What you mean is seeing only 80% of me. :-)
On Wed, Sep 13, 2023 at 1:10 PM Joshua C. Colp wrote:
> Glad to hear it, and look forward to seeing you there!
>
> On Wed, Sep 13, 2023 at 7:04 AM Nir Simionovich
> wrote:
>
>> Hi All,
>>
>> After a relatively l
Hi All,
After a relatively long hiatus and a very brief visit last year, Eric and I
are going to be back at ITExpo in full force.
Looking forward to meeting everybody.
On Tue, Sep 5, 2023 at 7:37 PM Joshua C. Colp wrote:
> On Tue, Sep 5, 2023 at 12:41 PM Fred Posner wrote:
>
>> +1 regarding
The usage of AMR codecs is mostly in the LTE space - which means that
you're probably aiming to run a slightly higher scale than a single
Asterisk server.
Why not simply install Kamailio+rtpengine as your signalling media
front-end and then relay everything back to Asterisk using a low-footprint
Nir S
--
Kind Regards,
Nir Simionovich
GreenfieldTech
(schedule) http://nirsimionovich.appointy.com/
(w) http://www.greenfieldtech.net
(p) +972-73-2557799(MSN): n...@greenfieldtech.net
(m) +972-54-6982826 (GTALK): nir.simionov...@gmail.com
(f) +972-73-2557202
Issue reported as ASTERISK-27545.
On Thu, Jan 4, 2018 at 8:36 AM Nir Simionovich <nir.simionov...@gmail.com>
wrote:
> Hi All,
>
> We've recently encountered an interesting bug with Asterisk 13 (the
> version we are testing with), but I believe
> as this is a fairly crazy
plan function to perform requests. Dialplan
>> functions should be for dialplan, in general I think they should not be
>> used as internal API's.
>>
>> On 12/22/2017 12:23 PM, Nir Simionovich wrote:
>>
>> Well,
>>
>> We can start with that implem
w Fredrickson
>
>
>>
>> On 22 December 2017 at 16:54, Matt Fredrickson <cres...@digium.com>
>> wrote:
>>
>>>
>>>
>>> On Fri, Dec 22, 2017 at 6:58 AM, Nir Simionovich <
>>> nir.simionov...@gmail.com> wrote:
>>>
>>
though the memory requirement for such a large dataset goes to 48GB . So I
> strongly believe that for such key value pair REDIS will be the right
> choice for ASTDB.
>
> Regards,
>
> Abhay
>
> On 22-Dec-2017, at 5:52 PM, Nir Simionovich <nir.simionov...@gmail.com>
> wr
to do so.
--
Kind Regards,
Nir Simionovich
GreenfieldTech
(schedule) http://nirsimionovich.appointy.com/
(w) http://www.greenfieldtech.net
(p) +972-73-2557799(MSN): n...@greenfieldtech.net
(m) +972-54-6982826 (GTALK): nir.simionov...@gmail.com
(f) +972-73-2557202
well, that was exactly my feeling.
On Mon, Nov 27, 2017 at 3:10 PM Joshua Colp <jc...@digium.com> wrote:
> On Mon, Nov 27, 2017, at 08:55 AM, Nir Simionovich wrote:
> > @corey,
> >
> > I've been looking into res_sorcery_astdb, but I think I'm missing
> >
it correct?
On Thu, Nov 23, 2017 at 2:03 AM Nir Simionovich <nir.simionov...@gmail.com>
wrote:
> Actually, I was more thinking about Redis as a PubSub mechanism, not as a
> static storage backend.
>
> Here is my take on things, developers need tools. Some developers prefer
> Re
s" in cdr_redis.c so menuselect
> will enable the module.
>
> Since redis is in-memory I'm not really sure about using it for CDR? I
> could see res_sorcery_redis being useful assuming it could be used as an
> alternative to res_sorcery_astdb or res_sorcery_memory.
>
> O
it's on gerrit - it's inside asterisk-team
On Thu, Nov 23, 2017 at 1:11 AM Corey Farrell <g...@cfware.com> wrote:
> Where did you push this branch? I'm not seeing it on gerrit or github.
>
> On 11/22/2017 06:01 PM, Nir Simionovich wrote:
>
> Hi All,
>
> I've s
it if someone can take a look for a minute and
see if I missed anything major in there.
