Kinsey Moore [Fri, 13 Jan 2012 21:42:12 +0000 (21:42 +0000)]
Make sure asterisk builds on OpenBSD
OpenBSD defines SO_PEERCRED, but it returns a 'struct sockpeercred', not
'struct ucred', which causes compilation of main/asterisk.c to fail in
read_credentials(). This allows configure to check for sockpeercred and
asterisk to deal with it properly.
(closes issue ASTERISK-18929) Reported-by: Barry Miller Patch-by: Barry Miller
........
Merged revisions 350730 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 350731 from http://svn.asterisk.org/svn/asterisk/branches/10
Matthew Jordan [Fri, 13 Jan 2012 17:00:12 +0000 (17:00 +0000)]
Realtime queues failed to load queue information without queue member table
Previously, realtime queues could be loaded without defining the queue member
table. This allowed for queue members to be dynamic, while the realtime
queue definitions could exist in some backing storage. Revision 342223 broke
this when it changed the return value for realtime_multientry to return NULL
when no results are returned. Previously, an empty ast_config object was
expected.
(closes issue ASTERISK-19170)
Reported by: Rene Mendoza
Tested by: Rene Mendoza
Patches:
rt_queue_member_patch.diff uploaded by Matt Jordan (license 6283)
........
Merged revisions 350552 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 350553 from http://svn.asterisk.org/svn/asterisk/branches/10
Matthew Jordan [Fri, 13 Jan 2012 16:48:06 +0000 (16:48 +0000)]
Fix crash from bridge channel hangup race condition in ConfBridge
This patch addresses two issues in ConfBridge and the channel bridge layer:
1. It fixes a race condition wherein the bridge channel could be hung up
2. It removes the deadlock avoidance from the bridging layer and makes the
bridge_pvt an ao2 ref counted object
Patch by David Vossel (mjordan was merely the commit monkey)
(issue ASTERISK-18988)
(closes issue ASTERISK-18885)
Reported by: Dmitry Melekhov
Tested by: Matt Jordan
Patches: chan_bridge_cleanup_v.diff uploaded by David Vossel (license 5628)
(closes issue ASTERISK-19100)
Reported by: Matt Jordan
Tested by: Matt Jordan
Richard Mudgett [Wed, 11 Jan 2012 21:56:12 +0000 (21:56 +0000)]
Make FollowMe optionally update connected line information when the accepting endpoint is bridged.
Like Dial and Queue, FollowMe needs to deal with
AST_CONTROL_CONNECTED_LINE information so when the parties are initially
bridged, the connected line information will be correct.
* Added the 'I' option just like the app_dial and app_queue 'I' option.
* Made 'N' option ignored if the call is already answered.
Terry Wilson [Wed, 11 Jan 2012 19:19:35 +0000 (19:19 +0000)]
Always treat arguments to get_by_name_cb as strings
Initially, support was left in for the old style of searching, even
though it wasn't actually used. In the case of name_len != 0, the
OBJ_KEY flag isn't passed because we aren't matching on a full key
and therefor can't use the hash function to optimize. The code left
in to support the old way of searching unfortunately treated a prefix
search like this as though an ast_channel struct was passed as an arg
and caused a crash.
This patch also adds needed parentheses around some matching conditions.
Terry Wilson [Mon, 9 Jan 2012 22:15:50 +0000 (22:15 +0000)]
Replace direct access to channel name with accessor functions
There are many benefits to making the ast_channel an opaque handle, from
increasing maintainability to presenting ways to kill masquerades. This patch
kicks things off by taking things a field at a time, renaming the field to
'__do_not_use_${fieldname}' and then writing setters/getters and converting the
existing code to using them. When all fields are done, we can move ast_channel
to a C file from channel.h and lop off the '__do_not_use_'.
This patch sets up main/channel_interal_api.c to be the only file that actually
accesses the ast_channel's fields directly. The intent would be for any API
functions in channel.c to use the accessor functions. No more monkeying around
with channel internals. We should use our own APIs.
The interesting changes in this patch are the addition of
channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to
channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to
use accessor functions when ast_channel is really opaque), and some re-working
of the way channel iterators/callbacks are handled so as to avoid creating fake
ast_channels on the stack to pass in matching data by directly accessing fields
(since "name" is a stringfield and the fake channel doesn't init the
stringfields, you can't use the ast_channel_name_set() function). I went with
ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a
setter.
The majority of the grunt-work for this change was done by writing a semantic
patch using Coccinelle ( http://coccinelle.lip6.fr/ ).
Walter Doekes [Mon, 9 Jan 2012 19:37:23 +0000 (19:37 +0000)]
Fix shutdown handling of sqlite3 astdb.
