Re: [asterisk-dev] [Code Review] 3069: Fix deadlock experienced during multi-party PJSIP transfer through masquerade rework

id pass without the patch. I found none on my machine, though I'll admit I did not run every test. Thanks, Mark Michelson -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list

Re: [asterisk-dev] [Code Review] 3069: Fix deadlock experienced during multi-party PJSIP transfer through masquerade rework

finish before actually > running the fixup. This makes it so the fixup is run without having to push a > task into a serializer at all. > > > Diffs > - > > /branches/12/res/res_pjsip.c 403349 >

Re: [asterisk-dev] [Code Review] 3070: bridges: Add two new properties to bridges and bridge snapshots - the name of a creator and the name the creator uses to refer to that bridge

7;d suggest populating the field with something like "None" to indicate that there is intentionally no name and/or creator in these situations. - Mark Michelson On Dec. 13, 2013, 7:54 p.m., Jonathan Rose wrote: > > --- > This i

Re: [asterisk-dev] [Code Review] 3046: framehooks: Re-iterate when frame is changed

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3046/#review10411 --- Ship it! Ship It! - Mark Michelson On Dec. 13, 2013, 4:28

Re: [asterisk-dev] [Code Review] 3069: Fix deadlock experienced during multi-party PJSIP transfer through masquerade rework

freely do what it needs to. I think such refactoring could actually be done, but I also feel like since that sort of refactoring does not directly affect the problem being fixed here, it should be saved for a separate review. - Mark Michelson On Dec. 12, 2013, 9:59 p.m., Mark Michelson wrote

[asterisk-dev] [Code Review] 3069: Fix deadlock experienced during multi-party PJSIP transfer through masquerade rework

ock occur, and with this patch it no longer occurs. I also ran a bevy of testsuite tests to see if there were any that did not pass with this patch but did pass without the patch. I found none on my machine, though I'll admit I did not run every test.

Re: [asterisk-dev] [Code Review] 3066: bridge_native_rtp: Deadlock during 4-way conference creation

, it turned out I was wrong instead :) The only suggestion I have for this is that since RTP glue update_peer() method is now consistently called with the channel locked, I would update its documentation in rtp_engine.h to note this. - Mark Michelson On Dec. 11, 2013, 11:20 p.m., Kevin Harwell

Re: [asterisk-dev] [Code Review] 3050: PJSIP: Add Path header support

> On Dec. 11, 2013, 3:22 p.m., Matt Jordan wrote: > > /trunk/res/res_pjsip_path.c, lines 105-126 > > > > > > Because this is using an ao2_callback to get the contact, it's going to > > be rather expensive. If we we

Re: [asterisk-dev] [Code Review] 3061: External MWI core support with AMI using it.

altered to expect a single mailbox string instead. - Mark Michelson On Dec. 9, 2013, 9:51 p.m., rmudgett wrote: > > --- > This is an automatically generated e-mail. To reply, visit: > https://reviewboard.asteris

Re: [asterisk-dev] [Code Review] 3034: Tests for PJSIP_ENDPOINT

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3034/#review10376 --- Ship it! Ship It! - Mark Michelson On Nov. 28, 2013, 4:31

Re: [asterisk-dev] [Code Review] 3035: Add a function PJSIP_ENDPOINT to retrieve endpoint configuration details from the dialplan

tps://reviewboard.asterisk.org/r/3035/#comment19771> Just curious, but why does "disallow" get this special treatment? - Mark Michelson On Dec. 10, 2013, 2:34 a.m., Matt Jordan wrote: > > --- > This is an automatic

Re: [asterisk-dev] [Code Review] 3060: pjsip tests: interactions with chan_sip cause test failures

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3060/#review10367 --- Ship it! Ship It! - Mark Michelson On Dec. 9, 2013, 10:52

Re: [asterisk-dev] [Code Review] 3038: Add CHANNEL function support for PJSIP

, this review deserves a ship it! really. - Mark Michelson On Dec. 10, 2013, 3:51 a.m., Matt Jordan wrote: > > --- > This is an automatically generated e-mail. To reply, visit: > https://reviewboard.asteris

