ua C. Colp wrote:
On Thu, Oct 1, 2020 at 2:18 PM Corey Farrell <mailto:g...@cfware.com>> wrote:
Yes I'm suggesting documenting that it is deprecated in minor
releases. Ending support in a minor seems bad unless support
already doesn't exist. Could we add
AST_MODULE_SUPP
Any reason we can't "documentation deprecate" things in minor releases?
No runtime warnings and keep building by default but if we deprecate a
module in master right now it seems like the next minor release of all
active branches should document the status of the module. The fact that
a
With the current process new features can be added with tests can target
18 as soon as the branch is cut but would not go into 18.0. What would
happen with new features in this scheme, would they just get held open
in gerrit until 18.0 is branched or be accepted as normal since 18.0.0
release
chan_sip is fragile and has no active maintainers. Personally I think
new features do not belong in chan_sip, should instead be implemented
for chan_pjsip.
On 11/19/19 8:06 AM, bala murugan wrote:
Hi Ernst ,
I can develop this for you in chan_sip driver on the version you
are using
iksemel-devel: To my knowledge library is not maintained and has been
removed from Fedora because it no longer builds from source. Honestly I
thought we had deprecated res_xmpp and chan_motif for this very reason.
libedit-devel: This is mandatory since Asterisk 16. The devel package
is found
Updating pjproject does take less time/effort than maintaining a fork
for the many years of an LTS release. That reduced effort isn't just
free time to developers, the time is instead spent working on
enhancements and bug fixes. Maintaining a fork would prevent users from
getting important
How will the ARI/dialplan integration handle specific extensions? For
example if I have a stasis app which registers itself to dialplan as
'somecontext', how does this integration decide which extensions are
handled by the app? Does that app get calls for all extensions or only
specific
If using Fedora / similar systems it could be an SELinux policy issue.
In Fedora /usr/sbin/asterisk is a confined process where python is not.
On Fedora/CentOS SELinux will forbid Asterisk from taking certain
actions even if allowed by the file system permissions set by chmod.
It's possible
Sorry for the delayed reply I've been busy with other projects lately.
On 06/06/2018 04:38 AM, Alexander Traud wrote:
I love to help. However, before I apply your patch, I need a running
Test Suite first. In Ubuntu 18.04 LTS with branch Master, 11 test cases
fail for me, for example
An Asterisk WIKI page [1] exists. It has the basics but it's a bit out
of date.
Some notes:
* For the purpose of this test you should use Linux. It may work on
*BSD but that would introduce a new mostly untested variable. If
you are doing this in a VM the options which have been
I've posted python3 compatibility changes for the testsuite [1]. This
has gotten some testing (which found issues that have been fixed), but
really a full testsuite run is required to verify no errors. I do not
have adequate hardware to complete this task, I'm hoping someone in the
community
I'm working on some PR's to update the Asterisk testsuite to be
compatible with Python3 (without breaking Python2). An issue I've hit
is starpy. A PR to deal with compatibility was started but never
finished. Can we move starpy to gerrit.asterisk.org and convert the
github repository to a
ilt-ins.
I tested OSX El Capitain, the configure script detected __atomic
built-ins so this will not create any new issues for Mac.
[1] https://gist.github.com/coreyfarrell/c096dd335afee5502a6faee2a507b012
On 01/26/2018 03:20 PM, Matt Fredrickson wrote:
On Wed, Jan 24, 2018 at 3:50 PM, Co
If you are looking to work with FreeBSD I'd suggest starting at
FreshPorts [1]. Someone has done the work to get Asterisk working on
FreeBSD and they've got Asterisk 13.19.0 (they're keeping it current).
[1] https://www.freshports.org/net/asterisk13/
On 01/26/2018 06:41 AM, Alexander Traud
need __sync_fetch_and_nand or __sync_nand_and_fetch we
would need to require gcc-4.4. I've looked into the ast_flags macro's,
implementing them with atomic operations would would not require nand
functions.
