP lines pretty reliably with no T.38 support. The biggest issue for
me is CED tones detection. If CED is detected then fax reception goes
On Wed, Oct 31, 2012 at 10:31 AM, Chris Nighswonger
cnighswon...@foundations.edu wrote:
I'm running Asterisk 10.7.0 with three sip trunks to my call termination
provider. For the most part everything works great.
However, at apparently random times and usually about 20 mins
At present I have two hardware identically freepbx/asterisk boxes. The
mysql db on one is slaved to the other and all config files are
rsync'd once every 24 hours (we have few configuration changes).
We use Polycom 321/331/550/650 phones, and I notice that these phones
can be configured with two
During the course of a conversation with an member of the IT group who
handles the E911 center for our county, I learned that all of the county's
E911 is voip based. This got me to wondering why we could not just
configure up a SIP or some such trunk directly to the E911 center to handle
our
not follow SIP RFC
http://en.wikipedia.org/wiki/Next_Generation_9-1-1. That is not saying
your county is not using standard SIP for E911, it just wouldn't be
considered NG911.
--
*From:* Chris Nighswonger
*Sent:* Fri 4/19/2013 11:41 AM
*To:* Asterisk Users Mailing List
On Fri, Apr 19, 2013 at 8:59 PM, Nathan Anderson nath...@fsr.com wrote:
On Friday, April 19, 2013 5:35 PM, Warren Selby wrote:
There are E911 providers that offer this functionality. I know off the
top of my head, 911Enable offers a service like this. A former client of
mine that
On Mon, Aug 19, 2013 at 2:40 PM, Patrick Lists
asterisk-l...@puzzled.xs4all.nl wrote:
On 08/19/2013 08:10 PM, Eric Wieling wrote:
One of Asterisk's dirty little secrets is that it does not show the
source IP when a device or hacker tries sending a call without registering.
The rejection
Has anyone done any integration of USB, etc. panic buttons and Asterisk?
The basic idea would be to have a USB based panic button[1] along with a
bit of code which would trigger a group SMS or perhaps a pre-recorded call
to a group.
Kind regards,
Chris
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