--
Kind Regards,
Nir Simionovich
GreenfieldTech
(schedule) http://nirsimionovich.appointy.com/
(w) http://www.greenfieldtech.net
(p) +972-73-2557799(MSN): n...@greenfieldtech.net
(m) +972
nfiguration construct.
Any assistance would be appreciated.
P.S.
Already looked into app_skel, didn't really provide me the information I
was looking for.
--
Kind Regards,
Nir Simionovich
GreenfieldTech
(schedule) http://nirsimionovich.appointy.com/
(w) http://www.greenfieldt
convention point of view, would you believe building two
packages named kamailio4 and
kamailio5 be beneficial?
On Sun, Nov 19, 2017 at 12:56 PM Olle E. Johansson <o...@edvina.net> wrote:
>
> On 16 Nov 2017, at 22:18, Nir Simionovich <nir.simionov...@gmail.com>
> wrote:
>
ll be able to understand it from the code. I think the best would be
>> for me to upload the SRPM in there as well, so people
>> can use that as well at ease.
>>
>>
>>
>> On Fri, Nov 17, 2017 at 5:43 PM Nir Simionovich <
>> nir.simionov...@gmail.com> wro
Asterisk Opus
repo, for Asterisk 13.7 as a tar.gz file. I'm confident
you'll be able to understand it from the code. I think the best would be
for me to upload the SRPM in there as well, so people
can use that as well at ease.
On Fri, Nov 17, 2017 at 5:43 PM Nir Simionovich <nir.simio
oh, the repos also have an SRPM repo in there, so you can also install from
there as well to see the SPEC file.
At least, before I go ahead and start the public repo. Maybe I should add
it to the "Documentation project" repo?
On Fri, Nov 17, 2017 at 5:41 PM Nir Simionovich <
No problem guys - I'll create respective repos on github later today.
On Fri, Nov 17, 2017 at 4:40 PM Jared Smith <jaredsm...@jaredsmith.net>
wrote:
> On Thu, Nov 16, 2017 at 4:18 PM, Nir Simionovich <
> nir.simionov...@gmail.com> wrote:
>
>> As part of our work,
packaging in the near future - once I get
around to finalizing my current work path.
Would appreciate some feedback and ideas.
--
Kind Regards,
Nir Simionovich
GreenfieldTech
(schedule) http://nirsimionovich.appointy.com/
(w) http://www.greenfieldtech.net
(p) +972-73-2557799
Well,
As long as the wiki entry indicates the original source and retains the
CC license, I have no
problem with "Cross publication".
On Tue, Nov 7, 2017 at 12:59 AM Matt Fredrickson <cres...@digium.com> wrote:
> On Sat, Nov 4, 2017 at 2:53 PM, Nir Simionovich <nir
.
For the time being, I will serve as both writer and curator - till other
people step in and provide additional assistance.
Regards,
Nir Simionovich
--
Kind Regards,
Nir Simionovich
GreenfieldTech
(schedule) http://nirsimionovich.appointy.com/
(w) http://www.greenfieldtech.net
(p
; don't have access to test any lower version.
>
> -Corey
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-dev mailing list
> To UNSUBSCRIBE or update options visit:
>ht
t;best practice" example in the Wiki, people will
do as they see fit, which in turn will
turn into a "review board" ping-pong, which can be avoided by a simple
sample in there.
On Mon, Oct 30, 2017 at 9:22 PM Kevin Harwell <kharw...@digium.com> wrote:
> On Mon, O
c/AMI/AGI that all astdb operations are serialized (dblock global
> mutex) and thus performance could suffer if used too much from too many
> threads? Do we have any guides/sample files showing how to replace astdb
> operations with alternatives (func_odbc for example)?
>
> On 1
t sure this is worth the effort at this point in time, maybe
in a later stage. :-(
On Thu, Oct 26, 2017 at 6:01 PM Nir Simionovich <nir.simionov...@gmail.com>
wrote:
> Correction, seems like this requires a bit more architecture than I
> anticipated.
>
> Basically, we need to sep
a pluggable module is a mandatory requirement for Asterisk to launch
correctly?
Is there anything like that in Asterisk? can someone point me in some
proper example
or preferably, something that I can look at and learn from?
