Yes - the USB connection carries the data. Keep in mind that the HA aspect
of this product just means you can connect to two asterisk servers. There is
not data replication, detection of asterisk failure, etc. (without buying more
xorcom products). Be sure to do your homework. But they do
...@cfmc.com
CfMC
http://www.cfmc.com/
On Apr 30, 2011, at 8:31 PM, Kaushal Shriyan wrote:
On Sun, May 1, 2011 at 8:48 AM, Michelle Dupuis
mdup...@ocg.camailto:mdup...@ocg.ca wrote:
Yes that's it - one PRI line in, 2 out (one to the PRI card in each server).
If you have lots of PRI lines, you may
Use simple RJ45 (8 wire) A-B switched controllable by serial port, and use
HAAST to throw the A-B switch to reroute the PRI.
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Adolphe Cher-aime
or point me
to any document of website.
--
Sent from my iPhone
On Apr 30, 2011, at 12:09 PM, Michelle Dupuis mdup...@ocg.ca wrote:
Use simple RJ45 (8 wire) A-B switched controllable by serial port,
and use HAAST to throw the A-B switch to reroute the PRI
...@lists.digium.com] On Behalf Of Kaushal Shriyan
[kaushalshri...@gmail.com]
Sent: Saturday, April 30, 2011 11:03 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] HA Asterisk
On Sun, May 1, 2011 at 2:13 AM, Michelle Dupuis
mdup...@ocg.camailto:mdup...@ocg.ca wrote:
There are lots out there, but here's
For the High Availability part check out the HAAST add-on for Asterisk at
www.generationd.com
It detects a variety of failures, shuts down the failing system, starts
asterisk on the peer, moves the IP over, etc. Runs with every Asterisk variant
and every Linux distro. No special hardware
I have a situation where an Asterisk server is NATted, sitting behind a PIX.
One public IP is used for one purpose, now a second public IP is required for
another.
Is there a way to have Asterisk use more than one public IP when behind NAT?
(I already use the externalIP setting)...
If not,
: Re: [asterisk-users] Multiple public address to one Asterisk
serverbehind NAT?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis
Sent: Tuesday, February 22, 2011 3:34 PM
To: Asterisk Users
I found some great pieces of script on the internet that I've combined to allow
Asterisk to send voicemails as an MP3 file, and encode the sender name and
number as well as message number as tags into the MP3 file. I even include a
cover art image which has our company logo and PBX symbol in
Ok - I've put the script up on the www.generationd.com web site. Just go to
the Downloads | Asterisk section to pull it down.
I would like to keep control of this script so please send me changes (don't
repost elsewhere) and I'll keep the latest version up for everyone. I'll add a
link to
Take a look at High Availability ASTerisk (HAAST) from www.generationd.com
Their software sits between the OS and asterisk, and can failover servers,
switch IP addresses, control external interfaces, etc.
It can run on different hardware (make a cluster from different/cheap boxes),
it allows
, 2011 2:44 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Dialplan to bridge 2 legs?
On Sun, 23 Jan 2011, Michelle Dupuis wrote:
Is it possible to have a call file enter the dialplan, and then initiate
2 outbound calls and then bridge them?
A call file can specify a channel
to bridge 2 legs?
Un-top-posting...
On Sun, 23 Jan 2011, Michelle Dupuis wrote:
Is it possible to have a call file enter the dialplan, and then
initiate 2 outbound calls and then bridge them?
On Sun, 23 Jan 2011, Steve Edwards wrote:
A call file can specify a channel and a context/exten/priority
We have an application that plays a variety of sound files on one leg of a call
(generated by a call file). We've been told that the party listening to the
audio files intermittantly hears robotic sounding audio (on/off during the
same call).
Anyone have ideas on cause? These calls are on an
I've searched through the wiki but I can't find what I need...I'm trying to
figure out what the max call duation is. I found references to show
application AbsoluteTimeout but that isn't in 1.6 (not even prepending core
to the front). A core help show didn't help...
--
Is the a CLI command that shows all channels in use at one time? (Whether IAX,
SIP, SCCP, etc)?