If a db_sync was scheduled just before shutdown, the atexit code calling
db_sync would have no effect, causing the astdb commit thread to stay
alive. This caused the SIP/realtime_sipregs test to fail. (The fallback
kill would run the atexit code again and that would wreak havoc.) This
fixes that the atexit kill condition is picked up properly.
(closes issue ASTERISK-18883)
Reviewed by: Terry Wilson
Update contrib script live_ast to invoke Asterisk with valgrind and suppression file.
* Added valgrind_compare script to compare two valgrind log files for
differences.
(issue ASTERISK-17339)
Reported by: Tzafrir Cohen
Patches:
valgrind_compare (license #5035) script uploaded by Tzafrir Cohen
live_ast_valgrind.diff (license #5035) patch uploaded by Tzafrir Cohen
live_ast_valgrind_v2.diff (license #5185) patch uploaded by Paul Belanger
........
r350128 | rmudgett | 2012-01-09 12:54:56 -0600 (Mon, 09 Jan 2012) | 11 lines
live_ast: valgrind: run asterisk under valgrind
Adds a new sub-command, "valgrind" to live_ast. It runs asterisk under
valgrind. The extra command-line parameters are passed to Asterisk as
usual, and parameters to valgrind are passed through LIVE_AST_VALGRIND_ARGS
in live.conf .
Review: https://reviewboard.asterisk.org/r/1109/
Merged revisions 326636 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 350127-350128 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 350129 from http://svn.asterisk.org/svn/asterisk/branches/10
Kinsey Moore [Mon, 9 Jan 2012 15:40:16 +0000 (15:40 +0000)]
Prevent SLA settings from getting wiped out on reload
If SLA was reloaded without the config file being changed, current settings got
wiped out before the SLA reload code decided it wasn't going to reload the file
since nothing was changed. Moving the settings reset later in the reload
process fixes this.
(closes issue AST-744)
........
Merged revisions 350023 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 350024 from http://svn.asterisk.org/svn/asterisk/branches/10
Terry Wilson [Fri, 6 Jan 2012 23:31:25 +0000 (23:31 +0000)]
Don't leak CID in From header when presentation=unavailable
When someone does Set(CALLERPRES()=unavailable) (or
Set(CALLERID(pres)=unavailable)) when sendrpid=no, the From header shows
"Anonymous" <anonymous@anonymous.invalid>. When sendrpid=yes/pai, the From
header will still display the callerid info, even though we supply an rpid
header with the anonymous info. It seems like we shouldn't leak that info in
any case. Skimming http://tools.ietf.org/html/draft-ietf-sip-privacy-04 seems
to indicate that one shouldn't send identifying info in the From in this case.
This patch anonymizes the From header as well even when sendrpid=yes/pai.
Kinsey Moore [Fri, 6 Jan 2012 21:26:16 +0000 (21:26 +0000)]
Fix lua goto detection to prevent unexpected behavior with confbridge
A bug in the pbx_lua goto detection was causing the dialplan to hangup
unexpectedly after confbridge exited if it had called lua dialplan code during
execution.
Patch-by: Timo Teras Acked-by: Matt Nicholson
(closes issue ASTERISK-18976)
........
Merged revisions 349928 from http://svn.asterisk.org/svn/asterisk/branches/10
Matthew Jordan [Thu, 5 Jan 2012 23:58:26 +0000 (23:58 +0000)]
Fix premature free'ing of the frame committed in r349608
Even though we set the frame to the ast_null_frame and return that,
the caller of the frame hook may still need the frame. This now is
a bit more careful about when it frees the frame, i.e., only under
the same conditions that applied when we duplicated it in the first
place.
........
Merged revisions 349822 from http://svn.asterisk.org/svn/asterisk/branches/10
Richard Mudgett [Thu, 5 Jan 2012 23:47:11 +0000 (23:47 +0000)]
Make not assume that the cel_sqlite3_custom SQL table primary key is AcctId.
If a table is created by some other application and the primary key is not
named "AcctId", cel/cel_sqlite3_custom.c will always try to create the
table and fail because it already exists.
* Change the SQL table query to not require AcctId as the primary key.
Richard Mudgett [Thu, 5 Jan 2012 23:06:17 +0000 (23:06 +0000)]
Make pbx_config.c use Gosub instead of Macro call for stdexten.
Users created by users.conf with hasvoicemail=yes have been documented as
using a Gosub to stdexten since v1.6.0. However, the code still generates
dialplan to access stdexten as a Macro as documented in v1.4; which does
not work with the newer extensions.conf.sample file.
