On Wed, 2011-11-30 at 20:03 -0500, Adam Moffett wrote:
You can make a pretty good prediction with ping.
sudo ping -f -i .02 -s 180 -Q 0xb8 [ip] gives a tolerable simulation
of voip traffic. let it run for awhile, then press ctrl+c and see how
many packets were dropped and also check the
On Wed, Nov 30, 2011 at 4:48 PM, Danny Nicholas da...@debsinc.com wrote:
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Yasin SULUHAN
*Sent:* Wednesday, November 30, 2011 8:39 AM
*To:* Asterisk Users Mailing List -
2011/11/30 Marco Mooijekind marco.mooijek...@gmail.com
Maybe use a power supply instead of PoE, see if problem still occurs.
Marco.
Yes, that's what I meant by not using PoE.
Op 30 nov. 2011 18:46 schreef Olivier oza_4...@yahoo.fr het volgende:
2011/11/30 Mike l...@net-wall.com
Hi
Thanks for Carlos for the response,
I have worked with bare asterisk + freepbx before. the mypbx was just
an example but my reference to appliances as a whole.
The appliances seem to have lower entry costs.
On 12/1/11, Carlos Alvarez car...@televolve.com wrote:
At the most basic level,
Hi Noll,
all works perfectly thanks a lot for your help and support i really
appreciate it :)
Best Regards
2011/12/1 Dale Noll dn...@wi.rr.com
On 11/30/2011 11:13 AM, salaheddine elharit wrote:
i have last question regarding this thread
with exten = 3,n,MYSQL(Query resultid ${connid}
Hello list,
I'm trying to populate my CDR logs with values which are available after the
call has started (e.g. signalling IP of remote user, media IP, codec etc.).
While CHANNEL function give me all I need for the incoming leg (leg A), I can't
get the relevant values for the outgoing channel.
2011/11/30 Mike l...@net-wall.com
Hi Olivier,
** **
It if occurs only on the sidecar, I would imagine this is either a
defective sidecar/Polycom phone, or a defective PoE switch not giving
enough power. Changing PoE port would eliminate of confirm the PoE port
being the issue, but I’m
Le 01/12/2011 13:44, Olivier a écrit :
[...]
I still can explain myself why a PoE switch (a Linksys SRW224P) would
succeed or fail to deliver power to a plugged IP phone, given that
only a couple of Polycom phones are using this switch a power source.
I think your switch deliver a max value of
On Tue, 29 Nov 2011, C F wrote:
On Mon, Nov 28, 2011 at 10:57 AM, Tom Browning ttbrown...@gmail.com wrote:
On Sun, Nov 27, 2011 at 8:47 AM, Gordon Henderson
gordon+aster...@drogon.net wrote:
Linux has excellent built-in subsystems to control firewalling and so on
without resorting to external
On Wed, 30 Nov 2011, Tom Browning wrote:
On Tue, Nov 29, 2011 at 4:44 PM, john Millican j...@millican.us wrote:
Maybe I am misunderstanding the gist of the comment
OP offered an invalid comparison of how iptables is better than Fail2Ban.
Whether or not OP knew that Fail2Ban simply feeds
On Wed, 30 Nov 2011, jon pounder wrote:
On 11/30/2011 09:01 AM, Tom Browning wrote:
I agree - its a bad comparison of 2 different things meant for different
purposes.
iptables is enforcement, fail2ban is detection.
iptables can also detect and log these detections.
if you have time to
On Tue, 29 Nov 2011, C F wrote:
BTW, you were just proven wrong, you need it for this hack.
In addition to the few hundred protected asterisk installations I run, I
also run a few honeypots.
Gordon
--
_
-- Bandwidth and
Hi all,
any idea about how to replace Skype For Asterisk?
Thank You.
Giorgio
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
On 12/01/2011 08:30 AM, gincantalupo wrote:
any idea about how to replace Skype For Asterisk?
Replace with what?
--
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/
On Thursday 01 December 2011, gincantalupo wrote:
Hi all,
any idea about how to replace Skype For Asterisk?
Thank You.
Giorgio
1. Migrate your Skype users over to a better product which supports proper
open standards.
2. Write to your elected representatives asking that they order
On Thu, 1 Dec 2011, Jonas Kellens wrote:
Like I said : I can play the sound file with Totem on Linux or
VLC-player on Windows. So it's not that the wav-file has no sound...
Can you post a link to a sample file?
--
Thanks in advance,
Hello,
when I use the Agi, sometimes not play the phrase:
WARNING[30391]: file.c:650 ast_openstream_full: File ggl_U0sBo0 does not exist
in any format
Regards
- Original Message -
From: Lefteris Zafiris
To: asterisk-users@lists.digium.com
Sent: Wednesday, November 30, 2011
On Thu, Dec 1, 2011 at 8:13 AM, Gordon Henderson
gordon+aster...@drogon.net wrote:
Yes, I know exactly how Fail2Ban works.
Then you should be able to proffer a better argument of why it isn't necessary.
--
_
-- Bandwidth and
From experience, that model is not reliable. I have changed those with HP
Procurve and my problems were gone.
