having to specify said 'endpoint' when
dialing an arbitrary SIP URI, which is very nice.
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tried stopping ICE and starting it using
ast_rtp_ice_stop/ast_rtp_ice_start?
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bump the version, but this is one of those
things that can only occur if you are using a feature provided by a
loadable module and are a blithering idiot.
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of someone trying to figure out
why a test failed, it would be a bit easier to look in those locations (as
you usually have to anyway to look at the Asterisk logs), and it doesn't
preclude using the logger.conf for the Python logger to create a central
location for log files.
Matt
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the same - a test failure would copy the
failing test's files over to logs/.
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/listinfo/svn-commits
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The results are fairly messy :(
The patch I've suggested is as simple as I could make it without just
rolling back the patch.
Hey Steve -
I replied on the issue, but your analysis looks correct. Oh what fun CDRs
are...
Matt
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on the issue; I'll comment on some of the specifics you noted as
well.
Again - nice job!
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in Asterisk 10.
So, to everyone who helped make Asterisk 10 successful, thank you!
Matt
[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
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, but it would be
more stuff to browse through.
Matt
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files as
well.
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* Checksum results for both tarballs and patch files in a single file
Matt
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);
for (i = objset; i; i = i-next) {
RAII_VAR(char *, camel, ast_to_camel_case(i-name), ast_free);
ast_str_append(buf, 0, %s: %s\r\n, camel, i-value);
}
return 0;
}
Hope that helps!
Matt
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On Fri, Jan 3, 2014 at 9:57 AM, Olle E. Johansson o...@edvina.net wrote:
On 03 Jan 2014, at 16:50, Matthew Jordan mjor...@digium.com wrote:
In sip.conf, skinny.conf and other places, it's setvar without underscore.
Why change the syntax already used?
Hey Olle -
The syntax here
for it later today.
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configuration. That will allow DTMF to pass
through the conference - it's possible that your code is looking at
the frames after they have passed through the softmix bridging
technology, in which case, they would have already been consumed in
the bridging core.
Matt
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/uuid.h], [])
AC_CHECK_FUNCS([uuid_generate_random], [SYSUUID=true], [SYSUUID=])
If you install the uuid-devel library, does that resolve the dependency for you?
If so, I'll get the install_prereq script updated.
Matt
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project - and the Asterisk
community!
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[2] https://wiki.asterisk.org/wiki/display/AST/Reviewboard+Usage
[3] https://wiki.asterisk.org/wiki/display/AST/Writing+a+Python+Test
[4] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Test+Suite+Documentation
Matt
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://svn.asterisk.org/svn/asterisk/team/group/media_formats/
Cheers,
Somewhat off topic, but is there a silk codec for 12?
Not yet, but shortly.
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[2] https://code.asterisk.org/code/changelog/asterisk?cs=401884
[3] http://svn.asterisk.org/svn/asterisk/tags/11.7.0/asterisk-11.7.0-summary.txt
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the issue.
It may be necessary for something either in res_rtp_asterisk or
res_fax_spandsp to verify that the number of samples in the RTP packet
(or what the voice frame has in it before it gets handed off) matches
what is expected.
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On Wed, Jan 29, 2014 at 5:44 PM, Steve Davies davies...@gmail.com wrote:
Hi,
Thanks for looking at this.
On 29 January 2014 16:34, Matthew Jordan mjor...@digium.com wrote:
[snip]
You'll note that the refcall is only called if ast_sched_del
(eventually) returns a valid ID that it deleted
probably be best implemented as an external application built on top
of ARI.
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identifier module.
Matt
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as the
local_candidates container.
Matt
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On Tue, Feb 11, 2014 at 11:55 AM, Daniel Pocock dan...@pocock.com.au wrote:
On 11/02/14 03:13, Matthew Jordan wrote:
On Mon, Feb 10, 2014 at 1:25 PM, Olle E. Johansson o...@edvina.net wrote:
On 10 Feb 2014, at 20:16, Daniel Pocock dan...@pocock.com.au wrote:
I'm looking at the way
this is by design, in which case it needs to be documented, or it's an
oversight,
in which case I'd be happy to fix it.
I'd go with oversight.
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.
If chan_sip can't find that, it falls back to using a general entry
point.
Also: please don't stay on Asterisk 10. That version is no longer
supported and is no longer receiving security fixes. You should move
to Asterisk 11, which is an LTS release, as soon as possible.
Matt
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On Thu, Feb 13, 2014 at 7:55 AM, Guillaume Maudoux
guillaume.maud...@escaux.com wrote:
Hello everyone.
We have been struggling for one week on a bug with H264 video in asterisk11.
I do not know if it also applies to Asterisk 12.
Apparently res_format_attr_h264.c does not write properly the
learning - and we all have to start learning
somewhere.
Thanks!
