.
It may be worthwhile having an assert in here, or something that indicates
that this should not have occurred - even if we do try to 'handle it
gracefully'.
- Matt Jordan
On Nov. 12, 2013, 6:43 p.m., Joshua Colp wrote
On Nov. 14, 2013, 4:53 p.m., Mark Michelson wrote:
/branches/12/res/res_stasis_snoop.c, line 142
https://reviewboard.asterisk.org/r/3003/diff/3/?file=48153#file48153line142
Hm, would it be more useful to publish the snoop messages on the spyee
channel's topic? I would suspect
/outbound_reregister_from/sipp/register.xml
https://reviewboard.asterisk.org/r/3020/#comment19584
And here
/asterisk/trunk/tests/channels/SIP/outbound_reregister_from/sipp/register.xml
https://reviewboard.asterisk.org/r/3020/#comment19583
Here too
- Matt Jordan
On Nov. 15, 2013, 9:42
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On Nov. 22, 2013, 5:54 p.m
/#comment19610
Is there any reason why we aren't exporting all ast_sip* symbols?
branches/12/res/res_pjsip_pubsub.c
https://reviewboard.asterisk.org/r/2959/#comment19611
Blobs
- Matt Jordan
On Nov. 15, 2013, 7:58 p.m., Kevin Harwell wrote
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- Matt Jordan
On Nov. 22, 2013, 7:36 p.m
shipping this in.
- Matt Jordan
On Nov. 15, 2013, 7:58 p.m., Kevin Harwell wrote:
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On Nov. 23, 2013, 5:38 p.m., Paul Belanger wrote:
Wouldn't this be the same path as /dial? I know asterisk has native
functionality to play numbers however, feel like libraries on top of ARI
could handle this. /me shrugs.
Joshua Colp wrote:
To play numbers in various languages
On Nov. 21, 2013, 4:40 p.m., Tilghman Lesher wrote:
Shouldn't something like this be a channel function from which you retrieve
the value, instead of specifying a variable into which the name is placed?
Seems like a significant regression in spitting things out to channel
variables,
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On Nov. 20, 2013, 11:01
hangups on these as well.
/asterisk/trunk/tests/apps/say_interrupt/test-config.yaml
https://reviewboard.asterisk.org/r/3018/#comment19645
While it works just fine, I don't think you need to shout LOCAL. Local is
just fine :-)
- Matt Jordan
On Nov. 15, 2013, 12:07 a.m., Jonathan Rose wrote
On Nov. 21, 2013, 3:38 p.m., Joshua Colp wrote:
I don't see where the user event is actually checked... shouldn't you have
specified the requirement of a Yay event in your test-config.yaml?
Mark Michelson wrote:
Nope, SimpleTestCase determines pass/fail based solely on the number
/#comment19656
I'd throw out an error here as well, at least to the CLI - %s not a valid
number , etc.
- Matt Jordan
On Nov. 23, 2013, 2:35 p.m., Joshua Colp wrote:
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On Nov. 26, 2013, 5:07 p.m
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On Nov. 21, 2013, 8:47 p.m
to
'subscribe-device-state'
- Matt Jordan
On Nov. 25, 2013, 6 p.m., Kevin Harwell wrote:
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On Nov. 25, 2013, 6:31 p.m
/appdocsxml.xslt PRE-CREATION
/branches/12/doc/appdocsxml.dtd 403234
/branches/12/Makefile 403234
Diff: https://reviewboard.asterisk.org/r/3035/diff/
Testing
---
Testsuite test written and available here:
https://reviewboard.asterisk.org/r/3034/
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Matt Jordan
-CREATION
Diff: https://reviewboard.asterisk.org/r/3034/diff/
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On Nov. 28, 2013, 9:16 p.m
://reviewboard.asterisk.org/r/3038/diff/
Testing
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and the values extracted in
this function, or pbx_substitute_variables should be responsible for unlocking
the contexts lock before calling the variable substiution function.