On Thu, Oct 26, 2017 at 4:47 PM Nir Simionovich <nir.simionov...@gmail.
mt, "");
DEFINE_REDIS_STATEMENT(get_redis_stmt, "");
DEFINE_REDIS_STATEMENT(del_redis_stmt, "");
Following this, we can simply point to the proper statements following the
engine selection.
What do you think, sounds reasonable?
On Thu, Oct 26, 2017 at 4:30 PM Olle E. Johansson
. Johansson <o...@edvina.net> wrote:
> On 26 Oct 2017, at 15:20, Nir Simionovich <nir.simionov...@gmail.com>
> wrote:
>
> Just looked into the code, this is not a simple task to put a new backend
> for astdb. The code isn't even designed
> for something like that. Judging fro
> calls happening before realtime got launched, but otherwise it worked just
> fine in production for a long time.
>
> /O
>
> On 26 Oct 2017, at 15:13, Nir Simionovich <nir.simionov...@gmail.com>
> wrote:
>
> I'd like to +1 on that idea.
>
> While I'm somew
p on
> gerrit at some time in the future. :-)
>
> --
> Matthew Fredrickson
> Digium, Inc. | Engineering Manager
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
>
;
> --
> _
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> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-dev
-
Hi All,
Following cdr_beanstalk, I've added a beanstalk backend to CEL as well.
I'll be uploading
that to geritt review under a new review issue.
On Mon, Oct 16, 2017 at 6:35 PM Nir Simionovich <nir.simionov...@gmail.com>
wrote:
> Hi All,
>
> So, my new cdr_beanstalkd modu
is one :-)
On Mon, Oct 16, 2017 at 5:02 PM Nir Simionovich <nir.simionov...@gmail.com>
wrote:
> Thanks. The module is now finished - and also tested. I want to generate
> some tests and make sure
> it holds up, but in general - it's working as I expected it to work.
>
>
>
Thanks. The module is now finished - and also tested. I want to generate
some tests and make sure
it holds up, but in general - it's working as I expected it to work.
On Mon, Oct 16, 2017 at 4:06 PM Joshua Colp <jc...@digium.com> wrote:
> On Mon, Oct 16, 2017, at 10:03 AM, Nir Si
16, 2017 at 1:16 PM Joshua Colp <jc...@digium.com> wrote:
> On Mon, Oct 16, 2017, at 06:45 AM, Nir Simionovich wrote:
> > Ok, that helped - looks like I'm linking correctly now.
> >
> > Different question, I remember their used to be a "safe string copy"
> >
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-dev mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-dev
>
--
Kind Regards,
Nir Sim
Hi All,
I'm in the process of adding a new module to Asterisk, in this case, a
new CDR backend.
The new backend relies on a library that I need to introduce to the linker,
however, I've tried
to figure out how the autotools work in there - and had failed miserably.
I would appreciate if
line so it is more specific than
> "Re: Contents of asterisk-dev digest..."
>
>
> Today's Topics:
>
>1. Deadlock in chan_sip, caused by weird media re-invite from
> remote side (Nir Simionovich)
>
>
> ------
Hi All,
We have several systems, some running Asterisk 13 some 12. We have
recently discovered a
possible dead-lock scenario in chan_sip. The dead-lock seems to manifest as
the below:
LCR-AMS-01*CLI> core show locks
===
===
Cool, too bad it isn't documented. I'll add it into PHPARI as well.
On Mar 8, 2015 6:18 PM, Matthew Jordan mjor...@digium.com wrote:
On Sun, Mar 8, 2015 at 10:51 AM, Nir Simionovich
nir.simionov...@gmail.com wrote:
Ok, I'll have a look into that one.
On Sun, Mar 8, 2015 at 1:03 PM, Olle E
Hi All,
So, I've been banging my head against an issue with ARI. While Channel
Originate enables
you to originate channels, you can't really do a SIPAddHeader type
functionality in there.
Originally, I was under impression that endpoints/message should be able
to give me the functionality I
Ok, I'll have a look into that one.
On Sun, Mar 8, 2015 at 1:03 PM, Olle E. Johansson o...@edvina.net wrote:
On 08 Mar 2015, at 09:52, Nir Simionovich nir.simionov...@gmail.com
wrote:
Hi All,
So, I've been banging my head against an issue with ARI. While Channel
Originate enables
Hi All,
I've managed to re-implement the basic functionality of app_dial using
ARI and PHPARI.