[1] https://gerrit.asterisk.org/8049
On 01/24/2018 04:50 PM, Corey Farrell wrote:
I've
I've posted ASTERISK-27619 [1] proposing that we drop support for GCC
versions older than 4.1.2. Specifically we'd be requiring that either
__sync or __atomic builtin functions be available (I'm unsure what this
will do to clang requirements). gcc-4.1.2 was released in February 2007
and was
The only wiki page I found on non-Linux is [1]. Probably the clearest
statement is in the README.md [2]. I think we should probably add a
more negative spin to the comments about Other platforms, at minimum
state that they are not actively tested and may break at any time.
As for
I'd suggest opening a ticket at https://issues.asterisk.org, include
full debug logs and minimal test case for reproducing the issue. See
https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
for details about what to post on the new ticket.
Just to ask have you tested
I'd be interested in what Loway would have to say about this (or any
other reporting systems). ATTENDEDTRANSFER already uses 4 data fields,
so I don't have any issue if CONNECT sets the 4th data field to
linkedid. I don't see this as a format change, if we were talking about
adding a 5th
On 12/22/2017 12:36 PM, George Joseph wrote:
On Fri, Dec 22, 2017 at 9:50 AM, Corey Farrell <g...@cfware.com
<mailto:g...@cfware.com>> wrote:
On 12/22/2017 10:22 AM, George Joseph wrote:
On Thu, Dec 21, 2017 at 1:44 PM, Corey Farrell <g...@cfware.com
<mail
I hope we consider creating a res_redis first, then base everything off
that. If a redis library can be used directly by any module that would
fine but I'd like to see us avoid following the example of curl where
everything uses a dialplan function to perform requests. Dialplan
functions
On 12/22/2017 10:22 AM, George Joseph wrote:
On Thu, Dec 21, 2017 at 1:44 PM, Corey Farrell <g...@cfware.com
<mailto:g...@cfware.com>> wrote:
George asked that I post some scenarios where this would be useful.
1. You are about to create updated asterisk pack
13.19.0-rc1
would appear out of order (before 13.19.0). The difference between rc1
and final release is always small, but the number of new commits to
mainline between that time can be quite large.
On 12/21/2017 10:45 AM, Corey Farrell wrote:
I just read `git help merge` again and I think
nal step of mkrelease.py.
No changes will be needed to our handling of '.lastclean', please ignore
those comments as I was wrong.
On 12/21/2017 08:19 AM, George Joseph wrote:
On Wed, Dec 20, 2017 at 3:14 PM, Corey Farrell <g...@cfware.com
<mailto:g...@cfware.com>> wrote:
One thin
One thing that might improve this is if releases were merged back to the
major branch. Currently the commit "Update for 13.19.0-rc1" is on the
13.19 branch and tagged as 13.19.0-rc1. I believe that if we added 'git
checkout 13 && git merge 13.19.0-rc1' we would get better information
from
I'd like to propose that we make minor release branches temporary. What
I mean is 13.15, 15.0, etc. The sole purpose of the '13.15' was to
allow cherry-picking fixes to the 13.15.x series. As soon as 13.16.0
was released the '13.15' branch was closed to new merges. The latest
commit on
, Corey Farrell <g...@cfware.com
<mailto:g...@cfware.com>> wrote:
It would be nice to strip out some/most of the per version
conditionals in tests. The need to always cherry-pick changes is
the only pitfall I see with your proposal. If a test never had
per version
It would be nice to strip out some/most of the per version conditionals
in tests. The need to always cherry-pick changes is the only pitfall I
see with your proposal. If a test never had per version differences
then the cherry-pick is trivial amount of extra work, but for tests with
From your configure.ac:
> AST_EXT_LIB_CHECK([REDIS], [redis], [redisReaderCreate],
[hiredis/hiredis.h])
This tries linking with '-lredis', but it needs to use '-lhiredis':
AST_EXT_LIB_CHECK([REDIS], [hiredis], [redisReaderCreate],
[hiredis/hiredis.h])
In addition instead of using variable
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 started a new branch at team/nirs/cdr-redis-support
I'm having some issues integrating the hiredis library into the
automake. It seems to be configured
,
Le 01/11/2017 à 09:08, Matt Fredrickson a écrit :
Hey Corey,
On Tue, Oct 31, 2017 at 3:52 PM, Corey Farrell <g...@cfware.com> wrote:
Hello all,
autoconf is required to run ./bootstrap.sh after changing configure.ac or
*.m4 in any folder of the Asterisk source tree (including menuselect and
ovich" <nir.simionov...@gmail.com
<mailto:nir.simionov...@gmail.com>> wrote:
Hi Corey,
Our production environment is based on CentOS 6.9 and we're
slowly migrating to 7.X
I would be happy to check out your patches inside our testing facility.
