On 06/05/2017 03:17 PM, Matt Fredrickson wrote:
On Mon, Jun 5, 2017 at 2:31 PM, Joshua Colp <jc...@digium.com> wrote:
On Mon, Jun 5, 2017, at 04:21 PM, Mark Michelson wrote:
Hi folks,
For those of you following along at home, I recently published review
https://gerrit.asterisk.org/#/
that would pertain to a single call. We can get to that
discussion after this part gets settled.
Thanks,
Mark Michelson
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To UNSUB
sent in these
events, channel variables can already be included. In your manager.conf
file, in the general section, set
channelvars = foo,bar,baz
Every time a channel is sent in a manager event, the channel variables
foo, bar, and baz will be s
On 02/23/2017 08:28 AM, Andreas Krüger wrote:
Hi,
I was wondering with the following setup, if you would use Symmetric NAT.
+--+
+--+| 89.192.76.11 |
+--+
|Client 1 | LAN| Router | Internet|
.6
Should an issue be created on Jira?
Regards,
Ross
Hi Ross,
This may be the same issue that George discovered earlier today while
doing some testing. If you apply the code change on this review
https://gerrit.asterisk.org/#/c/4902/ , do you still see the iss
API design is more important at this stage.
If people could take a look at the SDP API page and offer feedback, that
would be great. There are some specific questions at the bottom of the wiki
page that would be good to get answered.
Thanks,
Mark Michelson
[1] https://wiki.asterisk.org/wiki/displ
On 12/05/2016 04:53 PM, Julian Fleischhauer wrote:
Hey all,
I have a problem using a custom C-file. The error ouput received when
compiling is given below.
error output
.../system.c:1330: undefined reference to `XML_ParserCreate'
...
.../system.c:1465: undefined reference to `sip_get_header'
sk and ARI will be out of sync (e.g. a user might
assume that Asterisk 14.3.0 would require ARI version 14.3.0, when
that's not actually the case).
Feel free to voice your support for one of these options or to suggest
something of your own.
Thanks,
Mark
On 11/09/2016 01:11 PM, wferre...@nc.rr.com wrote:
Hi,
I have a customer who is running 12.8.2 and since their software needs to be
certified at considerable expense and time will not be upgrading Asterisk
anytime soon. In addition due to security requirements the only connection
between
I have opened https://issues.asterisk.org/jira/browse/ASTERISK-26492 and
have attached the patch there. Feel free to try it out and let me know on
the issue how it works for you.
On Fri, Oct 21, 2016 at 8:37 AM, Sébastien Duthil <sdut...@proformatique.com
> wrote:
> On 19/10/16 11:59
On 10/20/2016 09:51 AM, marek cervenka wrote:
hi,
we have questions from busy call centers if the retry time parameter
can be 0 (wait time before next agent call)
in app_queue its prohibited
} else if (!strcasecmp(param, "retry")) {
q->retry = atoi(val);
On 10/19/2016 06:45 AM, Matthew Jordan wrote:
On Tue, Oct 18, 2016 at 4:02 PM, Joshua Colp wrote:
Matthew Jordan wrote:
There are a few wrinkles with emitting channel variables with
arbitrary events (of which StasisEnd would qualify).
When an event is emitted out of the
in_features.c. This approach would be
better than copying/pasting because it keeps the code in one spot and
makes use of already tested code.
Hope this was helpful,
Mark Michelson
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n requested. One side-effect is that VarSet
> events would never be produced for this variable, not sure how much this
> would matter given better events to monitor ConfBridge joins/parts?
>
> On Tue, Aug 9, 2016 at 7:01 PM, Mark Michelson <mmichel...@digium.com>
> wrote:
>
>
the variable.
Let me know what your thoughts are on the matter.
Thanks,
Mark Michelson
[1] https://gerrit.asterisk.org/#/c/3445
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I think you should feel free to submit this patch if you want it included
in mainline Asterisk. If you are interested in submitting you patch, see
https://wiki.asterisk.org/wiki/display/AST/Gerrit+Usage . That page details
how to submit a patch to the Asterisk project.
Just as a note, I don't
you look at the HTTP traffic, you may be able
to tell more clearly why the 404 is being sent.