I've tested it and it supports handling of multiple calls at the same time.
Having said that,
I would highly appreciate some feedback in regards to the methodology, or
if anybody
can see something I
Hi all,
This is somewhat of an off topic discussion, however, I'm putting it here
- as most of your have more experience than me when it comes to using git.
So, we've been using GitHub for a year now as our Git repository and are
fairly happy with it. At the same time, we're using BitBucket
Hi All,
I'm not sure if the dev list is the proper list of this, however, due to
the fact that the issue at hand
revolves around documentation and proper usage, I think bringing it up here
is a good place.
So, during the past few days, I've been trying to implement the Dial
application using
+1 from me as well.
We use the methodology of using personalized repos for projects and it
works really well. We use either GitHub
or BitBucket, depending on the project - but both work equally well.
I'm confident that Atlassian will be happy to show their support by
contributing a Stash license
- 178.62.127.227:44972 [AP]
{
message: Application or extension must be specified
}
On Thu, Dec 18, 2014 at 9:48 AM, Nir Simionovich nir.simionov...@gmail.com
wrote:
One more thing - per your recommendations, I'm trying to re-implement
app_dial using ARI.
Now, if I read you all right, the process
In deed, that is interesting - but truly stirs away from ARI - not
something that I'm trying to do.
On Thu, Dec 18, 2014 at 1:00 PM, Kaloyan Kovachev kkovac...@varna.net
wrote:
Hi,
On 2014-12-18 01:01, Nir Simionovich wrote:
Let's try to stick to the tech for now and answer the first two
Ahh...
On Thu, Dec 18, 2014 at 1:32 PM, Kaloyan Kovachev kkovac...@varna.net
wrote:
I meant that for accessing bridge configuration for a Legacy (Dial,
Queue, FollowMe) from your other email
On 2014-12-18 13:26, Nir Simionovich wrote:
In deed, that is interesting - but truly stirs
December 2014 at 11:07, Paul Belanger paul.belan...@polybeacon.com
wrote:
On Thu, Dec 18, 2014 at 1:59 AM, Nir Simionovich
nir.simionov...@gmail.com wrote:
New question: Do we want to enable legacy features inside ARI?
New answer: I don't believe so.
I think this issue / question
:
On Thu, Dec 18, 2014 at 1:59 AM, Nir Simionovich
nir.simionov...@gmail.com wrote:
New question: Do we want to enable legacy features inside ARI?
New answer: I don't believe so.
I think this issue / question is the hardest thing to understand about
ARI. There really isn't any sort
Hi All,
After shipping out my first patch to ARI, I became hungry :-)
So, now I've set up a slightly higher goal, adding a much required
feature for ARI. I'll describe the problem first, then
I have some questions.
The Asterisk dial application enables us to limit the duration of the
call
- if assigned
3. Provide a means via ARI to manipulate the duration timers
Nir
On Wed, Dec 17, 2014 at 10:37 PM, Paul Belanger
paul.belan...@polybeacon.com wrote:
On Wed, Dec 17, 2014 at 3:16 PM, Nir Simionovich
nir.simionov...@gmail.com wrote:
Hi All,
After shipping out my first patch
17, 2014 at 3:46 PM, Nir Simionovich
nir.simionov...@gmail.com wrote:
Well,
In simple words yes. To be more specific, I'd like to do something
like
this:
1. Have a simple dialplan that will dialout using the L parameter in
Dial
application
2. Have ARI bridge list function
to the dialplan is literally three lines. The
minimum required. I never return.
Phil M
On Wed, Dec 17, 2014 at 4:12 PM, Nir Simionovich
nir.simionov...@gmail.com wrote:
Ok, I'll start with this - I agree with the both of you, ARI is the right
way to go.
However, when I look at ARI, I see
also no reason not to explore new methods, is there?
Phil M
On Wed, Dec 17, 2014 at 5:29 PM, Nir Simionovich
nir.simionov...@gmail.com wrote:
Phil,
It is one thing to say: I'm interested in advancement, it is
completely a different thing to say: I don't give a damn about backward
...@digium.com
wrote:
On 12/17/2014 05:01 PM, Nir Simionovich wrote:
snip
Let's try to stick to the tech for now and answer the first two
questions:
1. Is there a way to obtain the information in ast_bridge_config for
each of the iterated bridges?
ast_bridge_config is not used at all
?