- Matt Jordan
On Dec. 6, 2013, 10:44 p.m., Scott Griepentrog wrote
that
they don't otherwise consume.
- Matt Jordan
On Dec. 5, 2013, 5:20 p.m., Joshua Colp wrote:
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to detect/find
errors can't really be reviewed until this gets finalized.
- Matt Jordan
On Nov. 25, 2013, 11:10 p.m., Scott Griepentrog wrote:
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On Dec. 2, 2013, 5:35 p.m
/branches/12/UPGRADE.txt 403448
/branches/12/CHANGES 403448
Diff: https://reviewboard.asterisk.org/r/3057/diff/
Testing
---
* The CDR tests pass (now deterministicly)
* The hangup handler tests now pass, as they can get reliable values in the h
extension from the CDR engine
Thanks,
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.
/asterisk/trunk/tests/fax/tests.yaml
https://reviewboard.asterisk.org/r/3048/#comment19730
While you're here (and when you've committed this), go ahead and throw the
existing chan_sip based tests under a 'sip' directory.
- Matt Jordan
On Dec. 5, 2013, 5:24 p.m., Joshua Colp wrote
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On Dec. 9, 2013, 1:27 a.m
On Dec. 9, 2013, 8:02 p.m., Mark Michelson wrote:
The loop situation Matt described is possible.
When ast_channel_suppress() is called, it results in AST_FRAME_VOICE being
turned into AST_FRAME_NULL.
If there is a jitter buffer on a channel, then AST_FRAME_NULL gets turned
into
On Dec. 7, 2013, 3:48 p.m., Matt Jordan wrote:
/branches/1.8/main/pbx.c, lines 4477-4491
https://reviewboard.asterisk.org/r/3055/diff/4/?file=49180#file49180line4477
I don't think we need the extra lock here.
The purpose of the contexts lock is to prevent changes
and available here:
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handy.
- Matt
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/branches/12/channels/pjsip/dialplan_functions.c PRE-CREATION
/branches/12/channels/chan_pjsip.c 403470
/branches/12/channels/Makefile 403470
Diff: https://reviewboard.asterisk.org/r/3038/diff/
Testing
---
See https://reviewboard.asterisk.org/r/3037
Thanks,
Matt Jordan
://reviewboard.asterisk.org/r/3052/#comment19754
Since nothing else calls get_content_type, you don't really need this
function. You could have is_content_type be get_content_type.
- Matt Jordan
On Dec. 5, 2013, 10:42 p.m., Kevin Harwell wrote
On Dec. 4, 2013, 9:02 p.m., Mark Michelson wrote:
/branches/12/channels/pjsip/dialplan_functions.c, lines 316-327
https://reviewboard.asterisk.org/r/3038/diff/1/?file=48949#file48949line316
These descriptions are inaccurate when used on outgoing channels.
Matt Jordan wrote
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pass, as they can get reliable values in the h
extension from the CDR engine
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the mailbox. It should still just be a string unique
ID.
This design flaw has a ripple effect through this entire patch - but until we
resolve this fundamental problem, I'm not sure I'd look at this patch any
further.
- Matt Jordan
On Dec. 9, 2013, 9:51 p.m., rmudgett wrote
, and chan_console. I'm
pretty sure all of the old console tests no longer use chan_console.
- Matt Jordan
On Dec. 9, 2013, 10:52 p.m., Kevin Harwell wrote:
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On Dec. 10, 2013, 2:34 a.m., Matt Jordan wrote
*e_data = NULL;
...
if (!ast_strlen_zero(e-data)) {
*passdata = '\0';
} else {
e_data = ast_strdupa(e-data);
}
ast_unlock_contexts();
if (!ast_strlen_zero(e_data)) {
pbx_substitute_variables(...);
}
etc.
- Matt Jordan
On Dec
and available here:
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On Dec. 11, 2013, 12:16
this patch: bad audio, bad performance.
