both would be appreciated.
if you can send me a backtrace, that'd be great
On Jun 22, 2012, at 8:06 PM, Jeremy Kister wrote:
On 6/20/2012 8:24 AM, Darren Sessions wrote:
I just finished replying to your direct email (which you can disregard
now as this seems to be a different problem). I'm
Hi Jakob,
I just finished replying to your direct email (which you can disregard now as
this seems to be a different problem). I'm pretty sure I know what the issue
is, but I'll have to get back to you later this evening (my time).
- D
On Jun 20, 2012, at 4:41 AM, Jakob-Matthias Böttger
Hi folks,
Just a note to let everyone know I've finally finished up the new BETA release
of app_swift (now v3.0.1 b1).
This release introduces some pretty major changes to app_swift such as:
- The entire code-base has now been unified and the build system auto detects
which Asterisk version
Hi Folks,
After receiving a surprising amount of emails from Asterisk community
members, I thought I'd fire something off to the users list to clear
any confusion regarding the Asterisk Forge (forge.asterisk.org)
website and the future of the app_swift text-to-speech module.
With regards to the
Hey there folks,
I'd sent this to the list last night and got reject email this
morning. Apparently it is always a good idea to have an active
subscription to the list you are trying to post to - just one of those
things. :)
In any case, a new beta version of app_swift is available for Asterisk
You could use a sip proxy front end like Kamailio.
Sent from my iPhone
On Nov 18, 2010, at 7:39 AM, Antônio Theóphilo anto...@freeddom.com wrote:
Hi All
Does anyone know about any tool that does to Asterisk what mod_jk does for
JBoss/Tomcat: a load-balance/failover server that is
Well, the downside to wav files is the disk i/o. Asterisk will and does
translate the audio frames from ulaw to whatever other codec.
Sent from my iPhone
On Oct 24, 2010, at 9:42 AM, Zeeshan Zakaria zisha...@gmail.com wrote:
Do you recommend using wav files instead? Will there be any
Are you using app_swift or wav files?
On Oct 23, 2010, at 5:26 PM, Zeeshan Zakaria zisha...@gmail.com wrote:
Hello list,
I have been using Cepstral's 8KHz voices for my text-to-speech service for
some time now, and have been noticing that the voice quality is really poor,
doesn't
Just thought I'd let everyone know I've got a new beta version of app_swift up
for Asterisk 1.8 on http://forge.asterisk.org.
- Darren
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_
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New to Asterisk?
Hi all,
Thought I'd mention that the new version of the app_swift text-to-speech module
for Asterisk 1.2, 1.4, and 1.6 has been released at it's new home on the
Asterisk Forge.
http://forge.asterisk.org/gf/project/app_swift/
For those that are unaware, app_swift provides a direct interface
:05 AM, Darren Sessions wrote:
Hi all,
I'm trying to get the MeetMe system to take a caller and announce to them
they've joined the conference in addition to the other members of the
conference assuming previous members of the conference = 1.
I can see where the meetme.c app actually
Hi all,
I'm trying to get the MeetMe system to take a caller and announce to them
they've joined the conference in addition to the other members of the
conference assuming previous members of the conference = 1.
I can see where the meetme.c app actually processes it using the
What version of Asterisk and what version of app_swift?
On 29 Oct 2008, at 15:10, [EMAIL PROTECTED] wrote:
Hi, I have tried installing app_swift on both mac os x and ubuntu now
and am getting the same error. I must be missing something, as I have
tried multiple versions and everytime do sudo
Not sure about the Swedish, but Lumenvox has a great speech
recognition app for Asterisk.
- D
On 26 Oct 2008, at 19:53, Christian wrote:
Hi all,
Yes, this might not be the proper list for this, but i have a
question about Asterisk and voice recognition.
If I want to create a menu
thanks,
Christian
On 2008-10-26 at 20:32 Darren Sessions wrote:
Not sure about the Swedish, but Lumenvox has a great speech
recognition app for Asterisk.
- D
On 26 Oct 2008, at 19:53, Christian wrote:
Hi all,
Yes, this might not be the proper list for this, but i have a
question about
actually wrote one of these ages ago that worked fairly well with
a10 calls per second SER server. How many calls per second are you
looking to process?
- D
_
Darren Sessions
[EMAIL PROTECTED]
http://www.darrensessions.com
I know. :)
I've already mentioned some of the OpenSIPS options to him on the
OpenSIPS users list (LCR module specifically). Just brain dumping
everything that came to mind.