On Thu, Dec 18, 2014 at 8:59 AM, Nir Simionovich nir.simionov...@gmail.com
wrote:
I see your point now - that makes more sense. It was fairly clear to me
that ast_bridge_config is somewhat of a legacy data structure,
but I was assuming that in some respect it was used in ARI as well. What
your
Hi All,
So, if there is one thing I really like about PJSIP and WebRTC
(specifically with mobile) is the ability to produce meaningful MoS scoring
for calls in real time. Now, Asterisk doesn't have that capability, at
least, not during the actual call - but only after.
In itself, not an issue -
I'm in.
On Oct 17, 2014 10:08 PM, Leif Madsen lmad...@thinkingphones.com wrote:
Your encouragement is noted and discarded. See you all at AstriCon! :)
On 17 October 2014 16:00, Billy Chia bc...@digium.com wrote:
There will actually be an opportunity to grab a beer at the Hackathon
Reception
the
ast_ari_channels_continue_in_dialplan function, but had failed to do so.
Any pointers?
Nir S
On Mon, Oct 13, 2014 at 1:52 AM, Nir Simionovich nir.simionov...@gmail.com
wrote:
Ok,
I've opened an issue on JIRA (
https://issues.asterisk.org/jira/browse/ASTERISK-24412) with a small
patch submission
that will correct
with the implementation - and that I've done it right.
It's the first time I'm touching that side of the code, so I would
appreciate the assistance and feedback.
Cheers,
Nir S
On Sun, Oct 12, 2014 at 12:46 AM, Nir Simionovich nir.simionov...@gmail.com
wrote:
So, here's what I thought - instead of modifying
, Scott Griepentrog sgriepent...@digium.com
wrote:
Yes, that's the function that converts a label to a priority. You should
be able to use that to enable label lookup from the rest api.
On Fri, Oct 10, 2014 at 1:38 PM, Nir Simionovich
nir.simionov...@gmail.com wrote:
Well,
Can I assume you
to the method implementation in res/ari/resource_channels.c.
On Thu, Oct 9, 2014 at 4:56 AM, Nir Simionovich nir.simionov...@gmail.com
wrote:
Forgive me father for I have sinned, it has been over 25 years since
I've used GWBasic/Basica - please spare thy humble servant from doom, as I
repent my sins
.
For example:
exten = _x.,1,Answer()
same = n,GoToIf($[${GOTTAGONOW} = 1]?louie)
same = n,Playback(tt-monkeys)
same = n,Hangup()
same = n(louie),Playback(lyrics-louie-louie)
same = n,Hangup()
exten = _x.,10001,GoTo(louie)
On Wed, Oct 8, 2014 at 8:33 AM, Nir Simionovich nir.simionov
Hi Guys,
While working on PHPARI, I've come to a realization that the channels
REST API
has a slight issue - primarily, its usage of the priority member in the
REST API.
Currently, the specification states that priority is either int or
long (depending
on the request).
The problem with
Title: RE: [asterisk-dev] Feature Bounty - Pitch Control for MeetMe
Hi John,
Interesting application, although, it's not stable yet and not ready for production
use. However, I'll try using it.
Nir Simionovich
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL
://www.lobstertech.com/code/voicechanger/
John
On Fri, 2006-08-11 at 13:58 +0300, Nir Simionovich wrote:
Hi All,
We require a new feature for app_meetme to be added, a pitch control.
The idea is to enable pitch changes of the talking participent by pressing
the * or # buttons (activating this feature
can't, I have no control over the extensions. I basically interconnect via a
PRI to an external Avaya CTI system, thus, I have no way of implementing queues
in the system - due to constraints by the Avaya CTI system.
Regards,
Nir Simionovich
indicates that something else is
wrong.
So, anyone has an idea of what's going on here? Or better yet, a proposed
course of Action?
Regards,
Nir Simionovich
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Well,
I would and use something external, like the mon utility from the linux-ha
project. Monitoring if asterisk is alive, from the internal has its
advantages, however, monitoring as a concept is always performed from a 3rd
party process or server.
Nir S
-Original Message-
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