It's certainly possible there's a bug in Asterisk, but I would be more
willing to bet that there's some sort of error in the python example.
Although, li
that had been used in subversion.
Mark Michelson
[1] https://gerrit.asterisk.org/#/admin/projects/libpri
[2] https://gerrit.asterisk.org/#/admin/projects/libss7
[3] https://github.com/asterisk/libpri
[4] https://github.com/asterisk/libss7
[5] http://git.asterisk.org/gitweb/?p=libpri.git;a=summary
[6
On 03/04/2016 09:14 AM, Jeremy Kister wrote:
after installing a module in asterisk, I want to also install some
docs for it for "core show application"
I can easily insert the appropriate xml in
documentation/core-en_US.xml but the only way I've found Asterisk will
use the new xml is to
In most cases, flushing an audiohook is not a bad thing, nor is it
something to be worried about. That's why it's a debug message. If you want
it in your logs, then set the core debug level to 1 so it shows up. You
will get more messages than you are used to seeing, but by only setting the
core
Either with Page or Confbridge itself. Either
1) The Page application is not telling Confbridge not to announce the
number of users in the conference properly.
2) The Confbridge application is announcing the number of users despite the
Page application's request for quiet.
As a workaround, you
On 12/04/2015 01:00 PM, Stian Hvatum wrote:
Hi,
I have a problem with an accompanying solution that I wish to share, but I am
not sure if it is valuable enough or correct enough to be suggested as a patch
to Asterisk.
When SIP-phones are members of a queue, they tend to accumulate "missed
On 12/06/2015 07:57 PM, Matthew Jordan wrote:
Hello all -
One of the efforts that a number of developers in the community here
at Digium have been at work at are cleaning up test failures exposed
by Jenkins [1]. One of these, in particular, has been rather difficult
to resolve - namely,
we modify the existing connection pooling code in
res_odbc to be more user-friendly?
Thanks in advance,
Mark Michelson
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To U
The answer to this is actually pretty simple: adding Referred-By in
outgoing SIP REFERs is simply not implemented in chan_pjsip's
chan_pjsip_transfer() function.
As far as the syntax required for the Transfer() application, that's
probably a case where that needs to be clarified in
)
Thank you Mark
*From:*asterisk-dev-boun...@lists.digium.com
mailto:asterisk-dev-boun...@lists.digium.com
[mailto:asterisk-dev-boun...@lists.digium.com] *On Behalf Of *Mark
Michelson
*Sent:* Tuesday, August 25, 2015 10:30 AM
*To:* Asterisk Developers Mailing List
*Subject:* Re: [asterisk-dev
);
/ End of changes /
pjsip_xfer_send_request(sub, packet);
ast_queue_control_data(session-channel, AST_CONTROL_TRANSFER,
message, sizeof(message));
}
*From:*asterisk-dev-boun...@lists.digium.com
[mailto:asterisk-dev-boun...@lists.digium.com] *On Behalf Of *Mark
Michelson
*Sent
ast_tcptls_close_session_file() in order to tear down the TCP
connection.
Hopefully that points you in the correct direction.
Mark Michelson
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for ld(1), it
sounds like load-time binding would, at most, cause module loading to
take longer. Are there any other potential issues to making this change?
Mark Michelson
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Hey fellow developers,
I have created a wiki page that details the finer points of sorcery caching
[1]. The intended audience for the page is Asterisk administrators. If you
have any critiques of the page, please feel free to mention them.
Mark Michelson
[1] https://wiki.asterisk.org/wiki
I've had a look at the page now, and here are my thoughts:
1) One thing that isn't really made clear is the interaction between
multiple sorcery wizards. In a real-world example, you'd want to hit the
memory_cache first and then hit the database afterwards if you couldn't
retrieve the object
On 04/14/2015 12:11 PM, Matthew Jordan wrote:
snip
Yup.
SO!
The question is: is this change worth having, or should it be scrapped
in favour of some alternate approach that makes use of other
technology? My feelings won't be hurt if the answer is ditch it and
do something else - this was a fun
Mark Michelson has posted comments on this change.
Change subject: res_pjsip: Add external PJSIP resolver implementation using
core DNS API.
..
Patch Set 1:
(3 comments)
https://gerrit.asterisk.org/#/c/75/1/main
Mark Michelson has posted comments on this change.
