Hi,
As per you suggestion I write small php scripts but didn't get result. Below
is the php script and output of programs too.
*PHP Script:-*
?php
$priline = system('/usr/sbin/asterisk -rnx pri show spans',$pri);
$asterisk = system(/etc/init.d/asterisk status, $asterisks);
$mysql =
The error is pretty straight forward. It is telling you that no Asterisk
service is running in that machine
On 28 April 2011 07:19, virendra bhati virbh...@gmail.com wrote:
Hi,
As per you suggestion I write small php scripts but didn't get result.
Below is the php script and output of
linux-dahdi/README has a section on how to compile and install oslec
On 27 April 2011 22:15, Anthony Messina amess...@messinet.com wrote:
On 04/27/2011 02:06 PM, satish patel wrote:
Which echo cancellation is good between OSLEC and MG2. Dahdi by default
use MG2 echo cancellation on channel.
Hi,
Asterisk server is running on this machine then I tested and I got this
message after run the script.
On Thu, Apr 28, 2011 at 1:20 PM, Tiago Geada tiago.ge...@gmail.com wrote:
The error is pretty straight forward. It is telling you that no Asterisk
service is running in that machine
On
On 28/04/11 8:00 PM, virendra bhati wrote:
Hi,
Asterisk server is running on this machine then I tested and I got this
message after run the script.
What user are you running the script as?
It looks like you're running it as a web server when Asterisk is running
as root?
Try running the
Hi,
I am running this script from wabsite and want to make it just like FreePBX
show status information of Asterisk, mysql etc.
*Asterisk is running as root* at my end.
When I start script from command prompt then I am getting error message..
[root@cent68 mtnl]# php temp.php
PHP Parse error:
This may be Gas on the fire, but I think somebody (Digium/the
community/etc) needs to make a 1.4 parallel installation of 1.8 and get
the baseline in order. Once the parallel features are functional, then we
can all sweat the problems in the extra features. If I can install 1.8 and
know that I
I will throw in my 2 cents on this. I agree that 1.8 is not as stable as it
needs to be. From my perspective I will continue to use the 1.4.x or
1.6.2.x release that is the best fit for me and it should continue to do
what it does and it get's it's security releases.
If the primary development
Thanks for rely,
Actually i have build OSLEC with the help of
http://www.rowetel.com/blog/?page_id=454
And now i am getting error at loading module
root@shirley:~# lsmod | grep echo
dahdi_echocan_mg2 5662 23
dahdi 210313 50 dahdi_echocan_mg2,wanpipe
echo
AH! i dig into kernel and i found there was already echo.ko module exist in
kernel
ls -l /lib/modules/2.6.32-30-preempt/kernel/drivers/staging/echo/echo.ko
-rw-r--r-- 1 root root 10840 2011-03-01 21:48
/lib/modules/2.6.32-30-preempt/kernel/drivers/staging/echo/echo.ko
I remove that echo.ko
- Original Message -
PS. Please don't start a discussion about 1.8 quality in this thread,
that's a separate issue. I just want to know what you think about
closing 1.4 support now. If you want to discuss 1.8 quality, start a
new thread. Thanks.
I don't think it's a separate issue at
On Apr 28, 2011, at 9:53 AM, Russell Bryant wrote:
I don't think it's a separate issue at all. I would like to see discussion
of exactly which issues are preventing users from using Asterisk 1.8. We're
trying to shift focus to those issues and get them resolved as quickly and as
I was right i grab kernel 2.6.35 and build oslec against it and everything
works!!
Thanks all of you...
-S
From: satish...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Thu, 28 Apr 2011 14:42:00 +
Subject: Re: [asterisk-users] Echocancellation OSLEC vs MG2 ?
AH! i dig
- Original Message -
On Apr 28, 2011, at 9:53 AM, Russell Bryant wrote:
I don't think it's a separate issue at all. I would like to see
discussion of exactly which issues are preventing users from using
Asterisk 1.8. We're trying to shift focus to those issues and get
them
Hi everyone,
How can I introduce some distortion, echo, chopping sound and all other bad
quality things that can happen to a SIP trunk? I have plenty of bandwidth
and crisp clear lines so the only thing that I can think of is to limit
bandwidth but even that requires quite some scripting work.