With this patch: acceptable performance/audio.
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On Dec. 10, 2013, 10:48 a.m., Matt Jordan wrote
now pass, as they can get reliable values in the h
extension from the CDR engine
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On Dec. 12, 2013, 11:48 a.m., opticron wrote:
/branches/12/funcs/func_cdr.c, lines 472-474
https://reviewboard.asterisk.org/r/3057/diff/2/?file=49301#file49301line472
Return and reduce indentation.
Matt Jordan wrote:
Normally I'd agree, but that approach has one drawback
On Dec. 12, 2013, 11:48 a.m., opticron wrote:
/branches/12/funcs/func_cdr.c, lines 472-474
https://reviewboard.asterisk.org/r/3057/diff/2/?file=49301#file49301line472
Return and reduce indentation.
Matt Jordan wrote:
Normally I'd agree, but that approach has one drawback
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On Dec. 13, 2013, 12:50
://reviewboard.asterisk.org/r/3069/
---
(Updated Dec. 12, 2013, 3:59 p.m.)
Review request for Asterisk Developers, Joshua Colp, Jonathan Rose, Matt
Jordan, and rmudgett.
Bugs: ASTERISK-22936
https://issues.asterisk.org/jira/browse/ASTERISK
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- Matt Jordan
On Dec. 11, 2013, 9:31 a.m
there
are no other escaped characters that have special meaning in config files.
I'm not sure we want to go crazy here, but ideally we would have a way to
re-write out a '\' when we write out the config.
- Matt Jordan
On Dec. 16, 2013, 7:18 p.m., George Joseph wrote
/r/3067/#comment19887
I think a comment here stating that the locking inversion is okay here,
as no other thread can obtain the channel prior to linking it into the channels
container.
Normally, this would be a big ol' red flag :-)
- Matt Jordan
On Dec. 12, 2013, 11:43 a.m
there - we ran into it only
through some seriously aggressive use of channel redirect.
We should also probably put a \since 12.0.0 before the \note you've added that
the channel is returned locked.
- Matt Jordan
On Dec. 12, 2013, 11:43 a.m., Joshua Colp wrote
after all)
- Matt Jordan
On Dec. 16, 2013, 3:26 a.m., wdoekes wrote:
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/resource_channels.c 403703
Diff: https://reviewboard.asterisk.org/r/3068/diff/
Testing
---
Without this patch: bad audio, bad performance.
With this patch: acceptable performance/audio.
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On Dec. 17, 2013, 4:13 p.m
This change should be documented in CHANGES/UPGRADE as well
- Matt Jordan
On Dec. 13, 2013, 5:45 p.m., rmudgett wrote:
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)
/asterisk/trunk/tests/fax/pjsip/t38/run-test
https://reviewboard.asterisk.org/r/3048/#comment19911
Blob
- Matt Jordan
On Dec. 11, 2013, 6:33 a.m., Joshua Colp wrote:
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For formatting consistency, add a line of '' below VMCOUNT.
- Matt Jordan
On Dec. 18, 2013, 12:51 p.m., rmudgett wrote:
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now pass, as they can get reliable values in the h
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On Dec. 19, 2013, 9:10 a.m., opticron wrote:
Ship It!
Well, hold on. Don't Ship It! just yet. :-)
The fact that this isn't called in Asterisk 1.8 doesn't mean it shouldn't be
called - it just means someone, at some point in time, fixed the glitch. If
we aren't calling this function, then
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On Dec. 19, 2013, 2:06 p.m
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On Dec. 19, 2013, 2:04 p.m
external links.
branches/12/configs/pjsip.conf.sample
https://reviewboard.asterisk.org/r/3086/#comment19938
May as well align the comment with the others
- Matt Jordan
On Dec. 19, 2013, 6:18 p.m., rnewton wrote
res_pjsip_endpoint_identifier_ip.h and the
exports file.
