We are running Asterisk 1.8.5.0 with an uptime of 40 weeks. Just today our
streaming music on hold stopped working. I remember when we had first
installed 1.8 we had an issue where the streaming music on hold would not
work because Music On Hold was using the DAHDI timing module. We needed the
Here's what I have and it works for me in 1.8.5:
in sip.cfg
voIpProt
voIpProt.SIP
voIpProt.SIP.alertInfo
voIpProt.SIP.alertInfo.1.class=ringAutoAnswer
voIpProt.SIP.alertInfo.1.value=Ring Answer
voIpProt.SIP.alertInfo.2.class=autoAnswer
voIpProt.SIP.alertInfo.2.value=Auto Answer /
I would like to try the ILBC codec on one of our systems.
The system is currently running Asterisk 1.8.5.0 installed from the
Asterisk provided repositories for Centos 5.
Is there a process for installing the ILBC codec under this
environment, or will I have to un-install the RPMs and build
On Tue, Aug 2, 2011 at 4:39 PM, Lefteris Zafiris zaf@gmail.com wrote:
You can write a short makefile for just codec_ilbc module, build it and
install it on your running asterisk system. You will have to install the
asterisk18-devel package and get the asterisk source code either from
a tar
I'm still using macro with Asterisk 1.8.5.0
On Fri, Jul 15, 2011 at 3:17 PM, Paul Belanger pabelan...@digium.com wrote:
On 11-07-15 02:18 PM, Doug Lytle wrote:
--[ UxBoD ]-- wrote:
I back leveled to 1.8.3 and that works fine. What am I missing as
app_macro has been installed okay?
Macro
I agree that call files are not an appropriate way to solve this.
I would like to move back to using the original Page() application
which had always worked for us with 1.4
My initial testing found that MOH from a streaming source such as
Shoutcast only worked if I disabled the DAHDI timing
We just finished an upgrade of our Asterisk system to an HA
environment on a pair of servers using Linux-HA. As part of the
upgrade, we also moved to Asterisk version 1.8.4.3
Most things are working quite nicely on the new system. However, I’m
having trouble getting a paging feature to work. In
On Mon, Mar 8, 2010 at 7:08 PM, Mike l...@virtutel.ca wrote:
This seems like a basic thing to set up, so I have no doubt many people have
done this. Anyone care to point me in the right direction?
In our config files, we have:
softkey1 type: speeddial
softkey1 label: Voice Mail
softkey1 value:
Or charge for full access! Leave a few teasers, and charge some amount to
see them all. I would pay - even close to attendance price... could only
help you get past break even ;)
I agree, I would be quite willing to pay for full access to all the videos from
the Conference.
Bob
On Fri, 2009-06-19 at 11:45 +0500, ABBAS SHAKEEL wrote:
I am required to do some thing like Dail in modem .
User will have to call a modem just like we do in dail up connection
now we need to handle that request and retrieve some parameters
from that send a HTTp request to a web server
What are the HA options for Switchvox systems?
Is it possible to set up redundant systems with DRBD?
I know on the digium website they talk about Optional cold spare
failover What does this mean? Is this an active spare ready for some
sort of automated failover?
Thanks for you help,
Bob
On Thu, 2009-03-26 at 07:19 -0700, Vieri wrote:
Maybe I could simply do something like:
asterisk -rx show channels | grep -c -i zap
to get the number of zap/dahdi channels in use.
I was actually using a command similar to that up until a few months
ago.
/usr/sbin/asterisk -rx 'show
On Fri, 2009-03-06 at 09:41 -0500, m...@njycamps.org wrote:
Anyone know where I can get an Aastra 480i repaired? The phone works
on speakerphone, but when you lift the receiver offthe hook, the phone
does not engage. There is something wrong with the hook. The
receiver works fine, on another
Mark,
Are you still having trouble with your 8002? I had a lot of trouble with
mine initially, but after playing with it for about 8 hours I figured it
out. Now it works great all around our office. Our NOC technician loves
it!
There is a problem with the sample configs that Polycom publishes. I
On Wed, 2009-02-25 at 15:13 +, Jeff LaCoursiere wrote:
Aha! Mind posting that config?