On Tue, Oct 31,
Hello all,
autoconf is required to run ./bootstrap.sh after changing configure.ac
or *.m4 in any folder of the Asterisk source tree (including menuselect
and third-party folders). Currently we require 2.60, I'm working on a
patch that will require 2.64. Note autoconf is not required to run
I think astdb itself is an inappropriate place to deal with this. astdb
is initialized well before module preloads so it would be nearly
impossible for modules to provide the astdb implementation. In my
opinion astdb is a database "implementation", not a connector (and that
shouldn't
I propose that we restrict the kind of bugs/patches that are accepted
against chan_sip to only major/critical bugs, regressions and security
fixes. This means rejecting all new features, improvements and most
minor bugs (AKA known limitations) reported against chan_sip. I am not
proposing we
Correction on number of bugs, most PJSIP bugs are filed against resource
modules. If you select all of the "Resources/res_pj*" categories in
addition to Channels/chan_pjsip it shows 121 open bugs. People are
reporting issues against the pjsip modules, most are not against the
channel driver
CentOS 6 "Full Updates" support ends May 10th [1], after that "only
Security errata and select mission critical bug fixes will be
released". I think this combined with testing difficulties justifies
moving CentOS 6 into "extended" support for Asterisk. Probably worth
announcing to
Probably not the popular opinion, but I don't think any modules are
"generally" important enough to abort Asterisk start-up. For any module
that is important enough for a system the admin can use 'require' or
'preload-require' settings in modules.conf. The exception to this would
be when a
On 03/31/2017 10:47 AM, Kevin Harwell wrote:
On Thu, Mar 30, 2017 at 6:54 PM, Corey Farrell <g...@cfware.com
<mailto:g...@cfware.com>> wrote:
On 03/30/2017 07:14 PM, Kevin Harwell wrote:
I think it's worth referencing a previous discussion on this [1].
Yes, thank y
On 03/30/2017 07:14 PM, Kevin Harwell wrote:
[asterisk-branch-number].[minor].[patch]
Actually, the proposal might be better represented as the following:
[asterisk-branch-number].[major].[minor/patch]
Or another way to state it:
[asterisk-branch-number].[api breaking].[api non
On 03/16/2017 04:35 PM, Kevin Harwell wrote:
On Thu, Mar 16, 2017 at 2:54 PM, Matthew Jordan > wrote:
Questions than:
(1) Should there even be a line length rule?
I lean toward yes just to keep things sane (e.g. keeping someone
system.h*
>> void ast_system_get_address(struct ast_sockaddr *us);
>>
>> *system.c*
>> #include "asterisk/system.h"
>>
>> void ast_system_get_address(struct ast_sockaddr *us)
>> {
>>...
>> }
>>
>> *chan_sip.c*
>>
I'm not sure what you're trying to accomplish but this is not the approach
I'd take for expanding chan_sip. My advice is to simply put your system.c
into channels/sip. This will automatically link it into the binary
chan_sip module, but allow you to keep your source code in a separate
file. I
Overall I like it. I think leaving pbx_ael out of this is best as it
might not be straight forward. Anyone who wants pbx_ael to provide
information about config sources can create a follow-up patch once the
core functionality is implemented.
Some thoughts:
* It would be nice if the registrar
A review [1] has been posted to fix an issue where TLS servers would
not be restarted unless the bind address was changed. This would
prevent use of new certificates if available. Unfortunately this
change does cause an ABI change. Fields are added to public
structures 'struct ast_tls_config'
+1 for option 2. I'm against option 1 because the ARI version should
be enough to verify that the features of 14.2.0 are available.
One remaining question: how would we handle situations where a
critical or security bug requires a breaking change to ARI in Asterisk
13?
