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 va
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
co
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
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 (although reasonable) test
equests. 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 implementation as a base for l
t using REDIS wherein
>>>>> the client is made using a socket library and no other third party client
>>>>> library in C .
>>>>>
>>>>> This REDIS database has 400 million records and performs extremely
>>>>> well though the mem
quirement 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
> wrote:
>
> Hi All,
>
> Following a disc
pear this - would love 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.c
well, that was exactly my feeling.
On Mon, Nov 27, 2017 at 3:10 PM Joshua Colp 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
> > something in
e I got it correct?
On Thu, Nov 23, 2017 at 2:03 AM Nir Simionovich
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
> Redis, other may prefer b
redis" 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.
>
it's on gerrit - it's inside asterisk-team
On Thu, Nov 23, 2017 at 1:11 AM Corey Farrell 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 st
hly appreciate 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...@greenfield
us, seems fairly
alien to the configuration 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.app
.X.
From a naming 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 wrote:
>
> On 16 Nov 2017, at 22:18, Nir Simionovich
> wrote:
>
> and that the RPM repo for Kam
Iqbal
>
> ICTBroadcast - an Auto Dialer software for ITSP
> <https://www.ictbroadcast.com/how-become-internet-telephony-service-provider-itsp-using-ictbroadcast-sp-edition>
> SMS, Fax and Voice broadcasting & Inbound / Outbound Campaigns
> http://www.ictbroadcast.com/
>
ed traud's 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 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
wrote:
&g
No problem guys - I'll create respective repos on github later today.
On Fri, Nov 17, 2017 at 4:40 PM Jared Smith
wrote:
> On Thu, Nov 16, 2017 at 4:18 PM, Nir Simionovich <
> nir.simionov...@gmail.com> wrote:
>
>> As part of our work, we've noticed that Aste
nd Kamailio 5.X 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)
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 wrote:
> On Sat, Nov 4, 2017 at 2:53 PM, Nir Simionovich > wrote:
>
>>
join the project.
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
ecially since I
> 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:
>
roper "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 wrote:
> On Mon, Oct 30, 2017 a
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 10/29/2017 10:17 A
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
wrote:
> Correction, seems like this requires a bit more architecture than I
> anticipated.
>
> Basically, we need to separate this into several files and
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
wrote:
> I'll have a
Can be augmented with something like the following:
DEFINE_REDIS_STATEMENT(put_redis_stmt, "");
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.
PM Olle E. Johansson wrote:
> On 26 Oct 2017, at 15:20, Nir Simionovich
> 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 from what I can tell, and tell me if
ening 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
> wrote:
>
> I'd like to +1 on that idea.
>
> While I'm somewhat reluctant to using mySQL as the base
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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> _
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>
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> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-dev
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Ki
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
wrote:
> Hi All,
>
> So, my new cdr_beanstalkd module is available at
>
hand on this one :-)
On Mon, Oct 16, 2017 at 5:02 PM Nir Simionovich
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.
>
>
>
> On Mon,
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 wrote:
> On Mon, Oct 16, 2017, at 10:03 AM, Nir Simionovich wrote:
> &g
, Oct 16, 2017 at 1:16 PM Joshua Colp 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"
> > function that
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>
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Kind Regards,
Nir Simionovich
GreenfieldTech
(
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 someo
Tried initially with 13.2 - was exactly the same. I'll try latest 13 stable
and see if it re-creates.
On Tue, Apr 5, 2016 at 3:39 PM, Joshua Colp wrote:
> Nir Simionovich wrote:
>
>
>
>
>> Soft Phone -> Asterisk A -> Asterisk B -> Carrier
>>
>>
ubject 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
===
=== 12
Hey Matt,
One of the subjects I normally tackle first, when building a federated
system is the data store.
Normally, I would use something like Redis and utilize its PUB/SUB bus to
propagate information
across nodes.
Having a similar mechanism with Asterisk seems like a good idea, would
surel
Cool, too bad it isn't documented. I'll add it into PHPARI as well.