Change subject: .gitignore: Ignore tarballs (*.gz)
..
Patch Set 1: Code-Review+2 Verified+1
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Mark Michelson has posted comments on this change.
Change subject: .gitignore: Ignore tarballs (*.gz)
..
Patch Set 1: Code-Review+2 Verified+1
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Mark Michelson has submitted this change and it was merged.
Change subject: .gitignore: Ignore tarballs (*.gz)
..
.gitignore: Ignore tarballs (*.gz)
This patch updates the root .gitignore file to ignore files with a .gz
Mark Michelson has posted comments on this change.
Change subject: Add .gitignore and .gitreview files
..
Patch Set 1: Code-Review+2 Verified+1
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Mark Michelson has submitted this change and it was merged.
Change subject: Add .gitignore and .gitreview files
..
Add .gitignore and .gitreview files
Add the .gitignore and .gitreview files to the asterisk repo.
NB: You can
Mark Michelson has submitted this change and it was merged.
Change subject: main/editline: Add .gitignore.
..
main/editline: Add .gitignore.
This patch adds a .gitignore for main/editline to ignore all build results.
Change
Mark Michelson has posted comments on this change.
Change subject: main/editline: Add .gitignore.
..
Patch Set 1: Code-Review+2 Verified+1
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Mark Michelson has posted comments on this change.
Change subject: main/editline: Add .gitignore.
..
Patch Set 1: Code-Review+2 Verified+1
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Mark Michelson has submitted this change and it was merged.
Change subject: main/editline: Add .gitignore.
..
main/editline: Add .gitignore.
This patch adds a .gitignore for main/editline to ignore all build results.
Change
I have created a wiki page [1] with a test plan for DNS NAPTR/SRV for
PJSIP. Please take a moment to have a look at the page and provide any
feedback (on this list, not on the page please) about the test plan. Are
there any test cases that I left out that should be covered? Are there
any test
Mark Michelson has submitted this change and it was merged.
Change subject: Add .gitignore and .gitreview files
..
Add .gitignore and .gitreview files
Add the .gitignore and .gitreview files to the asterisk repo.
NB: You can
Mark Michelson has posted comments on this change.
Change subject: Add .gitignore and .gitreview files
..
Patch Set 1: Code-Review+2 Verified+1
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Mark Michelson has posted comments on this change.
Change subject: mapmantis: Remove dependency on digium_jira
..
Patch Set 2: Code-Review+1
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Hello Jared K. Smith,
I'd like you to reexamine a change. Please visit
https://gerrit.asterisk.org/69
to look at the new patch set (#2).
Change subject: digium_jira: Refactor module to wrap the Atlassian JIRA REST
client
Mark Michelson has submitted this change and it was merged.
Change subject: Fixing extconf compile
..
Fixing extconf compile
During the mass code deletion for clang support, a stray backslash was
left behind that was causing
Mark Michelson has submitted this change and it was merged.
Change subject: Fixing extconf compile
..
Fixing extconf compile
During the mass code deletion for clang support, a stray backslash was
left behind that was causing
Mark Michelson has submitted this change and it was merged.
Change subject: Fixing extconf compile
..
Fixing extconf compile
During the mass code deletion for clang support, a stray backslash was
left behind that was causing
Mark Michelson has posted comments on this change.
Change subject: Fixing extconf compile
..
Patch Set 1: Code-Review+2 Verified+1
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Change subject: Fixing extconf compile
..
Patch Set 1: Code-Review+2 Verified+1
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Change subject: Fixing extconf compile
..
Patch Set 1: Code-Review+2 Verified+1
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Mark Michelson has submitted this change and it was merged.
Change subject: .gitignore: Ignore tarballs (*.gz)
..
.gitignore: Ignore tarballs (*.gz)
This patch updates the root .gitignore file to ignore files with a .gz
Mark Michelson has posted comments on this change.
Change subject: digium_jira: Refactor module to wrap the Atlassian JIRA REST
client
..
Patch Set 2:
(2 comments)
https://gerrit.asterisk.org/#/c/69/2/digium_jira.py
File
channels, frames that
come into the core don't have to go through any extra translations in order to
be used in audiohooks.