--
On Fri, Oct 21, 2016 at 12:32 PM, Kevin Harwell <kharw...@digium.com> wrote:
> On Fri, Oct 21, 2016 at 10:49 AM, Corey Farrell <g...@cfware.com> wrote:
>>
>>
>> I'm in favor per app config. I do not yet use ARI, but when I do I
>> will have '#tryinclude
On Fri, Oct 21, 2016 at 9:37 AM, Sébastien Duthil
wrote:
> On 19/10/16 11:59 AM, Mark Michelson wrote:
>> I had thought about making the list of channel variables per-ARI app
>> instead of global, but I don't know if that's actually desired.
>
> The underlining
On Mon, Oct 10, 2016 at 10:39 AM, Matthew Jordan <mjor...@digium.com> wrote:
>
> On Fri, Oct 7, 2016 at 10:31 AM, Corey Farrell <g...@cfware.com> wrote:
> > Many people don't like chan_sip, most people hate working with the code.
> > The rush to throw out c
Many people don't like chan_sip, most people hate working with the code.
The rush to throw out chan_sip when PJSIP isn't ready to be the only SIP
stack annoys me a bit. Nobody is forcing anyone to use or contribute to
chan_sip. Digium changed chan_sip from core to extended support so they
can
On Thu, Oct 6, 2016 at 6:44 AM, Dan Jenkins wrote:
> So if I understand properly you'd like to see a working group around "My
> first Asterisk Commit" - which would drive forward making contributing to
> the project easier - whether it was improving documentation, or
A custom command-line option is undesirable due to the limited number
of possible option codes / chance of future conflict if we need to add
a standard option. If you really need a custom option just for your
own chan_sip, it would be better to add it to sip.conf and parse it
during config load.
d leaves, and you have a non-empty value for BRIDGEPEER when the
> channel is with multiple parties in a bridge. You won't get VarSets on the
> other channels in a bridge when a channel enters or leaves, though. I'm not
> sure if that's something that people rely on or not.
>
&
It might be a problem if BRIDGEPEER is being compared to an empty string to
see if a channel is alone.
What if you made BRIDGEPEER into a built-in channel variable (like
${EPOCH}). This would mean that you wouldn't be setting any channels,
you'd only do a lookup when requested. One side-effect
Hello everyone,
I've opened ASTERISK-26171 to track my effort to include a "%p" pointer
(debugstorage) in the REF_DEBUG log. Providing this information to
refcounter.py allows it to remove matching ref and unref from the output.
It also allows detection of indirect leaks where one object leaked
I make use of a couple patches so the latest Fedora spec is a good enough
base for me. If other people want to see current binaries for EPEL
hopefully they'll speak up. I had a chance to look over the asterisk.spec
from Fedora master, I have a couple comments.
* res_ari_mailboxes.so should be
in the
> CentOS 6 container (gcc complained about not being able to result in a
> binary or something).
>
> Long story short, use fedpkg to build the RPMs from the Fedora
> repositories. Just clone the Fedora 23 version (which gets you Asterisk
> 13.3.2 as of now), and then build it o
Jared,
I just looked through the EPEL website at EPEL6 and EPEL7, only found
Asterisk 1.8. Can you point me to the spec file you are using or an
SRPM?
Thanks,
Corey
On Thu, Nov 19, 2015 at 2:02 PM, Jared Smith wrote:
>
> On Thu, Nov 19, 2015 at 10:14 AM, Matthew
CentOS/RHEL 6 are under production support until Nov 2020 [1] and journald
is not available. Even if journald could improve Asterisk logging, that's
not an acceptable reason to force Asterisk users to upgrade from a
supported OS. If there was a module that allowed Asterisk logging to talk
to
I've discovered that any option set in asterisk.conf cannot be overridden
using the command-line. For example, if you have verbose=0 in
asterisk.conf, 'asterisk -cvvv' will start with verbose 0. The same
goes for -f / nofork or -F / alwaysfork.
The only way to fix this is to change the
My additions to the list:
1) Procedure for 'git review' of security related patches. I think this
could be done with an asterisk-security mirror repository setup in gerrit
with restricted access. I know this is already being thought about, just
wanted to make sure it on the list.
2) Is there a
thinking trunk only. If anyone disagrees let me know.
Diffs
-
/trunk/Makefile 433966
Diff: https://reviewboard.asterisk.org/r/4580/diff/
Testing
---
Repeatedly ran 'make menuselect' with and without files in the 3 locations. No
one else has tested this.