On Apr 28, 2011, at 10:21 AM, Russell Bryant wrote:
For us the biggest issue is multi-tenant parking not working. We've
really given up testing anything beyond that point because without
that feature there really isn't any way we could use it.
Broken as compared to 1.6.2? I ask since that
Hey,
Not sure why you would want to do this. I find nothing destroys a clean
link like running torrents. Try downloading the 10 most popular torrents
off of thepiratebay.org ( that are more than 4 gigs ).
( just make sure aren't breaking any copyright rules ).
That'll saturate your link and
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
Sent: Thursday, April 28, 2011 10:32 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] How to create distortion, echo, and chopping
sound in a SIP trunk?
On Thursday 28 April 2011, Bruce B wrote:
How can I introduce some distortion, echo, chopping sound and all other bad
quality things that can happen to a SIP trunk? I have plenty of bandwidth
and crisp clear lines so the only thing that I can think of is to limit
bandwidth but even that
- Original Message -
On Apr 28, 2011, at 10:21 AM, Russell Bryant wrote:
For us the biggest issue is multi-tenant parking not working. We've
really given up testing anything beyond that point because without
that feature there really isn't any way we could use it.
Broken as
Un-top-posting...
On 28 April 2011 07:19, virendra bhati virbh...@gmail.com wrote:
As per you suggestion I write small php scripts but didn't get result.
Below is the php script and output of programs too.
$priline = system('/usr/sbin/asterisk -rnx pri show spans',$pri);
Unable to
It also depends on what you want. Torrent will saturate your link, wget
can do that too, but Torrents always manage to bypass most QoS rules I
have found on the net, wget doesn't.
So I find torrents always give me a better result for testing my link.
On 2011-04-28 11:36, Danny Nicholas
In article banlktim8w+vjjj87oyy1mvppsfwflut...@mail.gmail.com,
Bruce B bruceb...@gmail.com wrote:
How can I introduce some distortion, echo, chopping sound and all other bad
quality things that can happen to a SIP trunk? I have plenty of bandwidth
and crisp clear lines so the only thing that
You can use tc (traffic control) on linux and limit your bandwidth
http://www.linuxtoday.com/infrastructure/2008092400820OSDBNT
To: asterisk-users@lists.digium.com
From: t...@softins.co.uk
Date: Thu, 28 Apr 2011 15:57:26 +
Subject: Re: [asterisk-users] How to create distortion, echo,
Le 28/04/2011 16:53, Russell Bryant a écrit :
- Original Message -
PS. Please don't start a discussion about 1.8 quality in this thread,
that's a separate issue. I just want to know what you think about
closing 1.4 support now. If you want to discuss 1.8 quality, start a
new thread.
Hi list,
yesterday I converted my voicemail.conf to realtime voicemail and also
configured to store the voicemessages in a database using odbc as described
here http://www.voip-info.org/wiki/view/Asterisk+RealTime+Voicemail and
here
Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?)
priline=Unable to connect to remote asterisk (does /var/run/asterisk.ctl
exist?)
I think, you should start asterisk before executing asterisk commands
regards
Juan.
Linux User #441131
On Thu, Apr 28, 2011 at 1:19 AM,
At 05:56 AM 4/28/2011, you wrote:
If I can install 1.8 and
know that I can turn off things to get to 1.4 solidness, then I don't
have a problem with this kettle of fish. BTW, where does 1.10 fit into this
conversation?
Personally, 1.8 has never lasted more than 12 hours on my box without
Where did you download asterisk 1.10 or trunk ? I search and found nothing.
could your point me there?
-S
Date: Thu, 28 Apr 2011 10:06:18 -0700
To: asterisk-users@lists.digium.com
From: i...@extrasensory.com
Subject: Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?