Matt
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!
Matt
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the intent of the configuration, when
in reality, the configuration is probably in error. I like your
proposed solution better as well: tell the user that these options are
mutually exclusive and have them correct it.
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for the Asterisk project, and look
forward to more of his contributions to the project and the new PJSIP
stack.
Welcome George Joseph!
(And we're sorry you have to use subversion to commit things. Git...
some day, some day.)
Matt
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used on a channel to a different set of formats, they can then
issue an AMI action Set to change the formats and issue a new
re-INVITE/UPDATE request.
Matt
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, that there would be any
issue for Asterisk 11+ with segmenting chan_dahdi into a separate
subpackage. I think that would be a little tough for Asterisk 1.8,
however, as MeetMe still requires DAHDI for mixing.
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://lists.digium.com/pipermail/asterisk-dev/2012-November/057591.html
[2] http://trac.pjsip.org/repos/milestone/release-2.2
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On Thu, Feb 27, 2014 at 3:13 PM, Jared Smith jaredsm...@jaredsmith.net wrote:
On Thu, Feb 27, 2014 at 12:26 PM, Matthew Jordan mjor...@digium.com wrote:
If the packages were restructured, it could be set up so that Asterisk
only provides chan_dahdi in a subpackage - although
] https://wiki.asterisk.org/wiki/x/QoCoAQ
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.
That way all our documentation / functionality is consistent among
channel drivers.
Yeah... that will never happen.
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On Thu, Mar 6, 2014 at 3:42 PM, Damien Wedhorn v...@facts.com.au wrote:
On 07/03/14 07:29, Matthew Jordan wrote:
On Thu, Mar 6, 2014 at 3:22 PM, Paul Belanger
paul.belan...@polybeacon.com wrote:
On Thu, Mar 6, 2014 at 3:31 PM, George Joseph
george.jos...@fairview5.com wrote:
For me
On Thu, Mar 6, 2014 at 4:32 PM, Damien Wedhorn v...@facts.com.au wrote:
On 07/03/14 08:21, Matthew Jordan wrote:
On Thu, Mar 6, 2014 at 3:42 PM, Damien Wedhorn v...@facts.com.au wrote:
On 07/03/14 07:29, Matthew Jordan wrote:
On Thu, Mar 6, 2014 at 3:22 PM, Paul Belanger
paul.belan
On Fri, Mar 7, 2014 at 8:19 AM, Paul Belanger
paul.belan...@polybeacon.com wrote:
On Thu, Mar 6, 2014 at 5:32 PM, Damien Wedhorn v...@facts.com.au wrote:
On 07/03/14 08:21, Matthew Jordan wrote:
Thanks Matt
A couple of observations. While I agree with your general advice
an issue for this, but I'd
suggest gathering all of the standard logs illustrating the problem
before doing so.
Matt
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. I'm sure having someone test out the branch would help get
this issue moving forward.
The next step would be for someone to put the change up on Review Board.
[1] http://lists.digium.com/pipermail/asterisk-dev/2013-September/062326.html
Matt
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/wiki/display/AST/Asterisk+11+Application_VoiceMail
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will no longer occur (such as
the optimization begin/end events), and new ones will definitely start to
happen (BridgeEnter/Leave messages for the new native bridge). That may be
worth doing - but this isn't a habit we should get into.
Matt
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On Mon, Mar 10, 2014 at 6:59 AM, Joshua Colp jc...@digium.com wrote:
Matthew Jordan wrote:
snip
It's important to point out that optimization's goal was never the
removal of the channel. If anything, nuking Local channels has - in
my opinion - always made life more difficult
On Mon, Mar 10, 2014 at 7:27 AM, Matthew Jordan mjor...@digium.com wrote:
On Mon, Mar 10, 2014 at 6:59 AM, Joshua Colp jc...@digium.com wrote:
Matthew Jordan wrote:
snip
The NLB compatibility code actually checks whether something like a
MixMonitor is on either Local channel and won't
- but not to the overall detriment of the project. Adhering
to the letter of the law while ignoring the spirit is not something
I'll ever advocate.
[1] https://wiki.asterisk.org/wiki/display/AST/Coding+Guidelines
[2] https://wiki.asterisk.org/wiki/display/AST/Code+Review+Checklist
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On Fri, Mar 14, 2014 at 12:49 PM, Joshua Colp jc...@digium.com wrote:
Matthew Jordan wrote:
snip
My problem is when I get arguments like it's there in PJIP so we
have to use it or we can't do anything because of PJSIP.
That's not my argument at all.
