If the res_calendar module was followed immediately by one of the
calendar tech modules and "core stop gracefully" was run, Asterisk
would crash.
This patch adds use count tracking for res_calendar so that it is
unloaded after the tech modules when shutting down gracefully. It
is now not possible to unload all the of the calendar modules via
"module unload res_calednar.so", but it is still possible to unload
them all via "module unload -h res_calendar.so".
Review: https://reviewboard.asterisk.org/r/1752/
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r356215 | mjordan | 2012-02-22 08:53:53 -0600 (Wed, 22 Feb 2012) | 32 lines
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r356214 | mjordan | 2012-02-22 08:50:20 -0600 (Wed, 22 Feb 2012) | 27 lines
Fix potential buffer overrun and memory leak when executing "sip show peers"
The "sip show peers" command uses a fix sized array to sort the current peers
in the peers ao2_container. The size of the array is based on the current
number of peers in the container. However, once the size of the array is
determined, the number of peers in the container can change, as the peers
container is not locked. This could cause a buffer overrun when populating
the array, if peers were added to the container after the array was created.
Additionally, a memory leak of the allocated array would occur if a user
caused the _show_peers method to return CLI_SHOWUSAGE.
We now create a snapshot of the current peers using an ao2_callback with the
OBJ_MULTIPLE flag. This size of the array is set to the number of peers
that the iterator will iterate over; hence, if peers are added or removed
from the peers container it will not affect the execution of the "sip show
peers" command.
Review: https://reviewboard.asterisk.org/r/1738/
(closes issue ASTERISK-19231)
(closes issue ASTERISK-19361)
Reported by: Thomas Arimont, Jamuel Starkey
Tested by: Thomas Arimont, Jamuel Starkey
Patches: sip_show_peers_2012_02_16.diff uploaded by mjordan (license 6283)
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The accessors names for the "emulate_dtmf_digit" field on the ast_channel
are misleading. Change them to ast_channel_dtmf_digit_to_emulate*.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356183 65c4cc65-6c06-0410-ace0-fbb531ad65f3
IAX2 uses the trunkfreq variable to determine how often to send trunk packets, but
this value is in milliseconds while ast_timer_set_rate() expects the rate argument
to be ticks per second. So we divide 1000 by trunkfreq and pass that in instead.
With a default of 20ms, this change makes IAX2 send trunk packets every 20ms
instead of every 50ms.
Tracked down by myself and Bob Wienholt.
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This fixes two main issues:
1. Asterisk would send a CANCEL to the route created by the provisional response
instead of using the same destination it did in the initial INVITE.
2. If a new route set arrives in a 200 OK than was in the 1XX response (perfectly
possible if our outbound INVITE gets forked), then the route set in the 200 OK
needs to overwrite the route set in the 1XX response.
(closes issue ASTERISK-19358)
Reported by: Karsten Wemheuer
Tested by: Karsten Wemheuer
patches:
ASTERISK-19358.patch uploaded by Mark Michelson (license 5049)
ASTERISK-19358.patch uploaded by Stefan Schmidt (license 6034)
Review: https://reviewboard.asterisk.org/r/1749
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The principle difference between libvorbisfile v1.1.2 and newer (at least
v1.2.0) is the addition of the predefined callbacks OV_CALLBACKS_xxx in
vorbis/vorbisfile.h used for ov_open_callbacks().
* Updated the configure script to detect if libvorbisfile.h declares
OV_CALLBACKS_NOCLOSE.
* Copied the declaration of OV_CALLBACKS_NOCLOSE from v1.2.0 to allow
v1.1.2 to compile.
(closes issue ASTERISK-19370)
Reported by: Jonn Taylor
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This change permits each verbose destination (consoles, logger) to have its
own concept of what the verbosity level is. The big feature here is that
the logger will now be able to capture a particular verbosity level without
condemning each console to need to suffer that level of verbosity.
Additionally, a stray 'core set verbose' will no longer change what will go
to the log.
Review: https://reviewboard.asterisk.org/r/1599/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355413 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Ogg/vorbis was fairly useless as a voicemail file format because it did
not implement the seek and tell format callbacks among other problems.
Since we were already using the libvorbis and libvorbisenc libraries we
can use libvorbisfile as it is also part of the vorbis library package.
* Made use the libvorbisfile to handle the ogg/vorbis file stream. The
format_ogg_vorbis.c is now mostly a wrapper around libvorbisfile.
(closes issue ASTERISK-16926)
Reported by: sque
Patches:
ogg_vorbis_use_libvorbisfile.patch (license #6108) patch uploaded by sque
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This patch adds a variable AGIEXITONHANGUP for res_agi. If this variable is
set to "yes" on a channel, AGI() will exit immediately once a channel hangup
has been detected. This was the behavior of AGI() in Asterisk 1.4 and earlier
and is still desired by some people.
Review: https://reviewboard.asterisk.org/r/1734/
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There can only be one database connection in res_config_pgsql just like
res_config_sqlite. If the connection is lost, the connection may not get
reestablished to the same database if the res_pgsql.conf and
extconfig.conf files are inconsistent.
* Made only use the configured database from res_pgsql.conf.
* Fixed potential buffer overwrite of last[] in config_pgsql().
(closes issue ASTERISK-16982)
Reported by: german aracil boned
Review: https://reviewboard.asterisk.org/r/1731/
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This simply fixes the compilation issue introduced in r354429 by
re-adding the 'quote' variable.
(closes issue ASTERISK-19337)
Reported by: John Taylor
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354799 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The config parser in Asterisk does not currently remove a backslash that is
used to escape a semicolon which would otherwise be interpreted as the start
of a comment.
The change here causes that backslash to be removed, but does not create a
real escape system in the config parser. The biggest complication with a real
escape system would be breaking existing configs everywhere (parsing \\ as \
and breaking on escaped non-semicolon characters) even though it would be the
"right" way to do things.
(closes issue ASTERISK-17121)
Review: https://reviewboard.asterisk.org/r/1724/
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This patch adds the auto_force_rport and auto_comedia NAT options. It
also converts the nat= setting to a list of comma-separated combinable
options: no, force_rport, comedia, auto_force_rport, and auto_comedia.
nat=yes remains as an undocumented option equal to
"force_rport,comedia". The first instance of 'yes' or 'no' in the list
stops parsing and overrides any previously set options. If an auto_*
option is specified with its non-auto_ counterpart, the auto setting
takes precedence.
This patch builds upon the patch posted to ASTERISK-17860 by JIRA user
pedro-garcia.
(closes issue ASTERISK-17860)
Review: https://reviewboard.asterisk.org/r/1698/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354597 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In ASTERISK-18924, SIP INFO DTMF handlingw as changed to account for both
lowercase alphatbetic DTMF events, as well as uppercase alphabetic DTMF
events. When this occurred, the comparison of the character buffer containing
the event code was changed such that the buffer was first compared again '0'
and '9' to determine if it was numeric. Unfortunately, since the first
character in the buffer will typically be '1' in the case of non-numeric
event codes (10-16), this caused those codes to be converted to a DTMF event
of '1'. This patch fixes that, and cleans up handling of both
application/dtmf-relay and application/dtmf content types.
Review: https://reviewboard.asterisk.org/r/1722/
(closes issue ASTERISK-19290)
Reported by: Ira Emus
Tested by: mjordan
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