At
On Thu, Apr 28, 2011 at 1:34 PM, satish patel satish...@hotmail.com wrote:
Where did you download asterisk 1.10 or trunk ? I search and found nothing.
could your point me there?
-S
svn co http://svn.asterisk.org/svn/asterisk/trunk /usr/src/asterisk_trunk
--
~~~ Andrew lathama Latham
On 11-04-28 01:06 PM, Ira wrote:
At 05:56 AM 4/28/2011, you wrote:
If I can install 1.8 and
know that I can turn off things to get to 1.4 solidness, then I don't
have a problem with this kettle of fish. BTW, where does 1.10 fit into
this
conversation?
Personally, 1.8 has never lasted more
Hey Paul,
We have migrate asterisk from 1.2 to 1.8 in production and we have this issue i
wouldn't say its critical but just thought point you out. This is open since
last long time and no one respond :(
https://issues.asterisk.org/view.php?id=18514
Now i am trying 1.10 and let see whether
Thanks for the input guys. What Tony and Satish suggested are alone the
lines of what I need. It gives me a controlled solution. So, I can change
the level of distortion as I please. Using tc I pretty much killed the line
to the point I wasn't able to receive call and terminal was really slow as
On Thu, Apr 28, 2011 at 11:09 AM, Rizwan Hisham rizwanhas...@gmail.comwrote:
Hi list,
yesterday I converted my voicemail.conf to realtime voicemail and also
configured to store the voicemessages in a database using odbc as described
here
I used succesfully huawei E1550
On 24 April 2011 16:45, Dovid Bender asteriskus...@dovid.net wrote:
Hi List,
I am looking to play around with chan_datacard. Any advice on the best
device to test with (that I can find on eBay) ?
Regards,
Dovid
--
On 11-04-28 12:04 PM, Administrator TOOTAI wrote:
Ok, so why not stay with asterisk 1.4 security *and* bug/regression fixes for
few weeks/monthes till 1.8 reaches the level that the community accept to
switch
to 1.8
What is the guide here? What is the level that the community accepts?
On 4/28/11 5:25 PM, Bruce B wrote:
Is there any easy way to simulate a distorted SIP line temporarily for
testing?
Build a Linux based router and use netem/tc to mess around with the
routed traffic. You can insert packetloss, jitter, etc and have it be
reproducable.
--
Andreas Sikkema
--
Hi List,
I have a client that wants me to replace their existing H323
gateway. I am able to get ooh323 and h323 to work fine in a native
environment, but the whole thing goes to heck when I have to cross networks.
Gnugk seems to be the answer to this, but I can't seem to get it to work
Le 28/04/2011 21:47, Leif Madsen a écrit :
On 11-04-28 12:04 PM, Administrator TOOTAI wrote:
Ok, so why not stay with asterisk 1.4 security *and* bug/regression fixes for
few weeks/monthes till 1.8 reaches the level that the community accept to switch
to 1.8
What is the guide here? What is the
At 10:43 AM 4/28/2011, you wrote:
On 11-04-28 01:06 PM, Ira wrote:
At 05:56 AM 4/28/2011, you wrote:
If I can install 1.8 and
know that I can turn off things to get to 1.4 solidness, then I don't
have a problem with this kettle of fish. BTW, where does 1.10 fit into
this
conversation?
On 4/28/11 10:30 PM, Danny Nicholas wrote:
Hi List,
I have a client that wants me to replace their existing H323
gateway. I am able to get ooh323 and h323 to work fine in a native
environment, but the whole thing goes to heck when I have to cross networks.
Gnugk seems to be the
On 11-04-28 04:33 PM, Administrator TOOTAI wrote:
Le 28/04/2011 21:47, Leif Madsen a écrit :
On 11-04-28 12:04 PM, Administrator TOOTAI wrote:
Ok, so why not stay with asterisk 1.4 security *and* bug/regression fixes
for
few weeks/monthes till 1.8 reaches the level that the community accept
On 28/04/11 10:53 PM, virendra bhati wrote:
Hi,
I am running this script from wabsite and want to make it just like
FreePBX show status information of Asterisk, mysql etc.