As well, when I SIP SHOW CHANNELS I see phones registering showing as
channels in use. Is there a way to filter this output?
Thanks!
MD
--
We have a small office installation running over a cable modem. (8M down, 500k
up confirmed with numerous speed test sites)
When a single call is up, call quality is fine. When a second call is up,
outbound audio is immediately choppy. We're using ulaw, and confirmed that
traffic with 2
Jitterbuffer affects inbound audio only, not outbound (the other side hears the
choppiness) so I don't think that will help/
Trunking only reduces overhead after 4+ calls, so that shouldn't help either.
(Since this occurs at 2 calls)
I can't wireshark the other end since the other end is my
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Benny Amorsen
[benny+use...@amorsen.dk]
Sent: Monday, September 27, 2010 10:35 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] Asterisk Redundancy
Michelle Dupuis
: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Vahan Yerkanian
[va...@arminco.com]
Sent: Monday, September 27, 2010 1:02 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Asterisk Redundancy
On 9/27/10 8:57 PM, Michelle Dupuis wrote:
HAAST
Check out HAAST (High Availability ASTerisk) at
www.generationd.comhttp://www.generationd.com (also on the voip wiki)
You get the cluster/heartbeat replication without needing to add openSER or
full HAlinux. A simpler approach - easier to config and manage
MD
Your (local phone) dialplan is not getting pushed out to the handset. Increase
the version number in your config to force it out to the handset...
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of
I realize this is getting a bit outside myth...but hopefully someone can offer
some ideas...
I'm using the latest NVIDIA drivers on Fedora 12, with Nvidia 8600GT. Although
the dual DVI outputs work great, the driver just won't detect anything
connected to the component video connector.
Is
Are there any best practices for using a SAN with Asterisk? In the past we've
kept config files local, but voicemail on a SAN. Aree there any issues with
latency putting voice prompts, configs, etc. on a SAN?
Anyone have some best practices to share?
MD
--
I need to grab the voicemail WAV file once the voicemail command is done. Is
there a hook to be notified that voicemail is done, and get the name of the
recorded file?
Thanks
MD
--
_
-- Bandwidth and Colocation Provided by
Is there a prebuild module/dialplan which gives me a nice interface to
recording messages? Assuming I can't use the voicemail command, I need to
offer users a way to record, playback, erase, rerecord, etc.
I can probably do it through dialplan but it feels like I'm reinventing the
wheel.
...@lists.digium.com] On Behalf Of Leif Madsen
[leif.mad...@asteriskdocs.org]
Sent: Tuesday, July 27, 2010 9:49 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Recording interface (pause/PLAY/RERECORD)
On 10-07-27 08:39 PM, Michelle Dupuis wrote:
Is there a prebuild module/dialplan which gives me a nice
-boun...@lists.digium.com] On Behalf Of Sherwood McGowan
[sherwood.mcgo...@gmail.com]
Sent: Tuesday, July 27, 2010 8:47 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Recording interface (pause/PLAY/RERECORD)
There's an app_record, and I believe app_dictate
On 7/27/2010 7:39 PM, Michelle
Of Leif Madsen
[leif.mad...@asteriskdocs.org]
Sent: Tuesday, July 27, 2010 9:22 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Grab voicemail WAV file when done
On 10-07-27 08:38 PM, Michelle Dupuis wrote:
I need to grab the voicemail WAV file once the voicemail command is done
...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Belanger
[paul.belan...@polybeacon.com]
Sent: Tuesday, July 27, 2010 10:10 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Recording interface (pause/PLAY/RERECORD)
On Tue, Jul 27, 2010 at 10:05 PM, Michelle Dupuis
:55:07PM -0400, Michelle Dupuis wrote:
And after cleaning everything 323/pwlib/ptlib related, and reinstalling ptlib
and h323plus, I can't even get asterisk to compile chan_h323 anymore.
Perhaps something old was left over.
My .configure run shows:
checking /usr/src/openh323plus/h323plus
I see that objective systems has updated their ooh323 stack, but it is not
compatible with the latest chan_ooh323 wrapper available on their site.
Has anyone update the chan_ooh323 wrapper for Asterisk 1.6.2.x ?
