just wrappers
> around maven.
> atlas-compile and atlas-package will do it.
>
>
>
>
>
>> On Thu, Jun 10, 2021 at 10:37 AM George Joseph wrote:
>>
>>
>>> On Thu, Jun 10, 2021 at 10:00 AM BJ Weschke wrote:
>>> I’d be willing to take a
I’d be willing to take a look at it for you George.
Sent from my iPhone
> On Jun 10, 2021, at 11:31 AM, George Joseph wrote:
>
>
>
> You already know about the SSH host key issue related to the upgrade of
> Gerrit we did on May 28th.That issue we knew about in advance so we gave
>
Makes sense to me!
Sent from my iPhone
> On Nov 9, 2020, at 8:24 AM, Joshua C. Colp wrote:
>
>
>> On Tue, Oct 13, 2020 at 7:55 AM Joshua C. Colp wrote:
>
>> Hey all,
>>
>> I just wanted to drop an email and say that this hasn't been dropped or
>> anything. A 2 year option just isn't
now can work only with the older os etc
>
>> On Thu, Oct 1, 2020 at 10:55 PM Joshua C. Colp wrote:
>>> On Thu, Oct 1, 2020 at 4:31 PM BJ Weschke wrote:
>>
>>> Four years, is indeed, really long. I do agree with this. As an example, I
>>> work with an
I’d personally be OK with this. The more I think about this, two years is
really long enough. With your approach below, someone would have an entire LTS
release cycle to make any necessary integration changes that come as a result
of a module that is removed. If someone complains about
Four years, is indeed, really long. I do agree with this. As an example, I work
with another project where the work involves some integrations with software
that is in the head units of vehicles. Right now, they’re working to certify
and lock down code and functionality for the 2023 vehicle
I’m in favor of this approach as well.
On October 1, 2020 at 10:23:30 AM, Jared Smith (jaredsm...@jaredsmith.net)
wrote:
On Thu, Oct 1, 2020 at 10:04 AM Joshua C. Colp wrote:
Not really, and I think part of the problem is that this entire thing hasn't
really been documented, communicated,
I’ve been using it in several production systems for nearly a year now on the
16 branch and it has yet segfault. My remaining chan_sip Asterisk 13 systems
dump code at least once or twice every 3 months or so. I feel very safe saying
chan_pjsip is stable enough for my production needs.
Sent
be your best bet to figure out
what went wrong when the Asterisk instance crashed.
--
BJ Weschke
Sent with Airmail
On September 12, 2018 at 3:37:24 PM, i...@magnussolution.com
(i...@magnussolution.com) wrote:
that’s correct. I wrote a ael context with func_odbc and this work very well
AGI is limited in its TPS scalability because it needs to fork an external
application to complete processing. func_odbc run from within the dial plan
does not need to fork anything external so it does not have the same
scalability issues that present with an AGI based solution.
--
BJ Weschke
Why a native application? What data are you looking to store in the DB? It
seems like you could do what you’re looking to do in the dial plan with
func_odbc.
--
BJ Weschke
Sent with Airmail
On September 12, 2018 at 3:12:35 PM, i...@magnussolution.com
(i...@magnussolution.com) wrote:
hi
I would have an immediate use case for something like func_redis. I think it
would be very useful.
On December 22, 2017 at 8:02:28 AM, Nir Simionovich (nir.simionov...@gmail.com)
wrote:
Abhay,
Migrating astsb from SQLlite to redis isn't the topic here. I'm talking adding
a new type of
I think you’re referring to the asterisk-app-dev mailing list where ARI, AGI,
and AMI are all covered.
http://lists.digium.com/cgi-bin/mailman/listinfo/asterisk-app-dev
On January 16, 2017 at 3:40:53 PM, Phil Mickelson (p...@cbasoftware.com) wrote:
Re asterisk-ari list. I don't believe
Short answer: yes. However you'll probably want to take this to asterisk-users
for the particulars and different ways it could be implemented.
Sent from my iPhone
On Feb 6, 2015, at 10:26 AM, David Radcliffe
david.radcli...@clockworkit.co.uk wrote:
Hi,
Does anyone know if Asterix has
+1
This sounds more than reasonable to me.
Sent from my iPhone
On Nov 10, 2014, at 5:57 PM, Matthew Jordan mjor...@digium.com wrote:
At AstriDevCon, we spent a good amount of time discussing whether or
not we should allow new features or improvements to be made in release
branches. As I
On 11/6/14, 4:04 PM, Matthew Jordan wrote:
eg -
Playback(http://myserver.com/monkeys.wavhttp://myserver.com/can.wavhttp://myserver.com/act.wavhttp://myserver.com/like.wavhttp://myserver.com/weasels.wav)
--- On an empty HTTP Media cache, the previous app invocation would
probably sound pretty
Matt -
This is a pretty neat idea, indeed, but I've got some
questions/thoughts on implementation. :-) Apologies if all of this was
already considered/accounted for already..