- Matt Jordan
On Dec. 19, 2013, 9:04 p.m., George Joseph wrote:
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On Dec. 19, 2013, 3 p.m
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On Dec. 20, 2013, 9:28 a.m
expected lifetime.
I personally dislike this use of RAII_VAR - it's wasteful in its processing
and it takes careful looking to understand that its cleaning up after the bump
from ao2_iterator_next; but at least we're not mixing it with the for loop now.
- Matt Jordan
On Dec. 20, 2013, 1
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On Dec. 20, 2013, 3:50 p.m
.
All CDR and hangup handler test in the test suite now pass. Previously, the
hangup handler tests would fail due to getting wrong data back from the CDR
engine when reading a value using the CDR function.
Thanks,
Matt Jordan
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On Dec. 27, 2013, 5:28 p.m
or 512.
- Matt Jordan
On Dec. 27, 2013, 5:28 p.m., Kevin Harwell wrote:
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On Dec. 30, 2013, 3:26 p.m., Kevin Harwell wrote:
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file with the
enhancement.
- Matt Jordan
On Dec. 30, 2013, 3:26 p.m., Kevin Harwell wrote:
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to have that here
as well.
- Matt Jordan
On Dec. 24, 2013, 10:49 a.m., Tzafrir Cohen wrote:
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. This is not a bug in 12+.
Diffs
-
/branches/11/channels/chan_sip.c 404856
Diff: https://reviewboard.asterisk.org/r/3107/diff/
Testing
---
Prior to the patch, the channels were stuck.
With the patch, the two channels are properly hung up.
Thanks,
Matt Jordan
:
Explosions. With the patch: Asterisk allowed the users to be kicked without
terminating itself.
Note that the tests will be posted under a separate review.
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With the patch, the two channels are properly hung up.
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, 2013, 1:24 p.m., Matt Jordan wrote:
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(Updated Dec. 31, 2013, 1:24 p.m
.
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On Jan. 3, 2014, 11:12 a.m., Mark Michelson wrote:
So...I just don't really like this.
1) Making a macro like this makes the code more esoteric than it previously
was.
2) The fact that the loop_obj has to be NULLed out before breaking or
returning from the loop if no reference
---
An automated test was written that reproduced the scenario. Without the patch:
Explosions. With the patch: Asterisk allowed the users to be kicked without
terminating itself.
Note that the tests will be posted under a separate review.
Thanks,
Matt Jordan
that MoH will now support wide
band audio. Nice work!
- Matt Jordan
On Jan. 9, 2014, 8:50 a.m., wdoekes wrote:
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on the description of pjsip_tx_data_invalidate_msg, I would suspect
we only need to call this once before returning. Do we need to call it for each
change we make to tdata?
- Matt Jordan
On Jan. 6, 2014, 2:06 p.m., Joshua Colp wrote
On Jan. 7, 2014, 3:29 a.m., Olle E Johansson wrote:
Please also check the sender IP address in the IP packet. That's where
things go wrong with chan_sip. Thanks.
Olle - just so I understand your comment, are you referring to what would
happen if the sender was behind a NAT?
- Matt
On Jan. 9, 2014, 2:11 p.m., opticron wrote:
/branches/12/res/res_pjsip_dialog_info.c, line 109
https://reviewboard.asterisk.org/r/3093/diff/2/?file=51127#file51127line109
dialog-info+xml seems to get used quite a bit in this file. Put this
in a #define or a static variable.
I'd
as much editing.
Plus it makes this look like the rest of our code, which is nice.
- Matt Jordan
On Jan. 9, 2014, 1:30 p.m., rmudgett wrote:
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.
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Matt Jordan
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- Matt Jordan
On Jan. 9, 2014, 5:07 p.m
called when
Asterisk is started/stopped.
- Matt Jordan
On Jan. 6, 2014, 3:59 p.m., Jonathan Rose wrote:
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