Thanks,
Corey Farrell
,
Corey Farrell
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Used grep, found a few tests that were still using parameters with the new
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Missed one spot where we should conditionally return Asterisk Version
for failure in output.
- Corey Farrell
On April 13, 2015, 1 a.m., gareth wrote:
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Gerrit-PatchSet: 5
Gerrit-Project: asterisk
Gerrit-Branch: master
Gerrit-Owner: Matt Jordan mjor...@digium.com
Gerrit-Reviewer: Corey Farrell g...@cfware.com
Gerrit-Reviewer: George Joseph george.jos...@fairview5.com
Gerrit-Reviewer: Matt Jordan mjor...@digium.com
Corey Farrell has posted comments on this change.
Change subject: PEP8 fixes
..
Patch Set 2:
(2 comments)
The PJSIP tests no longer fail. I have a full run of the testsuite going now,
so far no issues.
https
a
single error is received for lines 1025 or more, lines 1024 or less succeed.
Thanks,
Corey Farrell
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Change subject: AMI: Fix improper handling of lines that are exactly 1025 bytes
long.
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AMI: Fix improper handling of lines that are exactly
Corey Farrell has uploaded a new change for review.
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Change subject: Optional API: Fix handling of sources that are both provider
and user.
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Optional API: Fix handling of sources that are both
Corey Farrell has uploaded a new change for review.
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Change subject: res_monitor: Add dependency on func_periodic_hook.
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res_monitor: Add dependency on func_periodic_hook.
OPTIONAL_API has
.
Thanks,
Corey Farrell
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This was previously posted for review at
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It appears that it's not this change that is broken, it's my computer.
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Still working out an issue.
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(1 comment)
So I'm a bit indifferent about the finding against main/asterisk.c
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https://gerrit.asterisk.org/#/c/61/1/main/asterisk.c
File main/asterisk.c:
Line
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https://gerrit.asterisk.org/#/c/61/1/main/asterisk.c
File main
.c, lines 492-494
https://reviewboard.asterisk.org/r/4108/diff/5-6/?file=71780#file71780line492
The comment doesn't make sense. How is destructor_fn supposed to
access values in the weak proxy? The real object is no longer linked with
the weakproxy.
Corey Farrell wrote
the included test with REF_DEBUG enabled under valgrind. No reference
leaks or improper memory access. Though this does not test for races, I don't
know of an automated way to do that.
Thanks,
Corey Farrell
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Change subject: astobj2: Add support for weakproxy objects.
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astobj2: Add support for weakproxy objects.
This implements weak references
. No reference
leaks or improper memory access. Though this does not test for races, I don't
know of an automated way to do that.
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https://gerrit.asterisk.org/#/c/54/3//COMMIT_MSG
Commit
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Change subject: main/editline: Add .gitignore.
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main/editline: Add .gitignore.
This patch adds a .gitignore for main/editline to ignore all
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Couple minor things then this is good to go.
https://gerrit.asterisk.org/#/c/42/1/.gitignore
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Change subject: sounds: Add a .gitignore file for downloaded sound tarballs
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Any reason we can't just add to the root .gitignore:
*.gz
This way
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These two lines should be replaced with 'goto done;'. This way we will
return RESULT_FAILURE.
- Corey Farrell
On April 9, 2015, 1:05 a.m., gareth wrote:
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On April 3, 2015, 6:37 a.m., Corey Farrell wrote:
/trunk/main/manager.c, line 4873
https://reviewboard.asterisk.org/r/4391/diff/3/?file=72904#file72904line4873
If we successfully ran the command, it seems unsafe to claim failure.
We have to assume the the caller doesn't actually
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Ship it!
Ship It!
- Corey Farrell
On April 9, 2015, 3:57
April 9, 2015, 3:20 p.m.)
Review request for Asterisk Developers and Corey Farrell.
Bugs: ASTERISK-24935
https://issues.asterisk.org/jira/browse/ASTERISK-24935
Repository: Asterisk
Description
---
res_pjsip_phoneprov_provider was leaking references to phoneprov
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Ship it!
Ship It!
- Corey Farrell
On April 9, 2015, 10:57
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