Michelle
--
_
I'm really struggling with an Asterisk 1.6.2.7 install (on centos 5.4)
The pwlib + opal packages don't satisfy Asterisk's configure script (to let
H323 compile), so I removed those and added the latest ptlib + h323plus (from
h323plus.org)
I can compile ptlib and h323, but when I load chan_h323
+pwlib
from centos packages work? (trick asterisk .configure to accept them)?
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis
[mdup...@ocg.ca]
Sent: Sunday, June 20, 2010 7:45 PM
To: Asterisk
I'm looking for a small formfactor mobo for an install that needs to handle 25
phone sets (no transcoding). I found a new dual atom 1.66GHz mobo - anyone
know what kinds of call volume that will handle?
MD
--
_
-- Bandwidth
I checked out the sites and can't figure out what this thing is! (Without
delving into the documentation).
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Matthew J. Roth
[mr...@imminc.com]
Sent:
I'm succesfully using IAXmodem for faxing (using hylafax) with Asterisk. I
would like a little more control for outbound calls using IAXmodem, but I'm not
sure how to do it. It looks like dialing out over IAXmodem bypasses the
dialplan altogether...can anyone confirm this?
MD
--
This isn't an Asterisk issue, it's a routing issue. Take a look at iproute2
and routing policies.
Another way to view it is that Asterisk hands the communications over to Linux,
where the network route takes over. (The * bind statement just tells * what IP
to listen on)
If you have 3
I have a Centos 5.4 64 bit installation. I've tried installing asterisk
1.6.2.7 from source, and from RPM, and although overall things work, the
chan_ooh323.so module won't load. Every attempt to load causes Capabilities
failure for OOH323. OOH323 Disabled.
I looked at the source and the
The High Availability HASTerisk (HAAST) product on www.generationd.com is a
software solution that does automatic failover, etc between multiple asterisk
machines. I'm guessing this could be part of an overall solution for you
From:
I can't find this in the wiki/email history..but I'm sure it's based asked
before.
The port range define in rtp.conf - is that for connections initiated by
asterisk? Or the port range asterisk listens on? Or both?
Thanks!
MD
--
I would like to play music to an inbound caller, AFTER asterisk answers the
call, but before the call is bridged by DIAL. Is there a simple way to
achieve this?
MD
--
_
-- Bandwidth and Colocation Provided by
Exten = s,n,wait(10,m)
Exten = s,n,Dial.
This would wait 10 seconds playing MOH before dialing.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle
Dupuis
Sent: Monday, March 22, 2010 3:58 PM
To: 'Asterisk Users
,n,Dial(SIP/callwithus/17025551212,120,A(ginr3))
On Mon, Mar 22, 2010 at 6:41 PM, Michelle Dupuis mdup...@ocg.ca wrote:
I think I forgot some important information...
I'm actually running an AGI script after the answer (and before the dial).
I would like to play MOH while the AGI script
I've got Asterisk 1.6 bridging to an Avaya using H323. The Avaya is
autoanswering calls to music (as expected) and audio seems fine, but I see
this error on bridging:
WARNING[8833]: chan_ooh323.c:1054 ooh323_indicate: Don't know how to
indicate condition 20 on ooh323c_o_2
Is this a warning I
We have a client with Asterisk 1.6 installed via yum (onto Centos). It did
not included the chan_h323 driver apparently, so we installed add-ons by
yum. We then got ooh323.
Is it possible to install the H.323 drivers without compiling from source?
--
. ;)
On Wed, 10 Mar 2010 10:34:30 -0500, Michelle Dupuis wrote:
I'm trying to connect an Asterisk 1.6 to an Avaya with
gatekeeper (CLAN). When chan_ooh323 first loads it tries to establish a
connection with the gk but I it fails. I have the following extract from
the ooh323 log
In case someone wants to see the detailed ooh323 log (which shows the failed
attempt to connect to the gatekeeper). I appreciate any help!!
21:32:06:832 Sent GRQ message
21:32:06:885 GkClient Received RAS Message
21:32:06:885 Received RAS Message = {
21:32:06:885 gatekeeperConfirm = {
I'm trying to connect an Asterisk 1.6 to an Avaya with gatekeeper (CLAN).