Re: [asterisk-dev] [Code Review] 3038: Add CHANNEL function support for PJSIP

our-space tabs and lined up the arguments just past the opening paren on the preceding line, a la PEP-8 recommendations. However, such practice doesn't work well when dealing with tabs instead of spaces for indention. This applies to some other function invocations used in this funct

Re: [asterisk-dev] [Code Review] 3038: Add CHANNEL function support for PJSIP

> 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 ou

Re: [asterisk-dev] [Code Review] 3038: Add CHANNEL function support for PJSIP

> On Dec. 4, 2013, 9:02 p.m., Mark Michelson wrote: > > /branches/12/channels/pjsip/dialplan_functions.c, lines 338-345 > > <https://reviewboard.asterisk.org/r/3038/diff/1/?file=48949#file48949line338> > > > > These descriptions are inaccurate when used on ou

Re: [asterisk-dev] [Code Review] 3038: Add CHANNEL function support for PJSIP

> 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 ou

Re: [asterisk-dev] [Code Review] 3060: pjsip tests: interactions with chan_sip cause test failures

conflict does not apply for this test. The other question I have is if you plan to go through the tests that use chan_sip and chan_pjsip and add the appropriate deps to the properties in their test-config.yaml files. Were you holding off on that in case this approach was vetoed? - Mark Michelson

Re: [asterisk-dev] [Code Review] 3046: framehooks: Re-iterate when frame is changed

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 AST_FRAME_VOICE. While I don't think this will loop infinitely, it may result in the jitter buffer being depleted. -

Re: [asterisk-dev] [Code Review] 2944: PJSIP messaging: send message to URI.

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/2944/#review10338 --- Ship it! Ship It! - Mark Michelson On Dec. 9, 2013, 4:17

Re: [asterisk-dev] [Code Review] 3015: Testsuite: CONFBRIDGE_RESULT test

N /asterisk/trunk/lib/python/asterisk/apptest.py 4331 Diff: https://reviewboard.asterisk.org/r/3015/diff/ Testing --- Ran the test in a while loop and ensured that in 30+ executions, the test passed each time. Tha

Re: [asterisk-dev] [Code Review] 3009: Add channel variable that describes why a channel left a ConfBridge()

iff: https://reviewboard.asterisk.org/r/3009/diff/ Testing --- See review 3015 Thanks, Mark Michelson -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update opt

Re: [asterisk-dev] [Code Review] 3024: Testsuite: Test the MixMonitor 'f' option

tests.yaml 4331 /asterisk/trunk/tests/apps/mixmonitor_func/test-config.yaml PRE-CREATION /asterisk/trunk/tests/apps/mixmonitor_func/configs/ast1/extensions.conf PRE-CREATION Diff: https://reviewboard.asterisk.org/r/3024/diff/ Testing --- Test consistently passes. Thanks, Mark

Re: [asterisk-dev] [Code Review] 3023: Add MixMonitor() option to specify channel variable into which to store the recording filename

and 'i' options for MixMonitor, a user can easily manage multiple recordings on a single channel. Diffs - /trunk/apps/app_mixmonitor.c 402883 Diff: https://reviewboard.asterisk.org/r/3023/diff/ Testing --- See https://reviewboard.asterisk.org/r/3

Re: [asterisk-dev] [Code Review] 3044: Switch PJSIP auth to use a vector

pass. Thanks, Mark Michelson -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev

Re: [asterisk-dev] Asterisk Test Suite: Proposed Logging Change

Matthew Jordan Sunday, December 08, 2013 9:09 AM Back when we first added python logging to the Asterisk Test Suite, there were considerably fewer tests and the log files - while large - were slightly more manageable. Today, it feels like things have gotten a bit more c

Re: [asterisk-dev] [Code Review] 3053: testsuite: Test predial handlers for app_page

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3053/#review10326 --- Ship it! Ship It! - Mark Michelson On Dec. 6, 2013, 9:01