Just my 0.02
Jimmy
-Original Message-
From: oza_4...@yahoo.fr
Sent: Thu, 1 Dec 2011 13:44:44 +0100
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Issue
On Thu, Dec 1, 2011 at 8:30 AM, gincantalupo
gincantal...@fgasoftware.com wrote:
Hi all,
any idea about how to replace Skype For Asterisk?
Thank You.
Giorgio
We are going through this right now and have chosen to Pay The Man
via per channel subscription to Skype Connect.
Watch the fun
Hi. Aside from converting spaces to plus signs, you don't encode any
special characters before putting them in the URL. It might be safer to
run $line through some sort of encoding before calling Google with it,
even if most special characters probably don't result in any sound.
Google say and if
On Thu, 1 Dec 2011 09:43:29 -0500
bakko asannu...@gmail.com wrote:
Hello,
when I use the Agi, sometimes not play the phrase:
WARNING[30391]: file.c:650 ast_openstream_full: File ggl_U0sBo0 does
not exist in any format
Regards
Seems like the script failed to convert the mp3 data that
I personally don't like the use of mpg123 for playback - would prefer use of
the internal Playback/background functions. Still seems to be a nice
effort though.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
On Thu, 01 Dec 2011 17:23:59 +
Kingsley Tart kings...@skymarket.co.uk wrote:
Hi. Aside from converting spaces to plus signs, you don't encode any
special characters before putting them in the URL. It might be safer
to run $line through some sort of encoding before calling Google with
it,
On Thu, 1 Dec 2011 11:35:21 -0600
Danny Nicholas da...@debsinc.com wrote:
I personally don't like the use of mpg123 for playback - would prefer
use of the internal Playback/background functions. Still seems to
be a nice effort though.
mpg123 used to convert the mp3 data that we get from
Hello Everyone,
The timezone is set correctly on the OS America/Toronto:
mv /etc/localtime /etc/localtime.bak
cp /usr/share/zoneinfo/America/Toronto /etc/localtime
I even tried adding the timezone setting to sip.conf:
timezone=America/Toronto
However. Asterisk wants to be in Bucharest?
Assuming it's nothing quirky in some mysql or odbc, I would do
- grep Europe /etc/asterisk/*
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis
Sent: Thursday, December 01, 2011 2:36 PM
To: Asterisk
This is because you need to add /tmp to the STREAM command, ie:
print STREAM FILE /tmp/$tmpname \$intkey\\n;
$tmpname seems to not contain the path, so it will look in
/var/lib/asterisk/sounds for the file...
This at least made it work for me... (After fixing some other things to make
it work
I'm so sorry, i'm so sorry, i'm so sorry!
Good thing I did not have a chance yet
to transfer it to mysql realtime. It was
in extensions.conf.
Thanks for Everything,
Nick.
On Thu, Dec 1, 2011 at 3:41 PM, Danny Nicholas da...@debsinc.com wrote:
Assuming it's nothing quirky in some mysql or odbc,
On Thu, 1 Dec 2011 21:51:21 +0100
Torbjörn Abrahamsson torbjorn.abrahams...@gmail.com wrote:
This is because you need to add /tmp to the STREAM command, ie:
print STREAM FILE /tmp/$tmpname \$intkey\\n;
$tmpname seems to not contain the path, so it will look in
/var/lib/asterisk/sounds for
Hello,
I'm trying to setup an Asterisk (version 1.8.8) to do SRTP termination and then
send the call on to other servers, unencrypted. All the basics work fine.
I want the Asterisk to do as little as possible with the RTP packets and no
transcoding. We always make sure to force same codec on
This was run on an Fedora 8 machine, with perl 5.8.8. I also found it odd that
the path was not included...
// T
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lefteris Zafiris
Sent: den 1 december 2011
On Thu, 2011-12-01 at 14:02 +, A J Stiles wrote:
On Thursday 01 December 2011, gincantalupo wrote:
Hi all,
any idea about how to replace Skype For Asterisk?
Thank You.
Giorgio
1. Migrate your Skype users over to a better product which supports proper
open standards.
On Thu, 1 Dec 2011 23:23:56 +0100
Torbjörn Abrahamsson torbjorn.abrahams...@gmail.com wrote:
This was run on an Fedora 8 machine, with perl 5.8.8. I also found it
odd that the path was not included...
// T
It seems this is an issue with older versions of perl or at least with
5.8.8. Since
On Thu, Dec 1, 2011 at 8:15 AM, Gordon Henderson
gordon+aster...@drogon.net wrote:
On Tue, 29 Nov 2011, C F wrote:
BTW, you were just proven wrong, you need it for this hack.
In addition to the few hundred protected asterisk installations I run, I
also run a few honeypots.
Protected? You
Some ideas:
* http://www.clearone.com/voip-conference-phones.html
* http://www.konftel.com/Products/Konftel300IP
*
http://www.polycom.com/products/voice/conferencing_solutions/conference_phones/soundstation/soundstation_duo.html
We have tested all of these in our lab but I would prefer not to be
Hi Alex,
replace with anything which could make Asterisk connect to Skype
network, make and receive calls, etc...the usual stuff.
Giorgio
On 12/01/2011 02:40 PM, Alex Balashov wrote:
On 12/01/2011 08:30 AM, gincantalupo wrote:
any idea about how to replace Skype For Asterisk?
Replace
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