*Asterisk is running as root* at my end.
When I start script from command prompt then I am getting error message..
Le 28/04/2011 22:43, Leif Madsen a écrit :
On 11-04-28 04:33 PM, Administrator TOOTAI wrote:
Le 28/04/2011 21:47, Leif Madsen a écrit :
On 11-04-28 12:04 PM, Administrator TOOTAI wrote:
Ok, so why not stay with asterisk 1.4 security *and* bug/regression fixes for
few weeks/monthes till 1.8
On 11-04-28 04:35 PM, Ira wrote:
If you want to look at this with my help, an email off-list will get
your use of me and my Asterisk box.
I just posted a patch on the issue tracker, I'll need to get it reviewed
to see if this is the best approach.
--
Paul Belanger
Digium, Inc. | Software
At 03:22 PM 4/28/2011, you wrote:
On 11-04-28 04:35 PM, Ira wrote:
If you want to look at this with my help, an email off-list will get
your use of me and my Asterisk box.
I just posted a patch on the issue tracker, I'll need to get it
reviewed to see if this is the best approach.
I would
- Original Message -
Sure. Please follow the 2 next stories:
- had a customer running 1.4.26 We upgraded to a new server and
installed 1.4.39, last version at this time. Bang: voicemail doesn't
work as it should, had to fallback to 1.4.26 Customer is still running
this version.
-
- Original Message -
Thanks Matt.
There seems to be an unresolved deadlock since the birth of 1.8.
Using the most basic feature of a PBX, try to pickup some elses
ringing
extension - DEADLOCK.
But I'm on to it, https://issues.asterisk.org/view.php?id=18654 and
it's
more
On 29/04/11 10:10 AM, Alec Davis wrote:
Thanks Matt.
There seems to be an unresolved deadlock since the birth of 1.8.
Using the most basic feature of a PBX, try to pickup some elses ringing
extension - DEADLOCK.
But I'm on to it, https://issues.asterisk.org/view.php?id=18654 and it's
more
- Original Message -
I would comment that I've been complaining about this since RC1 or 2
and if you just fixed it in 2 hours that there is something seriously
wrong with the bug tracking system. I mean, I reported it a long
time ago and while it was probably not the best bug report
At 03:48 PM 4/28/2011, you wrote:
- Original Message -
I would comment that I've been complaining about this since RC1 or 2
and if you just fixed it in 2 hours that there is something seriously
wrong with the bug tracking system. I mean, I reported it a long
time ago and while it
On 11-04-28 06:39 PM, Ira wrote:
I would comment that I've been complaining about this since RC1 or 2 and
if you just fixed it in 2 hours that there is something seriously wrong
with the bug tracking system. I mean, I reported it a long time ago and
while it was probably not the best bug report
On Wed, 27 Apr 2011 14:15:10 +0300, Ashik Ali
beaasteriskg...@gmail.com wrote:
Anybody can explain me why asterisk is unable to detect ringback tone
from PSTN telco ? .
I guess it was a lot of work, and nobody bothered adding this to the
Zaptel driver.
--
Making an assumption here, I'm sure I cleared the remaining resequencing
issues up in 1.4 SVN and 1.6.2 SVN.
https://issues.asterisk.org/view.php?id=19032
The issues I uncovered and fixed were when a new voicemail is left, while a
mailbox is open for review and the user deletes a message.
Alec
Hi Vip,
On 28/04/11 05:34, vip killa wrote:
I just completed building a feature rich asterisk voicemail system
using perl, php, and mysql.
My only concern is that the system i built will not be able to handle
the call volume needed. Let me start by explaining my setup.
Incoming call -
Hi,
I'm about to deliver a production system based on Debian Squeeze and
Asterisk 1.6.2.9-2+squeeze1 from the Debian repositories. Asterisk 1.8
packages for Debian Ubuntu are available from packages.asterisk.org.