Re: [asterisk-dev] [Code Review] 3055: pbx.c: add lock around ast_exten use to prevent memory corruption

rous operation is not surrounded by the lock. - Mark Michelson On Dec. 6, 2013, 9:41 p.m., Scott Griepentrog wrote: > > --- > This is an automatically generated e-mail. To reply, visit:

Re: [asterisk-dev] [Code Review] 3053: testsuite: Test predial handlers for app_page

rk of review board or you may have purposely not included the old test since it is not relevant, but since it's not here I just wanted to make sure it hadn't disappeared. - Mark Michelson On Dec. 5, 2013, 11:36 p.m., Jonathan Rose wrote: > > ---

Re: [asterisk-dev] [Code Review] 3045: app_page: Add predial handlers

tps://reviewboard.asterisk.org/r/3045/#comment19721> Check for a NULL return here. This will go badly if app_stack.so is not loaded. - Mark Michelson On Dec. 5, 2013, midnight, Jonathan Rose wrote: > > --- > This is an automatically generated e

Re: [asterisk-dev] Presence State in Asterisk 11

Jason Parker Thursday, December 05, 2013 3:31 PM Presence State is a feature that was added in Asterisk 11, which allows for more information to be captured about the state of a user.  There appears to have been a rather large oversight in the i

Re: [asterisk-dev] [Code Review] 2944: PJSIP messaging: send message to URI.

Since cfg is scoped to this function, once this function returns, cfg->default_outbound_endpoint cannot be assumed safe to use since cfg (and its contents) may get freed. - Mark Michelson On Dec. 5, 2013, 8:26 p.m.,

Re: [asterisk-dev] [Code Review] 3037: Add tests for CHANNEL function for PJSIP

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3037/#review10300 --- Ship it! Ship It! - Mark Michelson On Dec. 1, 2013, 2:12

[asterisk-dev] [Code Review] 3044: Switch PJSIP auth to use a vector

https://reviewboard.asterisk.org/r/3044/diff/ Testing --- Ran inbound/outbound call and outbound registration tests in the testsuite to ensure they still pass. Thanks, Mark Michelson -- _ -- Bandwidth and Colocation Provid

Re: [asterisk-dev] [Code Review] 3038: Add CHANNEL function support for PJSIP

ces throughout this file. /branches/12/channels/pjsip/dialplan_functions.c <https://reviewboard.asterisk.org/r/3038/#comment19673> Just use ast_copy_string() /branches/12/channels/pjsip/dialplan_functions.c <https://reviewboard.asterisk.org/r/3038/#comment19674> Print a warning about

Re: [asterisk-dev] [Code Review] 3041: app_record: Add an option that allows DTMF '0' to act as an additional terminator

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3041/#review10298 --- Ship it! Ship It! - Mark Michelson On Dec. 3, 2013, 11:04

Re: [asterisk-dev] [Code Review] 2961: Add channel locking for calls that result in channel snapshot creation

est in a loop. After approximately 450 test runs, there were no negative consequences. Prior to this changeset, running the test 20-50 times would result in a crash. Thanks, Mark Michelson -- _ -- Bandwidth and Colocation Pro

Re: [asterisk-dev] [Code Review] 3040: bridge transfers: Make sure ATTENDEDTRANSFER variable gets set for all the expected channels when doing bridge attended transfers. Also make sure BLINDTRANSFER i

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3040/#review10296 --- Ship it! Ship It! - Mark Michelson On Dec. 2, 2013, 9:22

Re: [asterisk-dev] WebRTC over SRTP-DTLS

at configs/sip.conf.sample in Asterisk 11 and grep for "DTLS-SRTP CONFIGURATION". That will direct you to a section that explains the various DTLS-related configuration options for chan_sip. Mark Michelson -- _ -- Ban

Re: [asterisk-dev] [Code Review] 3027: Valgrind support in TestSuite

/sgriepentrog/testsuite-valgrind/runtests.py <https://reviewboard.asterisk.org/r/3027/#comment19658> I think you left a word out in this log message. - Mark Michelson On Nov. 25, 2013, 11:10 p.m., Scott Griepentrog wrote: > > --- > T

