Are there deb packages available for Asterisk 10 or for 11 beta?
Kind Regards,
Chris
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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
> >> 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
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
On 12-08-28 10:25 AM, Matthew Jordan wrote:
- Original Message -
From: "Vladimir Mikhelson"
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Cc: asteriskt...@digium.com
Sent: Monday, August 27, 2012 7:33:27 PM
Subject: Re: [asterisk-users] Asterisk community services - O
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
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
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
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 can
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 Inc.)
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" wrote:
> 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
>
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
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, A
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" wrote:
> Extensions/trunks. Another thought is that you might make your
> modules.conf not load anyth
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 st
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
http://younewplanet.com/index.php/articles/2012-a
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 conver
On 08/28/2012 10:04 AM, Andrew Latham wrote:
> On Tue, Aug 28, 2012 at 11:00 AM, Johan Wilfer 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
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 Message--
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
--
__
On Tue, Aug 28, 2012 at 11:00 AM, Johan Wilfer wrote:
> 2012-08-28 16:44, Andrew Latham skrev:
>>
>> On Tue, Aug 28, 2012 at 10:39 AM, Johan Wilfer 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
2012-08-28 16:44, Andrew Latham skrev:
On Tue, Aug 28, 2012 at 10:39 AM, Johan Wilfer 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
http://www.digium.com/en/products/ivr/audio-con
On Tue, Aug 28, 2012 at 10:39 AM, Johan Wilfer 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 their recom
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 po
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 conver
- Original Message -
> From: "Vladimir Mikhelson"
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Cc: asteriskt...@digium.com
> Sent: Monday, August 27, 2012 7:33:27 PM
> Subject: Re: [asterisk-users] Asterisk community services - Old Mantis
> instance to be shutdow
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
[mailto:asterisk-users-boun...@lists.digium.co
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
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