- Mark Michelson
On April 9, 2015, 8:50 p.m., rmudgett wrote:
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Ship it!
Ship It!
- Mark Michelson
On April 9, 2015, 2:57
Mark Michelson has posted comments on this change.
Change subject: rest_api/applications/stasisstatus: Make run-test executable
..
Patch Set 1: Code-Review+2 Verified+1
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do another
NAPTR lookup on the returned record? After all, the next NAPTR lookup might
give me back the SIP records I want. What if I get a mix of empty flags and
empty service records AND SIP+D2X records with s flag? What then?
- Mark Michelson
On April 8, 2015, 7:37 p.m., Joshua Colp wrote
Mark Michelson has submitted this change and it was merged.
Change subject: rest_api/applications/stasisstatus: Make run-test executable
..
rest_api/applications/stasisstatus: Make run-test executable
If it isn't executable
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Ship it!
Ship It!
- Mark Michelson
On April 9, 2015, 4:50
/trunk/include/asterisk/dns_test.h PRE-CREATION
/trunk/include/asterisk/dns_internal.h 434218
Diff: https://reviewboard.asterisk.org/r/4598/diff/
Testing
---
All DNS unit tests continue to pass.
Thanks,
Mark Michelson
is in the same
location where manager events are sent, you may also be able to get away with
fewer heap allocations, too, which is another bonus.
- Mark Michelson
On April 7, 2015, 6:25 p.m., warren smith wrote:
---
This is an automatically
Mark Michelson has posted comments on this change.
Change subject: stasis: set a channel variable on websocket disconnect error
..
Patch Set 5: Verified+1
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Change subject: stasis: set a channel variable on websocket disconnect error
..
Patch Set 5: Code-Review+2
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Change subject: stasis: set a channel variable on websocket disconnect error
..
Patch Set 5: -Verified
Removing the Verified since there are still outstanding findings
Mark Michelson has posted comments on this change.
Change subject: stasis: set a channel variable on websocket disconnect error
..
Patch Set 5: Code-Review+1
(2 comments)
It's looking good by me!
https://gerrit.asterisk.org
be replaced with a key-value store that has the
concept of replication built into it. If that were done, then presumably, there
would be no need to involve Stasis or SIP PUBLISH in the process of sharing
values between Asterisk boxes.
- Mark Michelson
On April 6, 2015, 2:22 a.m., Matt Jordan wrote
Mark Michelson has posted comments on this change.
Change subject: stasis: set a channel variable on websocket disconnect error
..
Patch Set 10: Code-Review+2 Verified+1
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Mark Michelson has submitted this change and it was merged.
Change subject: stasis: set a channel variable on websocket disconnect error
..
stasis: set a channel variable on websocket disconnect error
This test is to ensure
be relegated to a separate option.
- Mark Michelson
On April 6, 2015, 6:32 p.m., Kevin Harwell wrote:
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Mark Michelson has uploaded a new patch set (#3).
Change subject: sip_attended_transfer now supports pre-12 Asterisk versions.
..
sip_attended_transfer now supports pre-12 Asterisk versions.
The sip_attended transfer test
Mark Michelson has uploaded a new patch set (#3).
Change subject: sip_attended_transfer now supports pre-12 Asterisk versions.
..
sip_attended_transfer now supports pre-12 Asterisk versions.
The sip_attended transfer test
was present in
pjsip.conf. If default system configuration setup on startup fails, then
res_pjsip.so should fail to load.
I think an assertion here that cfg is non-NULL would do the trick instead
of trying again to create default configuration settings
- Mark Michelson
On April 7, 2015, 4:05
is not filled
in at all. What has been done to test this change?
- Mark Michelson
On April 6, 2015, 3:53 p.m., Juergen Spies wrote:
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On April 7, 2015, 1:05 a.m., rmudgett wrote:
/branches/13/apps/app_minivm.c, line 1842
https://reviewboard.asterisk.org/r/4541/diff/3/?file=73407#file73407line1842
Missing the !
if (!ast_strlen_zero())
Diederik de Groot wrote:
Thanks again for checking my stuff,
/trunk/include/asterisk/dns_internal.h 434218
Diff: https://reviewboard.asterisk.org/r/4598/diff/
Testing
---
All DNS unit tests continue to pass.