* Make generated dialplan access the stdexten dialplan with the documented
Gosub instead of the older Macro style.
(closes issue ASTERISK-18809)
Reported by: Jay Allen
Patches:
gosub_patch-pbx_config.patch (license #6323) patch uploaded by Jay Allen (modified)
Tested by: rmudgett
Kinsey Moore [Thu, 5 Jan 2012 22:11:41 +0000 (22:11 +0000)]
Allow playback of formats that don't support seeking
ast_streamfile previously did unconditional seeking on files that broke
playback of formats that don't support that functionality. This patch avoids
the seek that was causing the problem. This regression was introduced in
r158062.
(closes issue ASTERISK-18994) Patch-by: Timo Teras
........
Merged revisions 349731 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 349732 from http://svn.asterisk.org/svn/asterisk/branches/10
Jonathan Rose [Thu, 5 Jan 2012 22:02:33 +0000 (22:02 +0000)]
Fix an issue where dsp.c would interpret multiple dtmf events from a single key press.
When receiving calls from a mobile phone into a DISA system on a connection with
significant interference, the reporter's Asterisk system would interpret DTMF incorrectly
and replicate digits received. This patch resolves that by increasing the number of
frames a mismatch has to be detected before assuming the DTMF is over by 1 frame and
adjusts dtmf_detect function to reset hits and misses only when an edge is detected.
(closes issue ASTERISK-17493)
Reported by: Alec Davis
Patches:
bug18904-refactor.diff.txt uploaded by Alec Davis (license 5546)
Review: https://reviewboard.asterisk.org/r/1130/
........
Merged revisions 349728 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 349729 from http://svn.asterisk.org/svn/asterisk/branches/10
Jonathan Rose [Thu, 5 Jan 2012 16:16:51 +0000 (16:16 +0000)]
Ensures Asterisk closes when receiving terminal signals in 'no fork' mode.
When catching a signal, in no fork mode the console thread is identical to the thread
responsible for catching the signal and closing Asterisk, which requires it to first
dispense with the console thread. Prior to this patch, if these threads were identical,
upon receiving a killing signal, the thread will send an URG signal to itself, which
we also catch and then promptly do nothing with. Obviously this isn't useful behavior.
(closes issue ASTERISK-19127)
Reported By: Bryon Clark
Patches:
quit_on_signals.patch uploaded by Bryon Clark (license 6157)
........
Merged revisions 349672 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 349673 from http://svn.asterisk.org/svn/asterisk/branches/10
Matthew Jordan [Wed, 4 Jan 2012 22:23:28 +0000 (22:23 +0000)]
Fix for ConfBridge config parser unlocking channel mutex too many times
When looking up a ConfBridge profile, the config parser would, if it
found a channel datastore on the channel requesting the bridge profile,
unlock the channel mutex twice. Since that's a little aggressive,
it now only unlocks it once.
(closes issue ASTERISK-19042)
Reported by: Matt Jordan
Tested by: Matt Jordan
Patches:
19042 uploaded by David Vossel (license 5628)
........
Merged revisions 349619 from http://svn.asterisk.org/svn/asterisk/branches/10
Matthew Jordan [Wed, 4 Jan 2012 21:40:45 +0000 (21:40 +0000)]
Free successfully translated frame in fax_gateway_framehook
A frame that is translated via ast_translate is also duplicated via ast_frdup.
This will allocate a new frame on the heap, which needs to be free'd
at the appropriate time. This issue reporter used valgrind to find that this
occurred in res_fax's fax_gateway_framehook; a quick search through the code
showed that only place this was currently not handling the translatted frame
properly.
Richard Mudgett [Wed, 4 Jan 2012 20:55:59 +0000 (20:55 +0000)]
Fix segfault in chan_dahdi for CHANNEL(dahdi_span) evaluation on hangup.
* Added NULL private pointer checks in the following chan_dahdi channel
callbacks: dahdi_func_read(), dahdi_func_write(), dahdi_setoption(), and
dahdi_queryoption().
Kinsey Moore [Wed, 4 Jan 2012 20:24:25 +0000 (20:24 +0000)]
Make debian init script conform to the LSB standard
Previously, this init script would return 1 if Asterisk was already running.
This is incorrect behavior according to the LSB standard and has been fixed by
returning 0 instead.
Kinsey Moore [Wed, 4 Jan 2012 20:02:34 +0000 (20:02 +0000)]
Update autosupport script and man page
Added information collection from the output of the utilities: top, free, uptime, ifconfig
Added information collection from the output of the Asterisk command 'dahdi show status'
Added option / flag '-n, --non-interactive'
Updated man page to reflect new option / flag '-n, --non-interactive'
Patch-by: John Bigelow (itzanger)
(closes issue AST-749)
........
