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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@356291 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@356214 65c4cc65-6c06-0410-ace0-fbb531ad65f3
'p->outgoing' in chan_dahdi and sig_analog wern't kept in sync, particulary FXS ast_hangup didn't clear the 'outgoing' flag.
sig_pri and sig_ss7 were keeping 'outgoing' flag insync.
Now provides a callback for all the low level sig_XXX modules.
(issue ASTERISK-19316)
alecdavis (license 585)
Reported by: Jeremy Pepper
Tested by: alecdavis
Review: https://reviewboard.asterisk.org/r/1747/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@355850 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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@355746 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@355732 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When I made this change initially, I was under the false impression that the
audiohooks structure remained on the channel after all of the hooks had been
detached. This is not the case, ast ast_read takes care of removing the
audiohooks structure if the lists are empty.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@355622 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@355608 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Back in r646, TRUNK_CALL_START was added and defined as 0x4000. That same value
was also hard-coded in one part of the IAX2 code instead of using the #define.
TRUNK_CALL_START has changed over the years (for dealing with LOW_MEMORY), but
the hard-coded usage was never updated to match. This patch fixes that.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@355448 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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@355365 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This was causing identification that should have been
made private to be public.
(closes issue AST-814)
reported by Patrick Anderson
Patches:
chan_sip.c.diff uploaded by Patrick Anderson (license 5430)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@355268 65c4cc65-6c06-0410-ace0-fbb531ad65f3
send_apathetic_reply takes the incoming frame's source call number as the
destination call number for the outgoing frame. If the incoming frame was a
full frame, then the high order bit of the source call number is set and will be
interpreted as a retransmit when sent back out as the destination call number.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@355182 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Since the dir timestamp is available at one second resolution, we cannot
know if it was updated within the same second after we scanned it.
Therefore, we will force another scan if the dir was just modified.
* Changed to force another scan if the directory was just modified.
(closes issue ASTERISK-19081)
Reported by: Knut Bakke
Review: https://reviewboard.asterisk.org/r/1688/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@355056 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@354953 65c4cc65-6c06-0410-ace0-fbb531ad65f3
open_mailbox() was changed quite a long time ago to allocate this memory.
close_mailbox() should have been changed to be responsible for freeing it.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@354889 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The astman_get_header() never returns NULL so the check by the code for
NULL would never fail.
(closes issue ASTERISK-16974)
Reported by: Nuno Borges
Patches:
0018325.patch (license #6116) patch uploaded by Nuno Borges (modified)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@354835 65c4cc65-6c06-0410-ace0-fbb531ad65f3
CDRs cannot be modified after a bridge is torn down, (e.g. after
Dial() returns) even though the CDR() function may be called. Since
modifying the CDR code to change this behavior could very easily
break all kinds of things, this patch just documents this limitation.
(closes issues ASTERISK-16923)
Review: https://reviewboard.asterisk.org/r/1720/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@354749 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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@354655 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change makes it so computational cost is not taken into account
when deciding if a multistep path is better than a single-step path. This
means that the only time a multistep path will be chosen is if no single-step
path exists. This ensures a better quality translation even if it turns out
to be slightly slower.
(closes issue ASTERISK-16821)
reported by Andrew Lindh
Review: https://reviewboard.asterisk.org/r/1715
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@354594 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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@354542 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch removes some unnecessary locking of the channels container in
ast_hangup(). The reason this came up is that this lock can very quickly block
the entire system. If any of the channel cleanup code decides to block, it
causes a problem for the whole system. For example, when audiohooks get
destroyed, if that blocks for a while waiting on the mixmonitor thread to exit
because it's busy blocking on some I/O, it causes a problem for many other
threads in the meantime.
Review: https://reviewboard.asterisk.org/r/1712/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@354492 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1. Set lastms to 0 when clearing instead of ""
2. Don't set ipaddr or port to the string "(null)" when they are empty
3. Add missing required fields, set default for lastms to 0, and modify
the length of the ipaddr field to 45 in the Postgresql realtime.sql
file.
(closes issue ASTERISK-19172)
Review: https://reviewboard.asterisk.org/r/1703/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@354348 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Prior to this patch, attempts to reload cdr_pgsql.so would cause the column list to keep
its current data and then add a second copy during the reload. This would cause attempts
to log the CDR to the database to fail. This patch also cleans up some unnecessary null
checks for ast_free and deals with a few potential locking problems.
(closes issue ASTERISK-19216)
Reported by: Jacek Konieczny
Review: https://reviewboard.asterisk.org/r/1711/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@354263 65c4cc65-6c06-0410-ace0-fbb531ad65f3