- D
_
Darren Sessions
[EMAIL PROTECTED]
http://www.darrensessions.com
with OpenSER for
such a small amount of users.
Asterisk can do everything you'll need it to do otherwise.
- D
_
Darren Sessions
[EMAIL PROTECTED]
http://www.darrensessions.com
_
On Oct 1, 2008, at 7:44 PM, Alex Balashov wrote
Any particular reason you're using H323 instead of SIP ?
_
Darren Sessions
[EMAIL PROTECTED]
http://www.darrensessions.com
_
On Sep 16, 2008, at 12:04 PM, Guilherme Loch Waltrick Góes wrote:
I have a Cisco 3845 with a ISDN PRI port
this type of general setup in the past with a
great deal of success for remote offices and soft-phones on laptops.
_
Darren Sessions
[EMAIL PROTECTED]
http://www.darrensessions.com
_
On Sep 9, 2008, at 1:19 PM, Mattias Andersson wrote
You shouldn't have any delays at all.
Are you using ztdummy for timing? and what kind of load does the box
have on it?
_
Darren Sessions
[EMAIL PROTECTED]
http://www.darrensessions.com
_
On Sep 9, 2008, at 4:23 PM, George
A cheaper alternative would be the voip wiki.
http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf
_
Darren Sessions
[EMAIL PROTECTED]
http://www.darrensessions.com
_
On Sep 4, 2008, at 12:13 PM, Mark
Impressive work Bradley! I tested it and it worked great, even with my
mandatory 'use strict'.
Thanks,
- Darren
_
Darren Sessions
[EMAIL PROTECTED]
http://www.darrensessions.com
_
On Aug 29, 2008, at 5:47 AM, Watkins, Bradley
-verbose(”No subroutine name passed!!”, 1);
return(-1);
}
my $exec = \{$sub};
return($exec-());
}
set_variables();
dynamic_execute(”run_me”);
_
Darren Sessions
[EMAIL PROTECTED]
http://www.darrensessions.com
_
smime.p7s
Description
Are you using an Asterisk PBX?
_
Darren Sessions
[EMAIL PROTECTED]
http://www.darrensessions.com
_
On Aug 27, 2008, at 7:06 PM, Tom Moore wrote:
Hi guys,
What are your suggestions to people who have pbx systems that
interface
You can use an extremely simple Asterisk config to do the SIP-PRI
call conversion that'd be very solid.
_
Darren Sessions
[EMAIL PROTECTED]
http://www.darrensessions.com
_
On Aug 27, 2008, at 7:37 PM, Tom Moore wrote
, then that would also explain why outbound PSTN DTMF is
functional.
Hope this helps.
_
Darren Sessions
[EMAIL PROTECTED]
http://www.darrensessions.com
_
On Aug 23, 2008, at 12:39 AM, Max Alex wrote:
Hi everybody,
I have linksys phone
satellite.
_
Darren Sessions
[EMAIL PROTECTED]
http://www.darrensessions.com
_
On Aug 23, 2008, at 4:45 PM, Femi wrote:
I’ve used VOIP over satellite for years and while it’s not perfect
it is sometimes actually better than cellular
Just change your dial command and add the plus sign there.
_
Darren Sessions
[EMAIL PROTECTED]
http://www.darrensessions.com
_
On Aug 22, 2008, at 1:28 AM, ronald wrote:
Hi,
Is it possible to assign a plus sign on the callerid(num
, and the problems lies somewhere on your network between the
Asterisk server and whatever gateway / device. If it sounds awful, and
the codecs match, then it's time to start troubleshooting the server.
_
Darren Sessions
[EMAIL PROTECTED]
http://www.darrensessions.com
Not sure what you've heard before, but I have successfully used a
modem at 9600 baud (forced via AT commands) through a zaptel card on
several occasions.
_
Darren Sessions
[EMAIL PROTECTED]
http://www.darrensessions.com
_
On Aug
We recently discussed DeadAGI on the list - I'd check the archives
first.
I just finished doing a write up on DeadAGI and Perl on my website if
you're interested.
DeadAGI *can* be very reliable if done properly.
- Darren
_
[EMAIL PROTECTED]
I'd run top on the server to see if the CPU utilization is going
through the roof. If you use AGI, make sure there aren't any orphaned
processes consuming resources.
If all else fails on the software side of things, I'd restart the
server.
_
Darren Sessions
Ruddy,
I've used deadagi for years with perfect success.