When chan_ooh323 first loads it tries to establish a connection with the gk
but I it fails. I have the following extract from the ooh323 log. Can
anyone give some insight?
Thanks!
MD
23:02:59:045 Sent GRQ message
We are coordinating a connection to a SIP provider who told us they use two
port ranges for RTP, 7000-8000 and 1-2.
We've never encountered that before (and I believe rtp.conf only supports a
single range). We can obviously setup 7000-2 within RTP.conf, but I'm
wondering if there is
without any
problems. I need your ooh323.conf and all relevant CM config (signal-group,
trounk-group, ip-codec... ) before I can assist u. ;)
On Wed, 10 Mar 2010 10:34:30 -0500, Michelle Dupuis mdup...@ocg.ca wrote:
I'm trying to connect an Asterisk 1.6 to an Avaya with gatekeeper (CLAN).
When
problems. I need your ooh323.conf and all relevant CM config (signal-group,
trounk-group, ip-codec... ) before I can assist u. ;)
On Wed, 10 Mar 2010 10:34:30 -0500, Michelle Dupuis wrote:
I'm trying to connect an Asterisk 1.6 to an Avaya with gatekeeper (CLAN).
When chan_ooh323 first loads it tries
: [asterisk-users] Which H.323 to use in Ast 1.6
Which Avaya system are you running?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle
Dupuis
Sent: Wednesday, February 24, 2010 5:52 PM
To: 'Asterisk Users
Subject: Re: [asterisk-users] Which H.323 to use in Ast 1.6
I have always used ooh323 between Avaya and Asterisk.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle
Dupuis
Sent: Tuesday, February 23, 2010 2:24 PM
To: 'Asterisk Users
We're creating a SIP gateway for a client that will take one leg of a call
in via SIP, and out the other side via H.323. To minimize load on the
gateway, we would like to have the RTP stream bypass the gatewayy altogether
(directrtp/reinvite). Is this possible with these to protocols?
Thanks
We're doing a project that requires H.323 to an Avaya. Does anyone have
experience to share on which H.323 driver to use in asterisk 1.6? Is the
diference between h323 and ooh323 still worth the extra effort? (We've only
installed h323 under 1.4)
If you have setup/config experience with this
: event
17 feb 2010 kl. 23.15 skrev Michelle Dupuis:
Is it possible to just send an event from one Asterisk server to another?
(Perhaps some custom event that I could define?) Or would that break
the SIP protocol/handling in asterisk?
I think this discussion would be easier if you told us what you
To: Asterisk Users List
Subject: Re: [asterisk-users] Access to header field: event
Michelle Dupuis wrote:
I'm trying to pass additional call information (eg: customer ID) to a
call center along with the call itself.
At this point I would be happy just seeing everything that I can get
from
I need to extract the event header info from an incoming SIP call. Is
this accessible from within the dialplan?
I've reviewed RFC 3265 but I'd like to start with just dumping everything to
do with event (if accessible, in other words Asterisk doesn't strip this
away)
Thanks!
MD
--
-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P.
Fleming
Sent: Wednesday, February 17, 2010 4:58 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Access to header field: event
Michelle Dupuis wrote:
*I need to extract the event header info
We have inherited an installation with Ast 1.4 and Aastra phones. The
client complains that sometimes the call audio turns tinny and robotic...I
heard it and it sounds wierd.
Has anyone else experienced this? Cause? Solutions?
Thanks,
MD
--
network) only?
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105
- Michelle Dupuis supp...@ocg.ca wrote:
We have inherited an installation with Ast 1.4 and Aastra phones. The
client complains that sometimes the call audio turns tinny and robotic...I
heard
Please use your quill and ink pot as well, and remember we can't insert
blank paper into the front of a book, only writing on blank pages at the
end.
Oh wait, the advent of computers has allowed us to conveniently insert the
most recent text at the TOP of a message, to prevent people from having
You can address the order of detection problem using udev rules...
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Chavez
Sent: Tuesday, January 12, 2010 6:53 PM
To: Asterisk Users List
Subject: Re:
We have that solution running fine...
