Diff: https://reviewboard.asterisk.org/r/3318/diff/
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Testing
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/group/media_formats-reviewed/funcs/func_channel.c 409286
Diff: https://reviewboard.asterisk.org/r/3321/diff/
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On March 6, 2014, 7:58 p.m
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On March 6, 2014, 8:02 p.m
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On March 6, 2014, 8:05 p.m
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- Joshua Colp
On March 6, 2014, 8:15 p.m
. (Although, since channels stick
around... when would you need to use /n? Hrm...)
So, thoughts?
[1] http://svn.digium.com/svn/asterisk/team/file/bridge_unreal/
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optimize and how it
isn't inside of chan_local... what do you think NOW? ;)
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interface went into the message.
Some additional testing by others would be nice.
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Update the documentation ^_^
- Joshua Colp
On March 11
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- Joshua Colp
On March 12, 2014, 5:54
it
and confirmed that the DNS client was disabled. Set to auto (explicitly and by
default) and confirmed that the system nameservers were used.
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Olle E. Johansson wrote:
On 12 Mar 2014, at 19:53, Joshua Colp reviewbo...@asterisk.org
mailto:reviewbo...@asterisk.org wrote:
This change adds a configuration option for setting nameservers to
be used by the PJSIP DNS client. If this option is not set then the
system nameservers
they were used by PJSIP. Disabled it
and confirmed that the DNS client was disabled. Set to auto (explicitly and by
default) and confirmed that the system nameservers were used.
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On 12 Mar 2014, at 19:53, Joshua Colp reviewbo...@asterisk.org
mailto:reviewbo...@asterisk.org wrote:
This change adds a configuration option for setting nameservers to
be used by the PJSIP DNS client. If this option is not set then the
system nameservers
Olle E. Johansson wrote:
On 13 Mar 2014, at 11:42, Joshua Colp reviewbo...@asterisk.org
mailto:reviewbo...@asterisk.org wrote:
In case others are wondering as Olle was:
PJLIB-Util (part of pjproject) provides a DNS client which can
optionally (but is highly suggested) to be used with PJSIP
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- Joshua Colp
On March 12, 2014, 9:52
of the event you are forwarding it is unrelated to the channel
and allowed to pass.
- Joshua Colp
On March 12, 2014, 6:02 p.m., Jonathan Rose wrote:
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On March 10, 2014, 7:13
://reviewboard.asterisk.org/r/3324/#comment20800
Why not just use ast_asprintf for both of the to_str functions?
- Joshua Colp
On March 10, 2014, 7:04 p.m., Jonathan Rose wrote:
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Speeling mistake.
- Joshua Colp
On March 13, 2014, 3:57 p.m., Jonathan Rose wrote:
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On March 13, 2014, 4:11
their DNS
resolution pluggable. That would allow us to modify the res_pjsip
stack to use Asterisk's DNS support.
Inobtrusive, not obtrusive. Two different words.
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introducing problems, such as this. I
think reducing the chance of a crash is good enough but if it still presents
a problem we may want to think about reverting the original change.
- Joshua Colp
On March 14, 2014, 6:35 p.m., Scott Griepentrog wrote
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resulting bridge technologies are as expected.
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/team/group/media_formats-reviewed/funcs/func_pitchshift.c 409286
/team/group/media_formats-reviewed/funcs/func_frame_trace.c 409286
/team/group/media_formats-reviewed/funcs/func_channel.c 409286
Diff: https://reviewboard.asterisk.org/r/3321/diff/
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/branches/12/res/ari/resource_bridges.h 410512
/branches/12/CHANGES 410512
Diff: https://reviewboard.asterisk.org/r/3359/diff/
Testing
---
Using Swagger to create bridges of various types and confirmed that the
resulting bridge technologies are as expected.
Thanks,
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- 21e526ad3040 (branchpoint), add pjsip debug option
- 21e526ad3040 - 28887f25a46f (head), Create queue tables
- 21e526ad3040 - 4c573e7135bd (head), Fix tos and cos
field types
Looks like we're going to need to change them to get a clean upgrade path.
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Matthew Jordan wrote:
snip
Technically, it's probably not hard. I wonder how horrible it'd be for
end users, however.
Probably pretty horrible, since it's basically changing everything.
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/res_pjsip_multihomed.c 410669
Diff: https://reviewboard.asterisk.org/r/3369/diff/
Testing
---
Ran tests that were failing because of this, they now pass!
Ran normal calls which should trigger rewriting, and stuff got rewritten.
Thanks,
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calls which should trigger rewriting, and stuff got rewritten.
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. Disabled it
and confirmed that the DNS client was disabled. Set to auto (explicitly and by
default) and confirmed that the system nameservers were used.
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are as expected.
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Diff: https://reviewboard.asterisk.org/r/3318/diff/
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/media_formats-reviewed/include/asterisk/format.h 410186
/team/group/media_formats-reviewed/include/asterisk/codec.h 410188
/team/group/media_formats-reviewed/include/asterisk/callerid.h 409286
Diff: https://reviewboard.asterisk.org/r/3319/diff/
Testing
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409286
/team/group/media_formats-reviewed/bridges/bridge_simple.c 409286
/team/group/media_formats-reviewed/bridges/bridge_holding.c 409286
Diff: https://reviewboard.asterisk.org/r/3318/diff/
Testing
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://reviewboard.asterisk.org/r/3322/diff/
Testing
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- Joshua Colp
On March 24, 2014, 7:31
via the bucket API. Additional
information about the sounds is provided in the form of metadata.