If it's a perl agi module, you need to make absolutely sure that
you're using 'use strict' and 'use warnings' in the main agi file -as
well- as any includes. You'll need to test your agi while in console
mode, so any of the perl
Another thing you may want to do is try a simple ping test to the far
end host. While this may not always be a reliable way to test lag
given that the far end maybe just a proxy and your RTP may be
terminating to another device, it still should give you a good idea
what your lag times are
Set it so when they dial the number, it calls an AGI script that
instantly answers and generates a call file and hangs up. That way,
you could dial and then hangup, and the system generates a call file
that calls the door phone and does whatever it needs to do separate of
the initial call.
Here is a simple Perl implementation to generate call files . .
You'll still need something for it to execute after the call files are
generated; either a simple AGI app that streams a file, a Macro, or a
nice dialplan layout.
In any case, you could call something like this very rapidly
I would make absolutely sure you've got your linux distro's version of
libgsm installed. I can't really speak to the difference between those
two versions of Asterisk without looking at a change-log, but I highly
doubt a serious modification to the gsm code took place between sub-
versions.
thing is: if the softphone is
using GSM the sounds is perfect, if I use Alaw as the softphone
CODEC the sounds is pretty bad. The softphone is in the same LAN as
the Asterisk server, so I don't think it's a bandwidth issue.
Best Regards,
On Wed, Aug 6, 2008 at 10:13 AM, Darren Sessions
I have used virtually all versions of Asterisk 1.0+ (literally, either
in production or testing) with OpenSUSE 10+ and 11 on AMD and Intel
and haven't had any issues with gcc optimizations with regards to
audio sounding choppy. This scenario for me has always been the gsm
libs.
I can speak first hand to this having gone through it just a few
months ago . .
After being spoiled with all the features and standard compliance in
Postgres, I was put in a position with a new project to setup a
redundant (Master-Slave) database cluster.
I immediately jumped to Postgres
I've updated the app_flite module to work with the Asterisk 1.6.x code-
base in addition to it already working with the 1.4.x, and 1.2.x.
(1.0.x support is untested and unsupported).
It can be downloaded on my website at:
http://www.darrensessions.com/downloads/app_flite-0.6.tar.gz
If you had a dax in front of all your circuits, you could move them
from one server to another without physically touching anything.
I've done about 300 calls on a dual processor box doing just SIP with
an entirely AGI based setup and it held up just fine, but doing TDM,
I'd worry about
2008-07-08 - app_swift v1.6.2 released for Asterisk 1.6.x code-base
---
Added support for handling multiple dtmf input
Added support for input timeout and max input digits (similar to
AGI's get_data)
Ignores DTMF if no timeout and max digits
2008-07-09 - app_swift v1.2.2 released for Asterisk 1.2.x code-base
---
Added support for handling multiple dtmf input
Added support for input timeout and max input digits (similar to
AGI's get_data)
Ignores DTMF if no timeout and max digits
2008-07-08 - app_swift v1.4.2 released for Asterisk 1.4.x code-base
---
Added support for handling multiple dtmf input
Added support for input timeout and max input digits (similar to
AGI's get_data)
Ignores DTMF if no timeout and max digits
Thought I'd let everyone know I've released app_swift v1.6.1 which is
entirely based off of Will Orton's work he's placed in the public
domain.
Works great with Asterisk v1.6.0-beta7.1.
In any case, can be downloaded from my site at:
http://www.darrensessions.com
Go easy on me, this is my
Is there a way to retrieve the Call-ID from a call made using the 'Dial'
command on a SIP channel without CDRs (i.e. variable) ?
Thanks,
- Darren
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Darren Sessions wrote:
Is there a way to retrieve the Call-ID from a call made using the 'Dial'
command on a SIP channel without CDRs (i.e. variable) ?
(sometimes I wonder why we write documentation)
doc/README.variables has ${SIPCALLID} documented
@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Darren Sessions wrote:
If try and read in the SIPCALLID variable (which I already do on the
incoming call) after the dial, I still get the incoming call's call-id.
Your explanation could
*** If anyone has a better way of doing this, please post to the list. I
hadn't seen anything on this list or in channel.c/chan_local.c - which
prompted this email ***
I'm not sure how many VoIP providers out there are using Asterisk as a
service platform like we do, but I thought I'd share
I've been doing AGI now for 2 years, and this problem is making me feel
like I just started. :) I don't have this problem on pre 1.2
installations, so I'm assuming either this is something new, or I've
missed something in the change logs or on wiki.
