[asterisk-dev] [Code Review] 3422: testsuite: Fix Asterisk shutdown timeout in chan_sip session_timer tests by hanging up channels
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3422/ --- Review request for Asterisk Developers. Bugs: ASTERISK-23591 https://issues.asterisk.org/jira/browse/ASTERISK-23591 Repository: testsuite Description --- 11 of the chan_sip session_timer tests fail to hangup channels from sipp. This causes graceful shutdown to timeout, and asterisk is killed. Fixing this reduces overall test time by 2 minutes, more if the asterisk shutdown timeout is increased. Before anyone asks: yes, the added XML is identical on every single test. Most of the files were already identical except a line or two, the goal of this change is to fix the timeout's. Diffs - /asterisk/trunk/tests/channels/SIP/session_timers/uas_originate/small_minse_no_se/sipp/uac-session-timer.xml 4894 /asterisk/trunk/tests/channels/SIP/session_timers/uas_originate/small_minse_medium_se/sipp/uac-session-timer.xml 4894 /asterisk/trunk/tests/channels/SIP/session_timers/uas_originate/small_minse_large_se/sipp/uac-session-timer.xml 4894 /asterisk/trunk/tests/channels/SIP/session_timers/uas_originate/no_minse_no_se/sipp/uac-session-timer.xml 4894 /asterisk/trunk/tests/channels/SIP/session_timers/uas_originate/no_minse_medium_se/sipp/uac-session-timer.xml 4894 /asterisk/trunk/tests/channels/SIP/session_timers/uas_originate/no_minse_large_se/sipp/uac-session-timer.xml 4894 /asterisk/trunk/tests/channels/SIP/session_timers/uas_originate/medium_minse_no_se/sipp/uac-session-timer.xml 4894 /asterisk/trunk/tests/channels/SIP/session_timers/uas_originate/medium_minse_medium_se/sipp/uac-session-timer.xml 4894 /asterisk/trunk/tests/channels/SIP/session_timers/uas_originate/medium_minse_large_se/sipp/uac-session-timer.xml 4894 /asterisk/trunk/tests/channels/SIP/session_timers/uas_originate/large_minse_no_se/sipp/uac-session-timer.xml 4894 /asterisk/trunk/tests/channels/SIP/session_timers/uas_originate/large_minse_large_se/sipp/uac-session-timer.xml 4894 Diff: https://reviewboard.asterisk.org/r/3422/diff/ Testing --- Testing was done with patch from ASTERISK-23369 so that asterisk shutdown timeout caused failures. Retested with this patch, no failures in tests/channels/SIP/session_timers. Thanks, Corey Farrell -- _ -- 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] [svn-commits] rmudgett: branch 1.8 r411715 - in /branches/1.8: ./ channels/ configs/ includ...
On Sat, Apr 5, 2014 at 4:46 PM, Paul Belanger wrote: > On Sat, Apr 5, 2014 at 2:58 AM, Olle E. Johansson wrote: > > > > On 04 Apr 2014, at 20:32, SVN commits to the Digium repositories < > svn-comm...@lists.digium.com> wrote: > > > >> - case 'I': > >> - ast_set_flag(&ast_options, > AST_OPT_FLAG_INTERNAL_TIMING); > >> - break; > > > > Just checking... I would rather add a NOTICE log here that "i" is not > needed any more. Please make sure that configurations starting with "- i" > will not suddenly fail. > > > I agree with Olle here, this seems to be a massive change mid-release. > Removing a command-line option is certainly going to break some > peoples boxes. Why not a deprecated warning and then removal from > trunk to give people time to react? > FWIW, specifying this command line option or asterisk.conf option, even after it has been removed, should be fine. It will just be ignored and no new warnings will be generated, AFAICT. On the surface, this looks like a change that shouldn't be made in a release branch. However, this really is an option that should have never existed. It's never the right thing to turn it off. The change to make it the only way it works is really the right thing to do. It was equivalent to an option called "make_things_work_properly=yes". -- Russell Bryant -- _ -- 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] [svn-commits] rmudgett: branch 1.8 r411715 - in /branches/1.8: ./ channels/ configs/ includ...