1) Does the entire file need to be downloaded and in place on the HTTP
Media Cache before you can call an
On 11/4/14, 3:40 PM, Matthew Jordan wrote:
On Tue, Nov 4, 2014 at 12:57 PM, BJ Weschke bwesc...@btwtech.com wrote:
Matt -
This is a pretty neat idea, indeed, but I've got some questions/thoughts on
implementation. :-) Apologies if all of this was already
considered/accounted for already
On 10/22/14, 12:14 PM, Paul Albrecht wrote:
On Oct 22, 2014, at 10:33 AM, Joshua Colp jc...@digium.com wrote:
Paul Albrecht wrote:
Really? Shouldn’t something this major affecting the entire Asterisk
community get discussed on the lists? Any idea what Leif is talking
about when he says the
On 10/22/14, 3:06 PM, Paul Albrecht wrote:
On Oct 22, 2014, at 11:47 AM, BJ Weschke bwesc...@btwtech.com wrote:
On 10/22/14, 12:14 PM, Paul Albrecht wrote:
On Oct 22, 2014, at 10:33 AM, Joshua Colp jc...@digium.com wrote:
Paul Albrecht wrote:
Really? Shouldn’t something this major
Tilghman Lesher wrote:
On Tuesday 27 November 2007 11:26:34 Eliel Sardanons wrote:
On 11/27/07, Russell Bryant [EMAIL PROTECTED] wrote:
Eliel Sardanons wrote:
We could start a janitor for creating a 'foo reload' and we could make
de 'module reload *.so' do a module unload;
Olle E Johansson wrote:
20 nov 2007 kl. 22.58 skrev BJ Weschke:
Johansson Olle E wrote:
Friends,
Blitzrage and I had a discussion about busylevel and call-limit in
chan_sip on the IRC I wanted to expand to the rest of you deveopers
out there...
My proposal in this discussion
Olle E Johansson wrote:
15 nov 2007 kl. 13.22 skrev Tzafrir Cohen:
On Thu, Nov 15, 2007 at 12:43:40PM +0100, Johansson Olle E wrote:
While browsing the bug tracker today, I found a patch for adding more
concise commands to the SIP channel.
My personal opinion is that I don't like
On 3/29/07, Tilghman Lesher [EMAIL PROTECTED] wrote:
On Wednesday 28 March 2007 11:08, Vadim Lebedev wrote:
I hope it can be integrated in mainline
Why not just use the SetMusicOnHold application in the dialplan?
Because app_queue won't observe that. This patch is valid, but we
need to
On 1/3/07, Moises Silva [EMAIL PROTECTED] wrote:
Hi, I have been working with the patch for the Bridge manager action
and dial plan application. So far it seems to work, however, more
testing is needed, this bug/feature has more than 1 year on the
bugtracker, plz if someone has the time, test
On 11/2/06, Jared Smith [EMAIL PROTECTED] wrote:
On Thu, 2 Nov 2006, Stephen Davies wrote:
I posted up a change that gets chan_iax2 to log jitter buffer stats
into the logs regularly during active calls.
I've also responded to the bug (#8188), but I'll repost my comments below as
they're
On 9/13/06, Slim Shady [EMAIL PROTECTED] wrote:
Hello all,
I would like to know if anyone is interested in doing a quick and dirty
custom application module to be used similar to the monitor application
within features.conf. The application is for the medical industry and in my
oppinion I think
On 6/22/06, Thomas Kenyon [EMAIL PROTECTED] wrote:
In regards to the AddQueueMember, I simply forgot to add @context to
the end of the device name. I don't think that this should cause a
segfault. On the other hand, I wouldn't expect you to be testing for
Short of a dev conference call earlier this week to discuss, based on
JackEStorm's posts in #asterisk-bugs about his research into deadlock
issues with chan_agent/app_queue I've now also taken a harder look at
chan_agent.c this past week and I'm coming up with blanks at this
point trying to
On 4/17/06, Dov Bigio [EMAIL PROTECTED] wrote:
Hello, I made the following changes on my res_features.c to resolve an
instability I had with atxfer...
(Actually, I wasn't the one who did it cause I don't know C, but this
worked, so I am forwarding to you so that you can confirm it makes sense
On 4/11/06, Russell Bryant [EMAIL PROTECTED] wrote:
Hello everyone,
We've had various attempts to have ongoing developers' conference calls
in the past. Josh Colp, Kevin Fleming, and I were talking about this
again today and would like to propose that we start this up again.
The first
On 3/22/06, Wai Wu [EMAIL PROTECTED] wrote:
Hi all,
Has been poking through the * source code a bit and trying to identify the
most CPU demanding piece of code. Is trans-coding the most CPU demanding? I
happen to have access to a DSP pci board. If I can move the encoder/decoder
off the host
On 3/22/06, Matt Florell [EMAIL PROTECTED] wrote:
From: http://www.tmcnet.com/usubmit/2006/03/14/1456373.htm
Digium Inc., the creator of Asterisk(TM), and pioneer of open source
telephony, today announced the availability of new hardware solutions
to enhance Asterisk transcoding and echo
On 2/27/06, Andreas Sikkema [EMAIL PROTECTED] wrote:
The RTCP branch includes improved support of RTCP, but also a
reporting facility we do not use currently. Would it be
useful to add
this to a channel variable - or even better a CDR variable - so you
can add it to CDRs and make reports
On 2/17/06, Ed Greenberg [EMAIL PROTECTED] wrote:
Can somebody who understands chan_sip.c please explain this to me? THanks.