Re: [asterisk-dev] Separate Audio/Video

. Asterisk sends the audio and video to server B (as specified by the media_address option). Server B runs a program that redirects the audio to device C and the video to device D. Note that Asterisk and server B could presumably be the same server, if you chose. Mark

Re: [asterisk-dev] [Code Review] 3028: ari: Add 'number', 'digits', and 'characters' URI scheme playback implementations

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3028/#review10289 --- Ship it! Ship It! - Mark Michelson On Nov. 26, 2013, 8:04

Re: [asterisk-dev] [Code Review] 2963: chan_pjsip: Extend redirect handling support

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/2963/#review10288 --- Ship it! Ship It! - Mark Michelson On Nov. 16, 2013, 4:09

Re: [asterisk-dev] [Code Review] 3028: ari: Add 'number', 'digits', and 'characters' URI scheme playback implementations

tps://reviewboard.asterisk.org/r/3028/#comment19657> Any particular reason for this initialization being added? - Mark Michelson On Nov. 23, 2013, 2:35 p.m., Joshua Colp wrote: > > --- > This is an automatically generated e-mail.

Re: [asterisk-dev] [Code Review] 2961: Add channel locking for calls that result in channel snapshot creation

eway_nominal test in a loop. After approximately 450 test runs, there were no negative consequences. Prior to this changeset, running the test 20-50 times would result in a crash. Thanks, Mark Michelson -- _ -- Bandwidth and Co

Re: [asterisk-dev] [Code Review] 3024: Testsuite: Test the MixMonitor 'f' option

isk/trunk/tests/apps/tests.yaml 4331 /asterisk/trunk/tests/apps/mixmonitor_func/test-config.yaml PRE-CREATION /asterisk/trunk/tests/apps/mixmonitor_func/configs/ast1/extensions.conf PRE-CREATION Diff: https://reviewboard.asterisk.org/r/3024/diff/ Testing --- Test consistently passes. Tha

Re: [asterisk-dev] [Code Review] 3023: Add MixMonitor() option to specify channel variable into which to store the recording filename

'f' and 'i' options for MixMonitor, a user can easily manage multiple recordings on a single channel. Diffs (updated) - /trunk/apps/app_mixmonitor.c 402883 Diff: https://reviewboard.asterisk.org/r/3023/diff/ Testing --- See https://reviewboard.asterisk.org/r/3

Re: [asterisk-dev] Separate Audio/Video

matically send your audio to the destination in the audio section and your video to the destination in the video section. Even if you send the audio and video to the same IP address, with RTP the audio and video will be sent to different ports, so in a way they already are separat

Re: [asterisk-dev] [Code Review] 2961: Add channel locking for calls that result in channel snapshot creation

tps://reviewboard.asterisk.org/r/2961/#review10273 ------- On Nov. 19, 2013, 11:17 p.m., Mark Michelson wrote: > > --- > This is an automatically generated e-mail. To reply, visit: > https://reviewboard.asterisk.org/r/2961/ >

Re: [asterisk-dev] [Code Review] 3025: ARI: Implement device state API

lling ao2_find(device_state_subscriptions, name, OBJ_SEARCH_KEY | OBJ_UNLINK | OBJ_NODATA); - Mark Michelson On Nov. 21, 2013, 7:12 p.m., Kevin Harwell wrote: > > --- > This is an automatically generated e-mail. To reply, visit: > htt

Re: [asterisk-dev] [Code Review] 2959: pjsip: AMI commands and events

tps://reviewboard.asterisk.org/r/2959/#comment19578> Introduced a couple of red blobs here. - Mark Michelson On Nov. 15, 2013, 7:58 p.m., Kevin Harwell wrote: > > --- > This is an automatically generated e-mail. To reply,

Re: [asterisk-dev] [Code Review] 3023: Add MixMonitor() option to specify channel variable into which to store the recording filename

ated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3023/#review10242 --- On Nov. 20, 2013, 9:12 p.m., Mark Michelson wrote: > > --- > This is an