Is your VPN termination a Linux box? Is it also the office router? Is it
also the firewall?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ryan McCormack
Sent: Monday,
What do you mean internal timing?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of --[ UxBoD ]--
Sent: Saturday, January 09, 2010 8:11 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] Choppy MOH
-
Could you explain this one a bit more...
You run openSER on the same box as asterisk, and have multiple such boxes,
with the purpose of failover? But if a box goes down with openser on it,
then there is no forwarding. (And most phones can only reg with peer). If
you move openSER to another
I wrote a script to check clients and restart asterisk if registrations died
(external IAX)...but you could modify for your needs. Check it out on
www.generationd.com
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On
Are you sure this isn't a Windows zeroconfig problem? If Win drops the
connection while * is talking to your client, the registration could drop
too..
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nic Colledge
Sent: Monday,
List
Subject: Re: [asterisk-users] Can't restart asterisk from script
On Wed, 9 Dec 2009, Michelle Dupuis wrote:
However, I have a cron job that tries to restart asterisk and gets
this
error:
No such command 'restart gracefully' (type 'help restart gracefully'
for other possible commands
I encounter an interesting situation where the internet connection goes down
and then goes back up. The IAX trunks are then unregistered, and * is
confused...only a restart allows * to function again. I have a cron script
that tests for an internet outage and then restarts * after the
obvious - there may also be privilege issues
BillK
On Wed, 2009-12-09 at 22:32 -0500, Michelle Dupuis wrote:
I had double quotes originally - and that didn't work
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com
I'm running * 1.4 and can successfully restart asterisk from the command
line with:
/usr/sbin/asterisk -r -x restart gracefully
However, I have a cron job that tries to restart asterisk and gets this
error:
No such command 'restart gracefully' (type 'help restart gracefully' for
other possible
To: Asterisk Users List
Subject: Re: [asterisk-users] Can't restart asterisk from script
Warren Selby wrote:
On Wed, Dec 9, 2009 at 3:08 PM, Michelle Dupuis supp...@ocg.ca
mailto:supp...@ocg.ca wrote:
I'm running * 1.4 and can successfully restart asterisk from the
command
line
script
Doug Lytle wrote:
Warren Selby wrote:
On Wed, Dec 9, 2009 at 3:08 PM, Michelle Dupuis supp...@ocg.ca
mailto:supp...@ocg.ca mailto:supp...@ocg.ca wrote:
I'm running * 1.4 and can successfully restart asterisk from the
command
line with:
/usr/sbin/asterisk -r -x
variables in the crontab to get it to work.
Billk
On Wed, 2009-12-09 at 20:55 -0500, Michelle Dupuis wrote:
Interesting...I'll try that. Thanks
__
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun
: [asterisk-users] Can't restart asterisk from script
You should replace the single quote with double quote.
--Original Message--
From: Michelle Dupuis
Sender: asterisk-users-boun...@lists.digium.com
To: 'Asterisk Users List'
ReplyTo: Asterisk Users Mailing List - Non-Commercial Discussion
Subject
To: Asterisk Users List
Subject: Re: [asterisk-users] Understanding Congestion to incoming caller
2009/11/17 Michelle Dupuis supp...@ocg.ca
I have an * installation which will refuse incoming callers once a max (5
callers) is reached. Caller 6 and up should be notified of
congestion...without
We have setup an * box for a small client with 10 phones. They have a
4500/500k ADSL connection which works great when no more than 8 external
calls are in progress. (ulaw)
The problem is when all 10 people try to use an external channel, AND/OR, 8+
incoming calls arrive at once. The symptom
I have an * installation which will refuse incoming callers once a max (5
callers) is reached. Caller 6 and up should be notified of
congestion...without network load on my trunk. How would I do this?
The voipinfo wiki shows playing a congestion tone to the caller, but that
seems stupid since
I'll start with a guess - your asterisk box or firewall is blocking SIP
ports. Diagnose that first (stop iptables/check iptables if unsafe) and try
again...
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
That may not work for all sip phones. Some (like xlite/eyebeam) crash when
receiving a text, others drop the subsequent call (Aastra 5x). These
observations are based on a project we did in late 2008; so be sure to do a
proof of concept before you get too deep into the project.