Diffs
-
/trunk/res/res_bucket_sounds.c PRE-CREATION
Diff: https://reviewboard.asterisk.org/r/3251/diff/
Testing
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Executed CLI commands, output is as expected.
Thanks,
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/
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/channels/chan_jingle.c 409286
/team/group/media_formats-reviewed/channels/chan_gtalk.c 409286
Diff: https://reviewboard.asterisk.org/r/3390/diff/
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/team/group/media_formats-reviewed/channels/chan_skinny.c 409286
/team/group/media_formats-reviewed/channels/chan_mgcp.c 409286
Diff: https://reviewboard.asterisk.org/r/3388/diff/
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On March 24, 2014, 7:31
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- Joshua Colp
On March 25, 2014, 11:33
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- Joshua Colp
On March 26, 2014, 6:20
://issues.asterisk.org/jira/browse/ASTERISK-22961
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On March 21, 2014, 2:20
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- Joshua Colp
On April 2, 2014, 10:02
. The audiohook got just has to
replace the frame with ast_null_frame if a manipulation audiohook
returns NULL.
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allocation in most cases.
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it would be what I would do, indeed.
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is that there's no way around it, just go for
option #2 and do what Josh says again... ;-)
I think it would be what I would do, indeed.
Do you always do what you say you should do?
I fight for the users.
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as expected.
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documentation of how it works
would also be great. I'd like to have the documentation be such that someone
coming fresh into this could have a really good vision and understanding of
what is going on.
- Joshua Colp
On April 9, 2014, 7:49 p.m., Joshua Colp wrote
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On April 9, 2014, 12:46
, 10:46 p.m.)
Review request for Asterisk Developers, David Lee, Joshua Colp, and Matt
Jordan.
Bugs: ASTERISK-23433
https://issues.asterisk.org/jira/browse/ASTERISK-23433
Repository: Asterisk
Description
---
Adds a tones URI type to the playback resource. The tone can
these you don't
refer to them as plural. Busy tone for an example. Thoughts?
- Joshua Colp
On April 8, 2014, 10:46 p.m., Jonathan Rose wrote:
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- Joshua Colp
On April 14, 2014, 9:17
numbers of Local channels in chains (100, 200, 300) and confirmed
that stuff works as expected.
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/3427/#comment21371
This log message isn't really useful.
/branches/12/main/app.c
https://reviewboard.asterisk.org/r/3427/#comment21372
This doesn't allow the tone zone to be specified by the user, restricting
us to US only. This needs to be made configurable somehow.
- Joshua Colp
/res_pjsip.c 412327
Diff: https://reviewboard.asterisk.org/r/3448/diff/
Testing
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Reloaded multiple times changing configuration and confirmed that changes were
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Where do the anonymous values now get placed into the num/name/domain?
- Joshua Colp
On April 16, 2014, 9:23 p.m., Jonathan Rose wrote:
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was such that it wasn't
allowed... then the anonymous header would get added. Reading this logic I
don't see how the header would get added.
- Joshua Colp
On April 16, 2014, 9:23 p.m., Jonathan Rose wrote:
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that changes were
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- Joshua Colp
On April 17, 2014, 8:52
/3466/#comment21449
The other use of this uses tcp - is there any difference between that and
tls? Do we need to choose one or the other based on the connection?
I think this deprecates the change Scott originally did when closing the
connection.
- Joshua Colp
On April 18, 2014, 2:41 p.m
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- Joshua Colp
On April 18, 2014, 3:42
it is stopped.
Diffs
-
/branches/12/res/res_pjsip_sdp_rtp.c 412900
Diff: https://reviewboard.asterisk.org/r/3472/diff/
Testing
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confirmed fixed.
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and confirmed happy (after fixes were applied).
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On April 22, 2014, 7:50
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- Joshua Colp
On April 23, 2014, 7:28
://reviewboard.asterisk.org/r/3478/diff/
Testing
---
Ran test and confirmed failed on normal 12. Applied change. Re-ran test and
confirmed fixed.
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on here.
When you do a sip reload any realtime peers in chan_sip are pruned and
removed. This is probably causing the device state to change to an incorrect
value which app_queue is then using. The change probably works around this by
re-checking using any channels that exist.
- Joshua Colp
that format around, in which case why not just
add/set in the same API call?
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.
Diffs
-
/branches/12/channels/chan_pjsip.c 413007
Diff: https://reviewboard.asterisk.org/r/3480/diff/
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be just going crazy however)
For getting them you use ast_format_cap_get_framing.
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accessed there.
The question becomes... how do we notify the RTP side of stuff for
*setting up* the smoother. Since format capabilities are reference
counted we could pass in the same structure as the channel
nativeformats, and have it access that.
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://reviewboard.asterisk.org/r/3481/#comment21584
I'd remove this or change the message - otherwise you'll get it twice.
- Joshua Colp
On April 28, 2014, 6:18 p.m., opticron wrote:
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On April 29, 2014, 12:28 p.m., Joshua Colp wrote:
branches/11/res/res_http_websocket.c, line 195
https://reviewboard.asterisk.org/r/3481/diff/2/?file=58003#file58003line195
I don't see a close_sent change in here.
opticron wrote:
close_sent is set to 1 on line 206.
I meant
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On April 29, 2014, 1:48
:06 a.m., Joshua Colp wrote:
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