Thanks,
Mark Michelson
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Ship It!
- Mark Michelson
On April 6, 2015, 8:20
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On April 6, 2015, 10:46 p.m., Mark Michelson wrote
Mark Michelson has posted comments on this change.
Change subject: Enable support for directory containing custom tests.
..
Patch Set 1: Code-Review+1
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Mark Michelson has posted comments on this change.
Change subject: Testsuite: New test for FAX via PJSIP T38 with authentication
..
Patch Set 4: Code-Review+2
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Ship It!
- Mark Michelson
On March 25, 2015, 4:10
/group/dns/main/dns_core.c 433885
/team/group/dns/include/asterisk/dns_internal.h 433885
Diff: https://reviewboard.asterisk.org/r/4542/diff/
Testing
---
All previous DNS tests continue to pass, and all new tests added in this review
pass as well.
Thanks,
Mark Michelson
://reviewboard.asterisk.org/r/4579/diff/
Testing
---
The bug itself is incredibly difficult to have happen under normal
circumstances, but I have confirmed that this patch has not hindered operations
any.
Thanks,
Mark Michelson
the Posting Code
to Review Board section on this wiki page:
https://wiki.asterisk.org/wiki/display/AST/Review+Board+Usage
- Mark Michelson
On April 3, 2015, 4:56 p.m., Rodrigo Ramirez Norambuena wrote:
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Mark Michelson has abandoned this change.
Change subject: Add SIP attended transfer for Asterisk 11.
..
Abandoned
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Change subject: Add SIP attended transfer for Asterisk 11.
..
Patch Set 1:
I am abandoning this change in favor of change /c/29/
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Change subject: sip_attended_transfer now supports pre-12 Asterisk versions.
..
Patch Set 1:
(1 comment)
https://gerrit.asterisk.org/#/c/29/1/tests/channels/SIP
Mark Michelson has uploaded a new change for review.
https://gerrit.asterisk.org/29
Change subject: sip_attended_transfer now supports pre-12 Asterisk versions.
..
sip_attended_transfer now supports pre-12 Asterisk versions
Mark Michelson has uploaded a new patch set (#2).
Change subject: sip_attended_transfer now supports pre-12 Asterisk versions.
..
sip_attended_transfer now supports pre-12 Asterisk versions.
The sip_attended transfer test
---
On March 27, 2015, 2:45 p.m., Mark Michelson wrote:
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added in this review
pass as well.
Thanks,
Mark Michelson
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On April 1, 2015, 1:07 a.m., Matt Jordan wrote:
trunk/channels/chan_iax2.c, line 12370
https://reviewboard.asterisk.org/r/4536/diff/1/?file=72980#file72980line12370
The usage of max_retries here feels arbitrary. I'm not against this
being controlled more dynamically based on the
transaction, then
there would only ever be a single timer running, and the possibility of races
is eliminated.
- Mark Michelson
On March 31, 2015, 5:57 a.m., George Joseph wrote:
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Change subject: stasis: set a channel variable on websocket disconnect error
..
Patch Set 1: Code-Review+1
(4 comments)
I noticed that I gave the Code-Review a 0 last time. I
even lower than that is probably
fine).
- Mark Michelson
On March 29, 2015, 6:05 p.m., Joshua Colp wrote:
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Gerrit-MessageType: newpatchset
Gerrit-Change-Id: I1676801d90bcafc28ba25e8b6889f40ab08cc90e
Gerrit-PatchSet: 3
Gerrit-Project: testsuite
Gerrit-Branch: master
Gerrit-Owner: Mark Michelson mmichel...@digium.com
Gerrit-Reviewer: Ashley Sanders asand...@digium.com
Gerrit-Reviewer: Mark Michelson
Mark Michelson has uploaded a new patch set (#3).
Change subject: Rewrite sip_attended_transfer test to stop failing.
..
Rewrite sip_attended_transfer test to stop failing.
The sip_attended_transfer test has been bouncing
Mark Michelson has posted comments on this change.
Change subject: rest_api/channels/snoop_spy: Stop test on bridge destruction
..
Patch Set 1: Code-Review+1
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- Mark Michelson
On April 1, 2015, 9:20
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