My sip_allusers.cfg looks like this:
CODECS = g711u, g711a
PROXY1_TYPE = Asterisk
PROXY1_ADDR = 192.168.8.1:5060
#PROXY1_KEYPRESS_2833 = enable
PROXY1_KEYPRESS_INFO = disable
PROXY1_HOLD_IP0 = disable
On Wed, 2009-02-25 at 11:37 -0500, M Hulber wrote:
So I'm thinking, would this work if I had a sip_.conf as well as a
sip_.conf? What the relationship between the LINEs in the
sip_.cfg and the Reg on the phone? What's the relationship
between the AUTH and the LINEn_AUTH? This
I found that after moving to Asterisk 1.6 and the latest SVN of
ASterisk-GUI, the link changed from:
http://localhost:8088/asterisk/static/config/cfgbasic.html
to:
http://localhost:8088/static/config/cfgbasic.html
I don't know if you'll find the same or not...
Bob
On Mon, 2009-02-23 at 22:17
On Fri, 2009-02-13 at 15:24 -0500, Miguel Martinez wrote:
I am new to the Asterisk world, but have decided to use the business
edition, but am looking for a cost effective gui interface to manage
the software.
Does the Asterisk-GUI work with Asterisk Business Edition?
this link:
http://www.telos-systems.com/techtalk/digiphones/digiphones_4.htm
States the following:
Generic PBXs will not do for our broadcast application – they just
don’t have the features necessary. For example, while lines may
certainly be shared to multiple phones, there is no way to switch
I'm looking for some info on the Asterisk Appliance.
I understand it has a gui, but can I still do all the dialplan config
that I'm used of doing by hand outside of the gui? If I really wanted
to, could I even ignore that the device has a gui and do all my config
in the files? I guess I'm just
Has there been any work done on using Asterisk as an MGCP client?
I see the 'Asterisk MGCP channels' page on voip-info hasn't been updated
since 2006, and I was wondering if anyone has been able to accomplish
this yet.
I have a situation where it might be helpful to have an Asterisk system
I tried really quickly the other day to send an image to these phones
from the dialplan like this:
exten = 2821,n,SendImage(/var/lib/asterisk/images/asterisk-intro)
or
exten = 2821,n,SendImage(asterisk-intro)
It didn't work for me.
Should this work? Is anyone else using this with Polycom
On Mon, 2008-11-03 at 11:14 -0500, Lincoln King-Cliby wrote:
Any ideas why the audio quality would be so markedly different when
the only thing that seems to be different is where the call is
originating from (POTS line vs. SIP phone)?
Is it possible that calls from your POTS line are going
On Mon, 2008-11-03 at 13:17 -0500, Lincoln King-Cliby wrote:
It's conceivable, but how would I verify this and how would I change
it if that was the problem?
There's a few things you can do here.
1) Check the sip.conf on both sides to see what is defined there for the
trunk. Look for some
On Tue, 2008-10-21 at 13:56 -0500, Bob Pierce wrote:
Does anyone know what the significance is of the b1 being sent here?
Or, is there a way to make Asterisk not send the b1 character as a
test?
As an update to this, I noticed the following lines in libpri.h near
line 236:
/* Network
On Wed, 2008-10-22 at 12:11 -0500, Bob Pierce wrote:
On Tue, 2008-10-21 at 13:56 -0500, Bob Pierce wrote:
Does anyone know what the significance is of the b1 being sent here?
Or, is there a way to make Asterisk not send the b1 character as a
test?
As a further update to this, I've
I'm trying to send CallerID info to a MetaSwitch system over a PRI. The
MetaSwitch gets the info exactly as it is sent by Asterisk, but I think
it might be having trouble with the Hexadecimal b1 that is being sent
just before the first character of the CallerID Name.
Does anyone know what the
On Fri, 2008-09-05 at 09:19 -0700, Amaru Netapshaak wrote:
I have a Sangoma A104d T1 card, a Rhino 24-port FXS box, and am
running
Asterisk 1.4.21.2
I think you're mostly right on this setup, but I wonder if your A104d is
doing some hardware echo cancellation on these calls. If I'm not
On Wed, 2008-08-27 at 11:21 +0300, Atis Lezdins wrote:
If you doubt about some part, you're welcome to ask, i'll try to help
you, but i don't want to provide complete backport to you, as i won't
be able to test it :)
Thanks Atis,
I'll probably try this in a few weeks when I start rebuilding
On Mon, 2008-08-25 at 17:47 -0500, Bob Pierce wrote:
I'm glad to hear that you're enjoying Asterisk 1.4! Unfortunately,
the
shared_lastcall option is only in versions 1.6.0 and up.
