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03beadb6e9
The masquerade supertest frequently fails because either the local channel chain doesn't completely optimize out or the DTMF handshake doesn't completely get accross. Local channel optimization requires frames flowing to trigger when optimization can happen. When optimization happens the media frame that triggered the optimization is dropped. Sending DTMF requires frames to flow in the other direction for timing purposes while sending nothing. If internal timing is not enabled when MOH is playing, Asterisk switches to received timing when an audio frame is received. With optimization dropping media frames and MOH not sending frames unless it receives frames, occasionaly there are no more frames being passed and the test fails. * The asterisk command line -I option and the asterisk.conf internal_timing option are removed. Asterisk now always uses internal timing when needed if any timing module is loaded. The issue ASTERISK-14861 did this quite awhile ago in v1.4 but effectively is broken if other internal timing modules besides DAHDI are used. The ast_read_generator_actions() now only does received timing if it has no choice for frame generators like MOH, silence, and playback streaming. * Cleaned up some code dealing with frame generators in ast_deactivate_generator(), generator_write_format_change(), ast_activate_generator(), and ast_channel_stop_silence_generator(). * Removed ast_internal_timing_enabled(), AST_OPT_FLAG_INTERNAL_TIMING, and ast_opt_internal_timing. ASTERISK-22846 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3414/ ........ Merged revisions 411715 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 411716 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 411717 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411724 65c4cc65-6c06-0410-ace0-fbb531ad65f3
103 lines
4.1 KiB
Plaintext
103 lines
4.1 KiB
Plaintext
[directories](!)
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astetcdir => /etc/asterisk
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astmoddir => /usr/lib/asterisk/modules
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astvarlibdir => /var/lib/asterisk
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astdbdir => /var/lib/asterisk
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astkeydir => /var/lib/asterisk
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astdatadir => /var/lib/asterisk
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astagidir => /var/lib/asterisk/agi-bin
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astspooldir => /var/spool/asterisk
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astrundir => /var/run/asterisk
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astlogdir => /var/log/asterisk
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astsbindir => /usr/sbin
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[options]
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;verbose = 3
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;debug = 3
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;alwaysfork = yes ; Same as -F at startup.
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;nofork = yes ; Same as -f at startup.
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;quiet = yes ; Same as -q at startup.
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;timestamp = yes ; Same as -T at startup.
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;execincludes = yes ; Support #exec in config files.
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;console = yes ; Run as console (same as -c at startup).
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;highpriority = yes ; Run realtime priority (same as -p at
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; startup).
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;initcrypto = yes ; Initialize crypto keys (same as -i at
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; startup).
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;nocolor = yes ; Disable console colors.
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;dontwarn = yes ; Disable some warnings.
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;dumpcore = yes ; Dump core on crash (same as -g at startup).
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;languageprefix = yes ; Use the new sound prefix path syntax.
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;systemname = my_system_name ; Prefix uniqueid with a system name for
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; Global uniqueness issues.
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;autosystemname = yes ; Automatically set systemname to hostname,
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; uses 'localhost' on failure, or systemname if
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; set.
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;mindtmfduration = 80 ; Set minimum DTMF duration in ms (default 80 ms)
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; If we get shorter DTMF messages, these will be
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; changed to the minimum duration
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;maxcalls = 10 ; Maximum amount of calls allowed.
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;maxload = 0.9 ; Asterisk stops accepting new calls if the
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; load average exceed this limit.
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;maxfiles = 1000 ; Maximum amount of openfiles.
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;minmemfree = 1 ; In MBs, Asterisk stops accepting new calls if
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; the amount of free memory falls below this
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; watermark.
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;cache_record_files = yes ; Cache recorded sound files to another
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; directory during recording.
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;record_cache_dir = /tmp ; Specify cache directory (used in conjunction
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; with cache_record_files).
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;transmit_silence = yes ; Transmit silence while a channel is in a
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; waiting state, a recording only state, or
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; when DTMF is being generated. Note that the
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; silence internally is generated in raw signed
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; linear format. This means that it must be
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; transcoded into the native format of the
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; channel before it can be sent to the device.
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; It is for this reason that this is optional,
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; as it may result in requiring a temporary
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; codec translation path for a channel that may
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; not otherwise require one.
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;transcode_via_sln = yes ; Build transcode paths via SLINEAR, instead of
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; directly.
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;runuser = asterisk ; The user to run as.
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;rungroup = asterisk ; The group to run as.
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;lightbackground = yes ; If your terminal is set for a light-colored
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; background.
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;forceblackbackground = yes ; Force the background of the terminal to be
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; black, in order for terminal colors to show
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; up properly.
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;defaultlanguage = en ; Default language
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documentation_language = en_US ; Set the language you want documentation
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; displayed in. Value is in the same format as
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; locale names.
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;hideconnect = yes ; Hide messages displayed when a remote console
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; connects and disconnects.
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;lockconfdir = no ; Protect the directory containing the
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; configuration files (/etc/asterisk) with a
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; lock.
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;stdexten = gosub ; How to invoke the extensions.conf stdexten.
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; macro - Invoke the stdexten using a macro as
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; done by legacy Asterisk versions.
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; gosub - Invoke the stdexten using a gosub as
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; documented in extensions.conf.sample.
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; Default gosub.
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;live_dangerously = no ; Enable the execution of 'dangerous' dialplan
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; functions from external sources (AMI,
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; etc.) These functions (such as SHELL) are
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; considered dangerous because they can allow
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; privilege escalation.
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; Default no
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; Changing the following lines may compromise your security.
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;[files]
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;astctlpermissions = 0660
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;astctlowner = root
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;astctlgroup = apache
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;astctl = asterisk.ctl
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[compat]
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pbx_realtime=1.6
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res_agi=1.6
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app_set=1.6
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