I came up with the problem of Asterisk AGI commands length limitation
when started scripting IVRs. I have some Background commands with long
(theoretically unlimited) arguments (list of promts concatenated with
). See https://issues.asterisk.org/jira/browse/ASTERISK-20294 for details.
Please
Extensions/trunks. Another thought is that you might make your modules.conf
not load anything to start with so you can eliminate a rogue module as the
problem. Just change autoload=yes to autoload=no.
From: asterisk-users-boun...@lists.digium.com
- Original Message -
From: Vladimir Mikhelson v...@mikhelson.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Cc: asteriskt...@digium.com
Sent: Monday, August 27, 2012 7:33:27 PM
Subject: Re: [asterisk-users] Asterisk community
Hi,
I've used the shells-script at the end of this email to generate 8khz
mono wave-files for asterisk from a 144 khz recording.
The script does two things: resample normalize the audio volume.
Anyone like to share their recommendations / scripts for doing this
conversion? I've just
This isn't required, but you will notice a big quality difference if you run
the normalized files through Audacity and set the volume to -3
(Recommendation from the Asterisk PDF). There's most likely a way to do
this with SOX, I just haven't tried hard enough to find it. If/when you do,
please
On Tue, Aug 28, 2012 at 10:39 AM, Johan Wilfer li...@jttech.se wrote:
Hi,
I've used the shells-script at the end of this email to generate 8khz mono
wave-files for asterisk from a 144 khz recording.
The script does two things: resample normalize the audio volume.
Anyone like to share
2012-08-28 16:44, Andrew Latham skrev:
On Tue, Aug 28, 2012 at 10:39 AM, Johan Wilfer li...@jttech.se wrote:
Hi,
I've used the shells-script at the end of this email to generate 8khz mono
wave-files for asterisk from a 144 khz recording.
Try this to test with
On Tue, Aug 28, 2012 at 11:00 AM, Johan Wilfer li...@jttech.se wrote:
2012-08-28 16:44, Andrew Latham skrev:
On Tue, Aug 28, 2012 at 10:39 AM, Johan Wilfer li...@jttech.se wrote:
Hi,
I've used the shells-script at the end of this email to generate 8khz
mono
wave-files for asterisk from a
2012-08-28 17:04, Andrew Latham skrev:
Yep, check out repotools for that
http://svn.asterisk.org/svn/repotools/sound_tools/scripts/
Cool! Thank you!
--
Johan Wilfer
JT Technologies Telecommunications AB
Jabber: jo...@jttech.se | Phone: +46 31 3809100
--
Does the .c program compile stand-alone or as an add-on?
g++ check_sounds.c
check_sounds.c: In function âint main(int, char**)â:
check_sounds.c:152: error: invalid conversion from âvoid*â to âdirent**â
check_sounds.c:154: error: invalid conversion from âvoid*â to âdirent*â
-Original
On 08/28/2012 10:04 AM, Andrew Latham wrote:
On Tue, Aug 28, 2012 at 11:00 AM, Johan Wilfer li...@jttech.se wrote:
2012-08-28 16:44, Andrew Latham skrev:
Try this to test with
http://www.digium.com/en/products/ivr/audio-converter.php and compare
your output first...
Interesting. Didn't
On 08/28/2012 10:32 AM, Danny Nicholas wrote:
Does the .c program compile stand-alone or as an add-on?
g++ check_sounds.c
check_sounds.c: In function âint main(int, char**)â:
check_sounds.c:152: error: invalid conversion from âvoid*â to âdirent**â
check_sounds.c:154: error: invalid conversion
Hi
I wrote this article and at end i shared how to convert files have a look
http://younewplanet.com/index.php/articles/2012-articles-2/asterisk-configuration-step-by-step
Also i wrote an other article for file conversion you can also check that
Hi,
I'm banging my head on Freepbx 2.10 setup with which a SIP hardphone can
subscribe to some Freepbx-generated hints and still cannot subscribe to
other Freepbx-generated hints (404 Not Found).
I would be very curious to learn here a bit more about how Asterisk 1.8
(and above) deals with hint
I tried that too, what happens is asterisk is loading but after that if I
try to start any one module for example chan_sip.so, asterisk hangs.
Regards.
On Aug 28, 2012 6:44 PM, Danny Nicholas da...@debsinc.com wrote:
Extensions/trunks. Another thought is that you might make your
modules.conf
Check Jason Parker's post from today and see if you skipped any of the
preliminary build steps. It is possible that something like libpri is
biting you.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gopalakrishnan
N
Sent: Tuesday,
I am trying to record calls on demand both inbound and outbound calls. I can
record outbound calls just fine but not inbound calls or calls from an
internally between extensions. I am using the latest asterisk 1.8.x certified
version.
On an outbound call I see:
== Using SIP RTP CoS mark 5
If I don't need to install dahdi hardware, is it really I need to have
libpri installed?
Regards.
On Aug 28, 2012 10:26 PM, Danny Nicholas da...@debsinc.com wrote:
Check Jason Parker’s post from today and see if you skipped any of the
preliminary build steps. It is possible that something
I would install both dahdi and libpri. I brought up a 12.2 RC-2 VM on hyper-v
Windows 8 and followed our standard asterisk build and have no issues yet but
we have not run full testing to confirm. Also a point of not 12.2 is RC for
the next 8 days or so.
Thanks
Bryant Zimmerman (ZK Tech
PS: Another question
Let my system is configured with 2 hints like this :
*2711@timeconditions-toggle: Custom:TC11
State:InUse Watchers 0
6452@ext-local :
SIP/6452 State:Unavailable Watchers 0
Let say I
On 08/28/2012 01:51 PM, Olivier wrote:
Let say I cannot touch the files in which those 2 instructions are set:
[timeconditions-toggle]
exten = *2711,hint,Custom:TC11
...
[ext-local]
exten = 6452,hint,SIP/6452
...
Then what can I do allow a given SIP phone to successfully subscribe
to both
Hi,
I recently replaced a site that was using 1.4.[mumble] with
hylafax/iaxmodem. They have an RBS T1 and were using about half of
their 50 DID numbers for fax to email. This all broke with the new
system :(
The original chan_dahdi.conf had no mention of faxdetect, so I assume
it was
IIRC correctly this is sort of like the s extension; you set up your fax
handler in [default,fax,1]. Not sure how that is done in FreePBX.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
LaCoursiere
Sent:
On 12-08-28 10:25 AM, Matthew Jordan wrote:
- Original Message -
From: Vladimir Mikhelson v...@mikhelson.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Cc: asteriskt...@digium.com
Sent: Monday, August 27, 2012 7:33:27 PM
Subject: Re:
Hi,
I recently replaced a site that was using 1.4.[mumble] with
hylafax/iaxmodem. They have an RBS T1 and were using about half of
their 50 DID numbers for fax to email. This all broke with the new
system :(
The original chan_dahdi.conf had no mention of faxdetect, so I assume
it was
It allows to productively search and work with issues
recorded in it.
Search, convenient straight forward layout, patch download URLs,
everything just works there.
JIRA maybe is convenient for the management and
developers. I just
guess, as somebody must have loved it so it was
You should simplify until you have something that works, then add your
conditions back in one line at a time.
On 12-08-28 11:05 AM, Josh Hopkins wrote:
-- Executing [s@macro-one-touch-record:3]
ExecIf(SIP/1010-0161, 1?MacroExit()) in new stack
This is where the inbound call is
Are there deb packages available for Asterisk 10 or for 11 beta?
Kind Regards,
Chris
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