_
From:
Of Alex Balashov
Sent: Monday, November 09, 2009 9:50 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] Text messaging
What does Sendtext() actually do? Does it send a SIP request of method
MESSAGE? What does it do on a hardware channel - say, analog or TDM?
Michelle Dupuis wrote
I assume you're kidding?!
RTP is mangled/blocked by most hotspots and mid-size company firewalls...
IAX is often the only way our staff can connect while on the road.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On
with this problem
seriously.
I'll take your word for the fact that IAX may be easier, though.
Michelle Dupuis wrote:
I assume you're kidding?!
RTP is mangled/blocked by most hotspots and mid-size company firewalls...
IAX is often the only way our staff can connect while on the road
There is an admin manual you can download from Aastra..have you checked
there? (Not the user manual)
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Sunday, October 18, 2009 6:18 PM
To: Asterisk Users List
The 57i and 480i are good wireless phones but after 100ft you are out of
range (assuming business interiors). Of you still have to deal with buggy
firmware(and hit and miss tech support).
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Riddell
Sent: Thursday, October 08, 2009 10:47 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] g729 free codec any idea
On 9/10/09 3:31 PM, Michelle Dupuis wrote:
I believe that Intel placed a 729 codec into the public
I like the Qos functionality. Is that a linux based package available for
other distros?
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Knight
Sent: Thursday, October 08, 2009 11:15 AM
To: Asterisk Users List
Subject:
And how do you track incoming channels on this trunk?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Mathers
Sent: Thursday, October 08, 2009 2:01 PM
To: Asterisk Users List
Subject: Re:
Spinning off from another topic...what are people using for QoS / Shaping?
I'm using Wondershaper script with OK results...but I'd like better. Ideas?
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 - October
More specificallyI'm looking for a Linux package to allow shaping, QoS,
prioritization by port, etc.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle
Dupuis
Sent: Thursday, October 08, 2009 4:03 PM
To: Asterisk
I believe that Intel placed a 729 codec into the public domain (free), and
someone wrapped it in a nice Asterisk package for use.
No idea where - but I do recall that it is out there, and legal. Of course
it's nice to support a vendor, but free alternatives can't be shunned...
_
From:
Has anyone written an app that monitors SIP/IAX registration attempts? A
couple of clients are being flooded with SIP registrations (but the source
IP changes every few hours so IPtables won't do)..
I would think that any attempt to reg 5 times with a bad password should
cause a 5 minute
://lists.digium.com/pipermail/asterisk-users/2007-April/186456.html
On Fri, Oct 2, 2009 at 2:42 PM, Michelle Dupuis supp...@ocg.ca wrote:
Has anyone written an app that monitors SIP/IAX registration attempts?
A couple of clients are being flooded with SIP registrations (but the
source IP changes every few
...@lists.digium.com] On Behalf Of Michelle
Dupuis
Sent: Friday, October 02, 2009 2:24 PM
To: 'Asterisk Users List'
Subject: Re: [asterisk-users] app_hackblock to prevent SIP/IAX reg trolling
Good post. One of the recommendations is to limit the number of calls per
sip entity. Is there an easy way
Is it possible to set QOS/COS/DSCP on IAX packets? I see some parameters in
sip.conf but not iax.conf
Thanks
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...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle
Dupuis
Sent: Thursday, October 01, 2009 2:27 PM
To: Asterisk Users List
Subject: [asterisk-users] QOS/DSCP for IAX?
Is it possible to set QOS/COS/DSCP on IAX packets? I see some parameters in
sip.conf
-users-boun...@lists.digium.com] On Behalf Of Dave Fullerton
Sent: Thursday, October 01, 2009 3:01 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] QOS/DSCP for IAX?
Michelle Dupuis wrote:
I actually see the TOS setting in iax.conf, but the default (commented
out) is EF - which doesn't
I'm trying to enable res_crypto on a 1.4 installation, but menuconfig says
ssl is needed. I've installed openssl, openssl-devel, openssl-perl
but it's still not happy.
Anyone know what else is needed?
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