--On Thursday, February 16, 2006 6:20 AM -0800 Ed Greenberg
[EMAIL PROTECTED] wrote:
Back in Asterisk 1.0.5, when we sent our SDP packet to the distant end,
we sent
On 2/15/06, Tony Mountifield [EMAIL PROTECTED] wrote:
Has anyone done any work on enhancing the MeetMe keypad menu to allow
the initiation of an outgoing call which will be connected to the
conference? e.g. *5 followed by the number to call.
This would be very useful for adding participants
On 1/27/06, tim panton [EMAIL PROTECTED] wrote:
On 26 Jan 2006, at 12:22, Rich Adamson wrote:
However, in addition to the magic, anyone moving complete new code into
a high visibility production network without first testing it is nuts.
I agree with you, but in this case, we test 1.2
On 1/17/06, Phil Menico [EMAIL PROTECTED] wrote:
Paul,
Can you give us the details on this:
a .call file is sent to asterisk, which then calls you, detects pickup, and
then calls the remote party.
I am interested in making this work.
http://www.voip-info.org/wiki-Asterisk+auto-dial+out
On 1/13/06, Marc Haisenko [EMAIL PROTECTED] wrote:
Hi folks,
I spotted a bug in handle_request_info: in an if condition the code assumes
to receive NULL on error, while in fact it receives an empty string. The
attached trivial patch fixes this.
Patch is done against chan_sip.c from r8023.
= notice,warning,error,debug,verbose
but still extra detail is not logged into file!
On 12/28/05, BJ Weschke [EMAIL PROTECTED] wrote:
On 12/28/05, ast guy [EMAIL PROTECTED] wrote:
Hi!
Connecting to asterisk through command
# asterisk -r
( using ast_log fxn
On 12/28/05, Jason DiCioccio [EMAIL PROTECTED] wrote:
Greetings,
I was having a conversation with someone the other day and was informed
that ztdummy is basically unnecessary in BSD and perhaps in more recent
linux kernels. Is this indeed the case? Would you need to run asterisk
at a
On 12/20/05, Russell Bryant [EMAIL PROTECTED] wrote:
As we all already know, the n+101 priority jumping behavior of
applications is being deprecated.
For Asterisk 1.2, we made the default value of the global priority
jumping option to be on. However, if it was a new installation,
On 12/1/05, Charles Huang [EMAIL PROTECTED] wrote:
Hi, Alex:
After taking your suggestion change from em to fxoks, it still did not
work, and this time even calling to normal PSTN number also failed?
Any more suggestion?
Charles
Is this a dedicated LD trunk or a DID PRI trunk from
On 11/17/05, Michael Anderson [EMAIL PROTECTED] wrote:
I'm wanting to alter app_page so that I can specify an Alert Info sip
header to send (our Polycoms are set to auto answer on that one).
Eventually I would want to make it customizable, but for a first test
I thought I'd try the following:
On 11/16/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hi everybody.
I need develop a IAX softphone with Delphi, but i didnt find a OCX component.
Anyone know how can I
find this component ?
Tomas
I don't believe one exists as part of the standard distribution.
You're welcome to roll
Please post a bug for this in bugs.digium.com.
On 11/8/05, Patrick [EMAIL PROTECTED] wrote:
Hi all,
I hope this is not considered a -users question. If so please accept my
apologies for the noise. Compilation of cvs HEAD from about two hours
ago fails in res_config_odbc.c with the following
Welcome!
You can go here for some initial information about the Asterisk architecture:
http://www.digium.com/downloads/AstriconEurope2005Tutorial.pdf
And once you're ready, you can visit this link for information on how
to contribute:
http://www.asterisk.org/developers
On 11/8/05, Isack
You're right. The discrepancy does exist in the 1.0 tree. It was
fixed recently in CVS-HEAD and should certainly be in 1.2b2 and later.
On 11/5/05, Tilghman Lesher [EMAIL PROTECTED] wrote:
On Saturday 05 November 2005 12:45, Ronald Hartmann wrote:
Please accept my apology regarding
Are you using the name/record playback option?
On 10/18/05, Chih-Wei Huang [EMAIL PROTECTED] wrote:
BJ Weschke wrote: The bug was closed because the ztdummy behavior is not the specific cause
for the delay problem. That patch with USE_RTC was intended to make use of the real time resource
On 5/16/05, Olle E. Johansson [EMAIL PROTECTED] wrote:
BJ Weschke wrote:
Server A (IP 192.168.1.1)
Server B (IP(s) 192.168.1.2 [actual] 192.168.1.3 [vip])
Server C (IP(s) 192.168.1.4)
All servers are Asterisk installs. All servers have SIP canreinvite=yes.
Server A calls Server
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