Re: [asterisk-dev] [Code Review] 3024: Testsuite: Test the MixMonitor 'f' option

automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3024/#review10236 --- On Nov. 20, 2013, 9:12 p.m., Mark Michelson wrote: > > --- > This i

Re: [asterisk-dev] [Code Review] 3024: Testsuite: Test the MixMonitor 'f' option

> 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 determ

Re: [asterisk-dev] [Code Review] 3015: Testsuite: CONFBRIDGE_RESULT test

y 4331 Diff: https://reviewboard.asterisk.org/r/3015/diff/ Testing --- Ran the test in a while loop and ensured that in 30+ executions, the test passed each time. Thanks, Mark Michelson -- _ -- Bandwidth and Colocation Provid

Re: [asterisk-dev] [Code Review] 3023: Add MixMonitor() option to specify channel variable into which to store the recording filename

> assumptions by overwriting the value set from previous invocations of > MixMonitor. By using the 'f' and 'i' options for MixMonitor, a user can > easily manage multiple recordings on a single channel. > > > Diffs > - > > /t

Re: [asterisk-dev] devicestate

Johansson Olle E wrote: > 16 jan 2008 kl. 15.56 skrev Atis Lezdins: > >> On 1/16/08, Leif Madsen <[EMAIL PROTECTED]> wrote: I agree for real devices. However i wonder - why i can't change state for Local channels. >>> Funny enough, I had this same issue today within Queu

[asterisk-dev] Call Forwarding Loop branch - Ready to merge?

respond here. Mark Michelson ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev

Re: [asterisk-dev] Dynamic queue members with penalty and rrmemory

ybody confirm? > > Regards, > Atis > If you're using a recent version of 1.4, then members are not kept in any particular order in memory since they are now kept in a hash table. Most likely, even though B logs in first, C and D appear earlier when ite

Re: [asterisk-dev] [asterisk-commits] qwell: trunk r82800 - in /trunk: apps/app_queue.c configs/queues.conf.sample

Mark Michelson wrote: > Kevin P. Fleming wrote: > >> SVN commits to the Asterisk project wrote: >> >> >> >>> + if (update_cdr && qe->chan->cdr) >>> + ast_copy_string(qe->chan->cdr-&g

Re: [asterisk-dev] [asterisk-commits] qwell: trunk r82800 - in /trunk: apps/app_queue.c configs/queues.conf.sample

Kevin P. Fleming wrote: > SVN commits to the Asterisk project wrote: > > >> +if (update_cdr && qe->chan->cdr) >> +ast_copy_string(qe->chan->cdr->dstchannel, >> member->membername, sizeof(qe->chan->cdr->dstchannel)); >> > > This is buggy; member->membername

Re: [asterisk-dev] app_queue membercount

ibly even just MEMBER) function is a good idea too, but it would be better for retrieving information about a specific member, like paused status, penalty, number of calls taken, etc. setinterfacevar already provides this information but having it available with a more uniform syntax would be

Re: [asterisk-dev] plea for more verbose commit messages (Re: [asterisk-commits] file: branch 1.4 r75712 - in /branches/1.4: apps/ channels/ pbx/)

Luigi Rizzo wrote: > On Wed, Jul 18, 2007 at 08:00:23PM -, SVN commits to the Asterisk project > wrote: > >> Author: file >> Date: Wed Jul 18 15:00:23 2007 >> New Revision: 75712 >> >> URL: http://svn.digium.com/view/asterisk?view=rev&rev=75712 >> Log: >> Backport GCC 4.2 fixes. Without t

Re: [asterisk-dev] [svn-commits] jrothenberger: branch jrothenberger/asterisk-urgent r71904 - /team/jrothenber...

Russell Bryant wrote: > SVN commits to the Digium repositories wrote: > >> Author: jrothenberger >> Date: Tue Jun 26 14:17:49 2007 >> New Revision: 71904 >> >> URL: http://svn.digium.com/view/asterisk?view=rev&rev=71904 >> Log: (empty) >> >> Added: >> team/jrothenberger/asterisk-urgent/ >>

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