My argument is thus:
* PJSIP
/g711_8h_ca377b1f9c4a8b8f3211e5cfad9954ab.html
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On Mon, Mar 17, 2014 at 7:57 AM, Johann Steinwendtner
steinwendt...@gmx.net wrote:
On 2014-03-17 13:27, Matthew Jordan wrote:
On Mon, Mar 17, 2014 at 6:25 AM, Johann Steinwendtner
steinwendt...@gmx.net wrote:
On 2014-03-06 02:58, SVN commits to the Asterisk project wrote:
Hello
/065992.html
[5] http://lists.digium.com/pipermail/asterisk-dev/2014-March/066115.html
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for what will become
12.2.0.
In retrospect, we should have probably kept each module's realtime
database tables separate. That would have minimized these kinds of
changes.
How terrible would it be to nuke the existing scripts and separate
them all out? :-)
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On Mon, Mar 17, 2014 at 11:45 AM, Joshua Colp jc...@digium.com wrote:
Matthew Jordan wrote:
On Mon, Mar 17, 2014 at 11:02 AM, Joshua Colpjc...@digium.com wrote:
Greetings all,
Awhile ago when I was working on PJSIP DNS Matt brought up adding the
option
to Alembic. Through doing so I
On Tue, Mar 18, 2014 at 8:31 AM, Russell Bryant
russ...@russellbryant.net wrote:
On Mon, Mar 17, 2014 at 1:01 PM, Joshua Colp jc...@digium.com wrote:
Matthew Jordan wrote:
snip
Technically, it's probably not hard. I wonder how horrible it'd be for
end users, however.
Probably pretty
do you all think?
Matt
[1]
https://confluence.atlassian.com/display/AOD/Processing+JIRA+issues+with+commit+messages
[2] https://code.asterisk.org/code/
[3] http://lists.digium.com/pipermail/asterisk-commits/2014-March/067893.html
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instance, as opposed to
something forced.
In a world where all Asterisk instances know of the device states for
any other Asterisk instance, can you explain why publishing extension
states are needed?
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On Wed, Mar 19, 2014 at 10:00 AM, Olle E. Johansson o...@edvina.net wrote:
On 19 Mar 2014, at 15:55, Olle E. Johansson o...@edvina.net wrote:
On 19 Mar 2014, at 15:41, Matthew Jordan mjor...@digium.com wrote:
On Wed, Mar 19, 2014 at 9:26 AM, Olle E. Johansson o...@edvina.net wrote:
snip
-
Matt
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that someone doesn't like, we can
always discuss it on the -dev list :-)
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if there were, but I imagine the extraction process
out of pbx_dundi would be a non-trivial amount of work.
Matt
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on
contributing patches are available on the Asterisk wiki here:
https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process
It'd be great to get this fixed in the next point release of Asterisk 12.
Thanks!
Matt
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the conference. This is really all that is needed for this
problem, and the structure of ConfBridge lends itself better to this
approach.
Matt
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- sorry about that!
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On Tue, Apr 29, 2014 at 6:15 AM, Joshua Colp jc...@digium.com wrote:
Matthew Jordan wrote:
(1) Where will the packetization of a format on a channel reside? With
the format on the channel? Or with the capability of the channel?
It has to reside in the RTP engine layer and optionally can
On Tue, Apr 29, 2014 at 6:49 AM, Matthew Jordan mjor...@digium.com wrote:
On Tue, Apr 29, 2014 at 6:15 AM, Joshua Colp jc...@digium.com wrote:
snip
The ability to set the framing per-format, globally in a capabilities
structure, and to get the framing already exists. What doesn't exist
On Tue, Apr 29, 2014 at 6:55 AM, Joshua Colp jc...@digium.com wrote:
Matthew Jordan wrote:
snip
snip
The ability to set the framing per-format, globally in a
capabilities structure, and to get the framing already exists. What
doesn't exist is adding a format
is never used. As the object is
immutable, a public accessor for an attribute key could be used that
returns a const char *, which would fulfil both functions.
Thoughts? Flames?
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Check us out
On Wed, Apr 30, 2014 at 5:59 AM, Joshua Colp jc...@digium.com wrote:
Matthew Jordan wrote:
snip
:: Setting Attributes ::
The biggest sticky point is clearly #1. I don't think we should lose the
immutability convention - that feels bad, since there's a lot of benefit
to treating
On Wed, Apr 30, 2014 at 8:00 AM, Joshua Colp jc...@digium.com wrote:
Matthew Jordan wrote:
On Wed, Apr 30, 2014 at 5:59 AM, Joshua Colp jc...@digium.com
mailto:jc...@digium.com wrote:
Matthew Jordan wrote:
snip
If we allow attributes to be set after format creation, we have two
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://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
[2] https://wiki.asterisk.org/wiki/display/AST/Code+Review
[3] https://wiki.asterisk.org/wiki/display/AST/Coding+Guidelines
Thanks!
Matt
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information on submitting a patch to Asterisk
on the Asterisk wiki [3].