On Mar 8, 2015 6:18 PM, "Matthew Jordan" 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
Ok, I'll have a look into that one.
On Sun, Mar 8, 2015 at 1:03 PM, Olle E. Johansson wrote:
>
> On 08 Mar 2015, at 09:52, Nir Simionovich
> wrote:
>
> > Hi All,
> >
> > So, I've been banging my head against an issue with ARI. While Channel
> Orig
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
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 ca
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 f
+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
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
wrote:
> >>
> >> On Thu, Dec 18, 2014 at 1:59 AM, Nir Simionovich
> >> wrote:
> >> > New question: Do we want to enable legacy features inside ARI?
> >> >
> >> New answer: I don't believe so.
> >>
> >> I think t
11:07, Paul Belanger
> wrote:
>
>> On Thu, Dec 18, 2014 at 1:59 AM, Nir Simionovich
>> wrote:
>> > New question: Do we want to enable legacy features inside ARI?
>> >
>> New answer: I don't believe so.
>>
>> I think this issue / questi
Ahh...
On Thu, Dec 18, 2014 at 1:32 PM, Kaloyan Kovachev
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
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
wrote:
>
> Hi,
>
> On 2014-12-18 01:01, Nir Simionovich wrote:
>
> Let's try to stick to the tech for now and an
.
T 178.62.127.227:8088 -> 178.62.127.227:44972 [AP]
{
"message": "Application or extension must be specified"
}
On Thu, Dec 18, 2014 at 9:48 AM, Nir Simionovich
wrote:
>
> One more thing - per your recommendations, I'm trying to re-implement
> app_dial
quot;message": "Application or extension must be specified"
}
Am I missing something here? or is this a bug?
On Thu, Dec 18, 2014 at 8:59 AM, Nir Simionovich
wrote:
>
> I see your point now - that makes more sense. It was fairly clear to me
> that ast_bridge_config is
1:26 AM, Mark Michelson
wrote:
>
> On 12/17/2014 05:01 PM, Nir Simionovich wrote:
>
>
>
> 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
All I'd like is another way. The
> current methods don't have to disappear. It's not either or. However,
> there's 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
ll change?
>
> In my case, the connection 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
Dec 17, 2014 at 3:58 PM, Paul Belanger <
> paul.belan...@polybeacon.com> wrote:
>>
>> On Wed, Dec 17, 2014 at 3:46 PM, Nir Simionovich
>> wrote:
>> > Well,
>> >
>> > In simple words yes. To be more specific, I'd like to do something
>
- 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
> wrote:
> > Hi All,
> >
> > After shippi
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
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
I'm in.
On Oct 17, 2014 10:08 PM, "Leif Madsen" wrote:
> Your encouragement is noted and discarded. See you all at AstriCon! :)
>
> On 17 October 2014 16:00, Billy Chia wrote:
>
>> There will actually be an opportunity to grab a beer at the Hackathon
>> Reception in the same room as AstriDevCon
formation from within 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
wrote:
> Ok,
>
> I've opened an issue on JIRA (
> https://issues.asterisk.org/jira/browse/ASTERISK-24412) with a small
> patch sub
f base
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 wrote:
> So, here's what I thought - instead of m
s_handle_originate_with_id.
Makes the entire thing kind'a tricky - no?
On Fri, Oct 10, 2014 at 9:48 PM, Scott Griepentrog
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.
>
>
the new/different parameters to the rest
> methods, do a make ari-stubs to rebuild the rest handlers, and finally make
> the change to the method implementation in res/ari/resource_channels.c.
>
> On Thu, Oct 9, 2014 at 4:56 AM, Nir Simionovich > wrote:
>
>> "Forgive me
onkeys)
>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 > wrote:
>
>> Hi Guys,
>>
>> While working on
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
me.
http://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 #
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] [mai
", my answer would be: "I
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
hat the PRI
is causing issues, however, scenario 4 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
___
--Bandwidth and Colocatio
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-
From: [EMAI
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