On Sat, Apr 5, 2014 at 2:58 AM, Olle E. Johansson wrote: > > On 04 Apr 2014, at 20:32, SVN commits to the Digium repositories > wrote: > >> - case 'I': >> - ast_set_flag(&ast_options, >> AST_OPT_FLAG_INTERNAL_TIMING); >> - break; > > Just checking... I would rather add a NOTICE log here that "i" is not needed > any more. Please make sure that configurations starting with "- i" will not > suddenly fail. > I agree with Olle here, this seems to be a massive change mid-release. Removing a command-line option is certainly going to break some peoples boxes. Why not a deprecated warning and then removal from trunk to give people time to react? -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] [Code Review] 3362: func_periodic_hook: New function for periodic hooks.
---
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/3362/
---
(Updated April 5, 2014, 8:06 a.m.)
Status
--
This change has been marked as submitted.
Review request for Asterisk Developers.
Changes
---
Committed in revision 411768
Repository: Asterisk
Description
---
This commit introduces a new dialplan function, PERIODIC_HOOK().
It allows you run to a dialplan hook on a channel periodically. The
original use case that inspired this was the ability to play a beep
periodically into a call being recorded. The implementation is much
more generic though and could be used for many other things.
The implementation makes heavy use of existing Asterisk components.
It uses a combination of Local channels and ChanSpy() to run some
custom dialplan and inject any audio it generates into an active call.
The other important bit of the implementation is how it figures out
when to trigger the beep playback. This implementation uses the
audiohook API, even though it's not actually touching the audio in any
way. It's a convenient way to get a callback and check if it's time
to kick off another beep. It would be nice if this was timer event
based instead of polling based, but unfortunately I don't see a way to
do it that won't interfere with other things.
Diffs
-
/trunk/funcs/func_periodic_hook.c PRE-CREATION
/trunk/CHANGES 411684
Diff: https://reviewboard.asterisk.org/r/3362/diff/
Testing
---
Called the following extension (100@test), both letting it run all the way
through, as well as hanging up at various points in the middle.
[hooks]
exten => beep,1,Answer()
same => n,Verbose(1,Channel name: ${HOOK_CHANNEL})
same => n,Verbose(1,Hook ID: ${HOOK_ID})
same => n,Playback(beep)
[test]
exten => 100,1,Answer()
same => n,Set(BEEP_ID=${PERIODIC_HOOK(hooks,beep,5)})
same => n,Wait(20)
same => n,Set(PERIODIC_HOOK(${BEEP_ID})=off)
same => n,Wait(20)
same => n,Set(PERIODIC_HOOK(${BEEP_ID})=on)
same => n,Wait(20)
same => n,Hangup()
Thanks,
Russell Bryant
--
_
-- 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] [Code Review] 3417: Add AMI events for all device state and presence state changes
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3417/#review11505 --- I would like to see a configuration option for this, as it will generate a massive amount of events in busy servers. - Olle E Johansson On April 4, 2014, 9:38 p.m., Mark Michelson wrote: > > --- > This is an automatically generated e-mail. To reply, visit: > https://reviewboard.asterisk.org/r/3417/ > --- > > (Updated April 4, 2014, 9:38 p.m.) > > > Review request for Asterisk Developers. > > > Repository: Asterisk > > > Description > --- > > AMI does not emit events when device state or presence state changes. The > closest things that exist currently are the ExtenstionStatus and > PresenceStatus events, which inform about device state and presence state > events as they pertain to hints in the dialplan. These new events are raised > for every device state change or presence state change in Asterisk. > > > Diffs > - > > /trunk/main/presencestate.c 411714 > /trunk/main/manager.c 411714 > /trunk/main/devicestate.c 411714 > /trunk/include/asterisk/presencestate.h 411714 > /trunk/include/asterisk/devicestate.h 411714 > > Diff: https://reviewboard.asterisk.org/r/3417/diff/ > > > Testing > --- > > See /r/3418 > > > 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