Observing some recent discussions on this list, it seems that 1.8 might
not yet be
On 29/04/11 11:19 AM, Jan Bakuwel wrote:
Hi,
I'm about to deliver a production system based on Debian Squeeze and
Asterisk 1.6.2.9-2+squeeze1 from the Debian repositories. Asterisk 1.8
packages for Debian Ubuntu are available from packages.asterisk.org.
Observing some recent discussions on
On 11-04-28 07:02 PM, Ira wrote:
At 03:48 PM 4/28/2011, you wrote:
OK, maybe not, but if I thought it was a bug and you discover it was a bug and
fix it, than who was it who decided it wasn't a bug 15 minutes after I put it
in
the bug tracker and why did that person have that much power?
On 29/04/11 5:06 AM, Ira wrote:
At 05:56 AM 4/28/2011, you wrote:
If I can install 1.8 and
know that I can turn off things to get to 1.4 solidness, then I
don't
have a problem with this kettle of fish. BTW, where does 1.10 fit into
this
conversation?
Personally, 1.8 has never lasted more
Hi Matt,
On 29/04/11 11:26, Matt Riddell wrote:
On 29/04/11 11:19 AM, Jan Bakuwel wrote:
Hi,
I'm about to deliver a production system based on Debian Squeeze and
Asterisk 1.6.2.9-2+squeeze1 from the Debian repositories. Asterisk 1.8
packages for Debian Ubuntu are available from
On 11-04-28 07:09 PM, Alec Davis wrote:
Making an assumption here, I'm sure I cleared the remaining resequencing
issues up in 1.4 SVN and 1.6.2 SVN.
https://issues.asterisk.org/view.php?id=19032
The issues I uncovered and fixed were when a new voicemail is left, while a
mailbox is open for
On 29/04/11 11:51 AM, Ernie Dunbar wrote:
On 29/04/11 5:06 AM, Ira wrote:
At 05:56 AM 4/28/2011, you wrote:
If I can install 1.8 and
know that I can turn off things to get to 1.4 solidness, then I
don't
have a problem with this kettle of fish. BTW, where does 1.10 fit into
this
conversation?
At 04:49 PM 4/28/2011, you wrote:
Well the issue is that we currently have over 900 open issues in the Asterisk
project alone, and with only one primary bug marshal (myself) sometimes things
accidentally get closed if it looks like a configuration issue.
If anyone ever opens an issue they they
Let me try to better describe the test senario that I found, and have been
commited to 1.4svn, 1.6.2svn 1.8svn and trunk.
All aspects need to be thrased out though.
Leave Phone-A 2 new messages, and for this example we only have 2 new
messages.
Now to create the problem - (gaps in the message
On 29/04/11 1:16 PM, Ira wrote:
Well, I've no idea how to do that. I can duplicate the problem every
IRC is an online chat system like MSN or Skype except that it's more
like a mailing list - you can talk to lots of people at the same time.
On Windows you can use a program like mIRC to
Hi all,
Friday at 12 Noon EDT, we'll be talking to Emil Ivov of Jitsi.org
(formerly SIP Communicator) and Thiago Rocha Camargo (of Nimbuzz)
about Jabber, something the Asterisk community is becoming more
interested in by the day. Join us to learn more about Jabber and SIP
or to share your
On 29/04/11 2:15 PM, Alec Davis wrote:
Let me try to better describe the test senario that I found, and have been
commited to 1.4svn, 1.6.2svn 1.8svn and trunk.
All aspects need to be thrased out though.
Leave Phone-A 2 new messages, and for this example we only have 2 new
messages.
Now to
On 29/04/11 12:01 PM, Jan Bakuwel wrote:
Rather than testing and finding issues that have already been resolved,
I'd prefer to have an efficient way to upgrade Asterisk to released
versions. A package system provides an efficient way to do this. The
fact that something like packages.asterisk.org
Hi Friends,
I got hostname through dialplan ENV() function and set environment variable
hostname in asterisk init script.
Thanks for everyone to resolve my problem.
On Fri, Apr 22, 2011 at 2:27 AM, Mark Deneen mden...@gmail.com wrote:
On Thu, Apr 21, 2011 at 4:30 PM, Steve Edwards
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