Merged revisions 349504 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 349505 from http://svn.asterisk.org/svn/asterisk/branches/10
Matthew Jordan [Thu, 29 Dec 2011 15:16:46 +0000 (15:16 +0000)]
Handle AST_CONTROL_UPDATE_RTP_PEER frames in local bridge loop
Failing to handle AST_CONTROL_UPDATE_RTP_PEER frames in the local bridge loop
causes the loop to exit prematurely. This causes a variety of negative side
effects, depending on when the loop exits. This patch handles the frame by
essentially swallowing the frame in the local loop, as the current channel
drivers expect the RTP bridge to handle the frame, and, in the case of the
local bridge loop, no additional action is necessary.
(issue ASTERISK-19040)
(issue ASTERISK-19128)
(issue ASTERISK-17725)
(issue ASTERISK-18340)
(closes issue ASTERISK-19095)
Reported by: Stefan Schmidt
Tested by: Matt Jordan
Kevin P. Fleming [Wed, 28 Dec 2011 18:59:16 +0000 (18:59 +0000)]
Improve T.38 gateway V.21 preamble detection.
This commit removes the V.21 preamble detection code previously added to the
generic DSP implementation in Asterisk, and instead enhances the res_fax module
to be able to utilize V.21 preamble detection functionality made available by
FAX technology modules. This commit also adds such support to res_fax_spandsp,
which uses the Spandsp modem tone detection code to do the V.21 preamble
detection.
There should be no functional change here, other than much more reliable V.21
preamble detection (and thus T.38 gateway initiation).
........
Merged revisions 349248 from http://svn.asterisk.org/svn/asterisk/branches/10
Matthew Jordan [Tue, 27 Dec 2011 20:55:15 +0000 (20:55 +0000)]
Fix timing source dependency issues with MOH
Prior to this patch, res_musiconhold existed at the same module priority level
as the timing sources that it depends on. This would cause a problem when
music on hold was reloaded, as the timing source could be changed after
res_musiconhold was processed. This patch adds a new module priority level,
AST_MODPRI_TIMING, that the various timing modules are now loaded at. This
now occurs before loading other resource modules, such that the timing source
is guaranteed to be set prior to resolving the timing source dependencies.
(closes issue ASTERISK-17474)
Reporter: Luke H
Tested by: Luke H, Vladimir Mikhelson, zzsurf, Wes Van Tlghem, elguero, Thomas Arimont
Patches:
asterisk-17474-dahdi_timing-infinite-wait-fix_v3_branch-1.8.diff uploaded by elguero (License #5026)
asterisk-17474-dahdi_timing-infinite-wait-fix_v3_branch-10.diff uploaded by elguero (License #5026)
asterisk-17474-dahdi_timing-infinite-wait-fix_v3.diff uploaded by elguero (License #5026)
Sean Bright [Tue, 27 Dec 2011 17:17:58 +0000 (17:17 +0000)]
Once an audiohook is attached to a channel, we continue to transcode all of the
frames, even after all of the hooks are detached. This patch short-cicuits us
out before we transcode unnecessarily.
........
Merged revisions 349144 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 349145 from http://svn.asterisk.org/svn/asterisk/branches/10
Matthew Jordan [Fri, 23 Dec 2011 21:19:52 +0000 (21:19 +0000)]
Allow overriding of IMAP server settings on a user by user basis
This patch allows the imapserver, imapport, and imapflags settings to be
overridden for any voicemail user. It also documents the settings in
the sample voicemail.conf file, and updates the voicemail schema to
allow storage of those columns.
(closes issue ASTERISK-16489)
Reporter: Hubert Mickael
Tested by: Matt Jordan
Jonathan Rose [Fri, 23 Dec 2011 20:42:21 +0000 (20:42 +0000)]
INFO/Record request configurable to use dynamic features
Adds two new options to SIP peers allowing them to specify features (dynamic or builtin)
to use when sending INFO/record requests. Recordonfeature activates whatever feature
is specified when recieving a record: on request while recordofffeature activates
whatever feature is specified when receiving a record: off request. Both of these
features can be disabled by setting the feature to an empty string.
(closes issue ASTERISK-16507)
Reported by: Jon Bright
Review: https://reviewboard.asterisk.org/r/1634/
Jonathan Rose [Fri, 23 Dec 2011 20:19:33 +0000 (20:19 +0000)]
chan_sip autocreatepeer=persist option for auto-created peers to survive reload
This patch moves destruction of sip peers to immediately after the general section of
sip.conf is read so that autocreatepeer setting can be read before deletion of peers.