[1] https://issues.asterisk.org
[2] https://wiki.asterisk.org/wiki/display/AST/Digium+License+Agreement
[3] https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process
Thanks!
Matt
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}
It's possible to not get events with this bug, as bridge_softmix never
detects the change in energy.
Can you test with 11.10.0-rc1/12.3.0-rc1 (now available in SVN, and in
an hour or so, tarballs) or the latest from the 11/12 branch?
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sense.
I'm curious, what happens if you just remove the checks in res_rtp_asterisk?
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- is not looking to address any
performance problems so much as DTMF emulation extending digits when it
shouldn't.
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?
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; they should just be compatible (based on their
respective types).
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On Tue, Mar 18, 2014 at 1:25 PM, Matthew Jordan mjor...@digium.com wrote:
Hey everyone -
Due to a security vulnerability in JIRA, we recently had to upgrade
JIRA to version 6.2. While this was a good thing (tm), it did break
the snot out of the Subversion plugin. That plugin does a number
On Mon, Jun 2, 2014 at 4:14 PM, James Cloos cl...@jhcloos.com wrote:
MJ == Matthew Jordan mjor...@digium.com writes:
MJ That is incorrect. The sip_sendhtml callback will update the url
MJ stringfield on the SIP pvt. It then transmits a re-INVITE via
MJ transmit_reinvite_with_sdp
On Thu, May 29, 2014 at 3:31 PM, Gunnar Hellstrom
gunnar.hellst...@omnitor.se wrote:
On 2014-05-28 17:46, Matthew Jordan wrote:
On Tue, May 27, 2014 at 4:59 PM, Gunnar Hellstrom
gunnar.hellst...@omnitor.se wrote:
snip
It works fine for calls with only real-time text ( red + t140 ).
Still
, not to solicit
help for deployment situations. Please use the asterisk-users mailing
list for questions of this type - you will be far more likely to
receive useful responses.
(And, 99% of the developers are subscribed to and read that mailing
list as well)
Matt
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libraries to the versions mentioned above.
When I just use the python websocket-client library, I can connect to the
websocket of ari just fine. It just doesn't work with the library.
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or debugging
mode. The code change to do this is trivial but I thought I'd run it by you
guys first.
Thoughts?
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.
There are some other users abusing it as well. (Apparently unit tests and
frame formats?)
Works for me.
On Mon, Jun 9, 2014 at 12:46 PM, Matthew Jordan mjor...@digium.com
wrote:
On Sun, Jun 8, 2014 at 10:03 AM, George Joseph
george.jos...@fairview5.com wrote:
Right now, the non-ref
for the confusion -
Matt
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the
community in the development, review, and testing of security patches?
Matt
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On Fri, Jun 13, 2014 at 2:07 AM, Timo Teras timo.te...@iki.fi wrote:
On Fri, 13 Jun 2014 01:57:25 -0500
Matthew Jordan mjor...@digium.com wrote:
On Fri, Jun 13, 2014 at 1:50 AM, Timo Teras timo.te...@iki.fi wrote:
On 13 Jun 2014 01:39 -0500
Asterisk Development Team asteriskt
On Fri, Jun 13, 2014 at 4:41 AM, Steven Howes steve-li...@geekinter.net
wrote:
On 13 Jun 2014, at 08:12, Matthew Jordan mjor...@digium.com wrote:
Apologies if this e-mail gets a bit rambling; by the time I send this it
will be past 2 AM here in the US and we've been scrambling to fix
with Asterisk. Any pointers on
how to set up the SIP address shall be appreciated.
Regards,
Sujatha
The asterisk-dev list is used for discussions regarding development of the
project. Please use the asterisk-users mailing list for assistance in
configuring and setting up Asterisk.
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Matthew Jordan
has commit access would like access to this group,
please let me know.
Thanks -
Matt
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Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com http://asterisk.org
* ast_codec_samples_count - move to frame.h. Rename to
ast_frame_get_codec_samples.
[1] https://issues.asterisk.org/jira/browse/ASTERISK-23715
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com http://asterisk.org
to be.
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com http://asterisk.org
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if the function 'survived', usage of it is now different. If you
don't dispose of the reference returned to you, you'll now leak the
format object.
Breaking changes should be obvious. I'll replace functions where appropriate.
On Sat, Jun 21, 2014 at 6:10 PM, Matthew Jordan mjor...@digium.com wrote
in the queue does get
announcements. The aforementioned behaviour is a possibility, however
- so expectations should be set for your agents when this feature is
enabled.
Matt
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Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http
and when someone is actively working it.
Soliciting help on the mailing list without an issue is not a good idea.
The issue tracker helps to coordinate efforts by various bug marshals, and
helps to ensure that your issue is not lost and gets a response.
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Matthew Jordan
Digium, Inc. | Engineering
it in r417419
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com http://asterisk.org
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