Does anybody have a workaround for this in 1.4?
Or maybe a better question:
How stable is 1.6
On Tue, 2008-08-26 at 17:30 +0300, Atis Lezdins wrote:
I'd say - go for backport instead. shared_lastcall is commited in
http://svn.digium.com/view/asterisk/trunk/apps/app_queue.c?r1=86820r2=86985
and it seems that there are no bugfixes for it since. So, backporting
should be fairly simple.
On Tue, 2008-08-26 at 17:53 +0300, Atis Lezdins wrote:
Are there any plans to back port this feature into upcoming 1.4
releases?
No, new features are added only in trunk, and released in next major
release (1.6).
So what would be involved in back porting this feature for our system?
We have just moved up to Asterisk 1.4.21.2 from 1.2.18
We are now dynamically adding and removing members from our queues. Just
like before, our members are shared across multiple queues.
Today was our first day with the system in full production, and the only
minor glitch that I've run across
On Mon, 2008-08-25 at 17:46 -0500, Mark Michelson wrote:
I'm glad to hear that you're enjoying Asterisk 1.4! Unfortunately, the
shared_lastcall option is only in versions 1.6.0 and up.
Does anybody have a workaround for this in 1.4?
___
--
I know you cannot describe the whole scenario in an email, what I need
is a line or some words for each step :) Or if anyone can do the
whole scenario, please send me an email for further discussions.
Would a combination of call files and conference rooms achieve this?
Quick thoughts
Hi all,
I've been googling for a solution here and haven't really come up with
anything yet. We're doing an Asterisk install for a local radio station,
and we're looking for a phone that they can use in their control room
hooked up to their mixer board for recording calls. So, when you phone
in
On Thu, 2008-04-10 at 11:25 -0500, Tilghman Lesher wrote:
If you instead use a separate extension, you can use groups to
restrict the number of people accessing a particular mailbox:
Thanks Tilghman,
I didn't think of that. I'm sure that will work just fine for what we
need.
Have a great
We are using Asterisk 1.2.18 at this site. One of the users brought this
to my attention today.
We have a problem when we take the message off the voice mail. If I am
taking off the messages it used to be [on the old phone system] that no
one else was able to go in take off the message. Now I
On Fri, 2008-02-01 at 15:01 +0100, Alberto Pastore wrote:
Olivier ha scritto:
Hi,
Does such card exist ?
It seems all existing models are designed for PCI buses.
Regards
We're using Aastra 480i phones, and their speakerphone is great.
I even have one in our datacenter and the speakerphone is usable even
with all the noise of the server fans.
I also have a great contact if you happen to be in Canada wanting to buy
some.
Bob
On Mon, 2007-11-26 at 12:30 -0700,
On Sun, 2007-11-18 at 22:14 +0100, Giuseppe Barichello wrote:
I have successfully compiled and installed Asterisk on an Alix board
(AMD Geode 500 Mhz + 256 Mb RAM) on top of Voyage linux (a Debian
variant).
I'm using it at home for a month.
That's very interesting! I've been curious about
Anyone know of a good package for reporting on Queue statistics from
Asterisk?
Bob
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
I've made a change to my manager.conf file in asterisk 1.2.18
Is there a way to reload that config file from the CLI without
restarting asterisk?
Bob
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
On Wed, 2007-10-24 at 13:31 -0700, Richard Lyman wrote:
every time there is a new connection to the asterisk manager
interface, the manager.conf file is reread.
(meaning, it reloads itself)
Great. Thanks for your help!
___
--Bandwidth and Colocation
On Thu, 2007-09-27 at 11:48 -0400, Eric B. wrote:
I'm a complete newbie to Asterisk and have been reading through
documentation and sites for the last couple of hours trying to
understand what to do to start my learning curve with Asterisk, and am
very confused.
The best starting point IMHO
45 matches
Mail list logo