Scenario:
Customer disables caller id on
Is there a way to play gsm audio files on Windows Media Player ?
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Does such a thing exist?
Here is my problem. I've got 300+ people that want to be on a single
conference call. Not sure if a single Asterisk server could survive it.
I was thinking of putting trunks in between the servers - but quickly
realized I'm just giving the audio an extra HOP to traverse
Or for that matter, is there a planned G729 binary for Mac OSX ?___
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Amen
On Oct 22, 2004, at 1:26 PM, Kevin Walsh wrote:
Kanuri, Seshu (Company IT) [EMAIL PROTECTED] lazily
top-posted:
Just my $0.02 Cents
I propose that an Asterisk development fund be set up to hold all of
these $0.02 donations. People who are not quite as cheap could donate
a little bit more.
When I execute the following AGI command in *, if the caller hangs up
during the record - it fails to run the callback sub -BUT- during any
other portion of the call, if the caller hangs up then it gets called
just fine.
Here are some code excerpts:
use Asterisk::AGI;
$AGI = new Asterisk::AGI;
Noticed it here too.
On Oct 21, 2004, at 10:58 AM, Joseph wrote:
Using cvs build from CVS-HEAD-10/15/04-06:13:19
it seems the the mwi is randomly not lighting the phone when there is a
message.
Has any one else noticed this?
Sometimes it works, sometimes it seems to *miss* messages.
Using mostly
Does the -EXACT- same thing if I do a straight print on the record
command.
$rc = print STDOUT RECORD FILE tmp_msgs/$sessionId wav #*0 7;
On Oct 21, 2004, at 11:00 AM, Darren Sessions wrote:
When I execute the following AGI command in *, if the caller hangs up
during the record - it fails
We use SER + Asterisk. One heck of a powerful combination.
On Oct 21, 2004, at 11:59 AM, Nahuel Alejandro Ramos wrote:
Hi everyone,
I have some doubt about use or not to use SER.
I need a solution using a single linux box that manages, aproximatly
500-1000 registred SIP users, but not more
On 10/21/2004, Darren Sessions [EMAIL PROTECTED] wrote:
We use SER + Asterisk. One heck of a powerful combination.
On Oct 21, 2004, at 11:59 AM, Nahuel Alejandro Ramos wrote:
Hi everyone,
I have some doubt about use or not to use SER.
I need a solution using a single linux box that manages
SER most definitely does CDR archiving via MySql database. It's a
hellaciously fast and stable proxy - sounds like it'd be a good choice
for the core of your network with all the different components.
On Oct 19, 2004, at 10:01 AM, Andreas Anderson wrote:
Hi Guys,
i need to do some kind of CDR
Someone should put a bounty on T38. We're using spandsp right now and
have had success - but it was an absolute pain to get it to work.
On Oct 19, 2004, at 12:38 PM, Steve Underwood wrote:
Michael Loftis wrote:
Just my $0.02 but seems to me the VoIP community as a whole needs to
extend SIP (or
Darren Sessions wrote:
Someone should put a bounty on T38. We're using spandsp right now and
have had success - but it was an absolute pain to get it to work.
On Oct 19, 2004, at 12:38 PM, Steve Underwood wrote:
Michael Loftis wrote:
Just my $0.02 but seems to me the VoIP community as a whole needs
Is there a way to stream or at least load into a variable with AGI, gsm
or wav files out of a MySql database (contained in MySql as blob
fields) directly from asterisk without having to write the files to
disk first before you stream them out?
I've seen a hack for mpg123 that lets you open
The PAP2 is essentially a Sipura. Other than the different skin, a
couple cool L.E.Ds, and an updated web interface - they might as well
be the same box. Linksys's entire line of VoIP boxes are based on the
Sipura technology.
Our experience has been that the Sipura rules supreme in features
Ok.. total brain fart.. sorry..
lol
:)
On Oct 19, 2004, at 5:55 PM, Matthew Boehm wrote:
2 PAP2NA's with 2 ports each = 4 lines
Matthew
- Original Message -
From: Darren Sessions [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Tuesday
Is there a way to get the Call ID off of a call that runs through *
without loading any kind of billing CDR platform?
If not, I think it would be a great addition to * if the Call ID was
passed as variable (in AGI).
Thanks,
- Darren
___
Call-ID as in SIP Call-ID *not* Caller ID.
:)
Thanks though Danny.
On Oct 18, 2004, at 10:02 AM, Danny Froberg wrote:
Hi Darren,
It is today, check the variables CALLERID, CALLERIDNUM CALLERIDNAME
/Danny
At 15:58 2004-10-18, you wrote:
Is there a way to get the Call ID off of a call that runs
Critchfield wrote:
On Mon, 2004-10-18 at 09:58 -0400, Darren Sessions wrote:
Is there a way to get the Call ID off of a call that runs through *
without loading any kind of billing CDR platform?
If not, I think it would be a great addition to * if the Call ID was
passed as variable (in AGI).
It would
WORKS PERFECT !!!
THANK YOU !!!
:)
On Oct 18, 2004, at 3:55 PM, Olle E. Johansson wrote:
Darren Sessions wrote:
Call-ID as in SIP Call-ID *not* Caller ID.
In chan_sip2: ${SIPCALLID}
Very useful, indeed.
And looking at the chan_sip source code, I've obviously ported it to
standard Asterisk as well
Works with Verizon and G729. I've got a Samsung i700 - works like a
champ! If you're in a moving vehicle it can get choppy depending on
signal strength - but works well.
On Oct 18, 2004, at 5:01 PM, Brian McSpadden wrote:
It kind of works...I've done it from my notebook. I wouldn't use it
all
I can tell you from first hand experience that unless you've got +1000
extensions completely configured, it's not a problem in the slightest.
After that, you'll start getting to many files open messages (on a
vanilla system install) and the server will go temporarily unresponsive
(which can be
I've use Sipuras with * using G729 - with no problems.
Double check that G729 is turned on in the sipura and your sip.conf is
correct - if anything post excerpts from your sip.conf.
On Oct 16, 2004, at 6:27 AM, Jefferson Carvalho wrote:
Hello All,
I purchased yesterday two G729 licenses from
http://sourceforge.net/projects/wifi-box/
On Oct 14, 2004, at 3:43 PM, TC wrote:
I run asterisk at my house on a linksys router. I have it sitting in
the DMZ of the router so it acts like its outside.
Works perfectly fine.
is this a wrt54gs ?
if so did you get this to compile with the openwrt54
Duh.
Simply posting another interesting link. Smart guy.
On Oct 14, 2004, at 5:03 PM, Jeremy McNamara wrote:
Darren Sessions wrote:
http://sourceforge.net/projects/wifi-box/
Yo, smart guy this thread is about running asterisk ON the WRT54GS.
Jeremy McNamara
Why not use an NTP timing source - go stratum 2 or 3. That should be
plenty for a stable clock source.
On Oct 12, 2004, at 9:52 AM, Eric Wieling wrote:
Roy Sigurd Karlsbakk wrote:
hi
with silence suppression enabled I get these:
Oct 12 15:45:55 NOTICE[1104014256]: rtp.c:289 process_rfc3389:
of Asterisk to
understand what kind of timing you're after. I assumed you were just
after a reference clock.
On Oct 12, 2004, at 10:12 AM, Christopher L. Wade wrote:
Darren Sessions wrote:
Why not use an NTP timing source - go stratum 2 or 3. That should be
plenty for a stable clock source.
*Timing
I am getting some weird behavior and a rash of interesting messages in
the log files. If anyone has some ideas, it would be appreciated.
Using Asterisk v1.0.1 on Suse Enterprise Linux v8.0. HP DL380 Server.
4GB Ram - Dual 3.2ghz processors.
This first entry is when asterisk simply goes
What is the rational for only supporting 32kbps G726 and not 16kbps?
Thanks,
- Darren
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I've changed the spool directory in asterisk.conf to point to a different
directory. Everything works/gets created just fine with the exception of the
unavailable messages. When a user tries to create one, I get this on the
console (below).
I changed the directory to /vm in asterisk.conf.
Any
Fyi,
Successfully compiled Asterisk on an Apple G4 PPC with Yellow Dog Linux -
without any source modifications.
Worked fast and smooth.
- Darren
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I've got Asterisk loading 100,000+ extensions in extensions.conf. This
process is taking a little upwards of 10 minutes to complete on each of my
dual 3.2Ghz HP DL380 with SuSE Linux Enterprise 8 boxes.
Although asterisk creates child processes, it appears that it is only using
a single processor
Ill apologize right away for asking stupid questions.
J
System Setup:
SER = Proxy
Asterisk = Voicemail
All sip based setup.
What Is required to make
asterisk NOT- accept inbound calls/signaling from an unknown host?
I tried the peers in sip.conf but it still
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