If autocreatepeer=persist at reload, then peers created by the autocreatepeer setting
will be skipped when purging the current SIP peer list.
In ChanSpy, don't create audiohooks that will never be used.
When ChanSpy is initialized it creates and attaches 3 audiohooks:
1) Read audio off of the channel that we are spying on
2) Write audio to the channel that we are spying on
3) Write audio to the channel that is bridged to the channel that we are
spying on.
The first is always necessary, but the others are used only when specific
options are passed to the ChanSpy application (B, d, w, and W to be specific).
When those flags are not passed, neither of those audiohooks are ever sent
frames, but we still try to process the hooks for each voice frame that we
recieve on the channel.
So in short - only create and attach audiohooks that we actually need.
........
................
Richard Mudgett [Fri, 23 Dec 2011 02:35:13 +0000 (02:35 +0000)]
Fix extension state callback references in chan_sip.
Chan_sip gives a dialog reference to the extension state callback and
assumes that when ast_extension_state_del() returns, the callback cannot
happen anymore. Chan_sip then reduces the dialog reference count
associated with the callback. Recent changes (ASTERISK-17760) have
resulted in the potential for the callback to happen after
ast_extension_state_del() has returned. For chan_sip, this could be very
bad because the dialog pointer could have already been destroyed.
* Added ast_extension_state_add_destroy() so chan_sip can account for the
sip_pvt reference given to the extension state callback when the extension
state callback is deleted.
* Fix pbx.c awkward statecbs handling in ast_extension_state_add_destroy()
and handle_statechange() now that the struct ast_state_cb has a destructor
to call.
* Ensure that ast_extension_state_add_destroy() will never return -1 or 0
for a successful registration.
* Fixed pbx.c statecbs_cmp() to compare the correct information. The
passed in value to compare is a change_cb function pointer not an object
pointer.
* Make pbx.c ast_merge_contexts_and_delete() not perform callbacks with
AST_EXTENSION_REMOVED with locks held. Chan_sip is notorious for
deadlocking when those locks are held during the callback.
* Removed unused lock declaration for the pbx.c store_hints list.
Matthew Jordan [Thu, 22 Dec 2011 22:39:29 +0000 (22:39 +0000)]
Fix for memory leaks / cleanup in cel_pgsql
There were a number of issues in cel_pgsql's pgsql_log method:
* If either sql or sql2 could not be allocated, the method would return while
the pgsql_lock was still locked
* If the execution of the log statement succeeded, the sql and sql2 structs
were never free'd
* Reconnection successes were logged as ERRORs. In general, the severity of
several logging statements was reduced
(closes issue ASTERISK-18879)
Reported by: Niolas Bouliane
Tested by: Matt Jordan
Terry Wilson [Thu, 22 Dec 2011 20:39:48 +0000 (20:39 +0000)]
Allow packetization vaules > 127
According to the RTP packetization documentation, and the maximum values
listed in AST_FORMAT_LIST, we should support values > that the signed
char array that ast_codec_pref makes available to store the value. All
places in the code treat the framing field as though it were an int
array instaead of a char array anyway, so this just fixes the type of
the array.
This adds support for setting several safe_asterisk parameters using
environment variables and also enables a custom run directory for asterisk
(instead of defaulting to /tmp).
Patch by: Byron Clark (byronclark)
(closes ASTERISK-17810)
Richard Mudgett [Mon, 19 Dec 2011 21:43:19 +0000 (21:43 +0000)]
Fix crashes on other platforms caused by interference from Darwin weak symbol support.
Support weak symbols on a platform specific basis. The Mac OS X (Darwin)
support must be isolated from the other platforms because it has caused
other platforms to crash. Several other platforms including Linux have
GCC versions that define the weak attribute. However, this attribute is
only setup for use in the code by Darwin.
(closes issue ASTERISK-18728)
Reported by: Ben Klang
Terry Wilson [Mon, 19 Dec 2011 01:36:21 +0000 (01:36 +0000)]
Add a separate buffer for SRTCP packets
The function ast_srtp_protect used a common buffer for both SRTP and SRTCP
packets. Since this function can be called from multiple threads for the same
SRTP session (scheduler for SRTCP and channel for SRTP) it was possible for the
packets to become corrupted as the buffer was used by both threads
simultaneously.
This patch adds a separate buffer for SRTCP packets to avoid the problem.
(closes issue ASTERISK-18889, Reported/patch by Daniel Collins)
........
Merged revisions 347995 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 347996 from http://svn.asterisk.org/svn/asterisk/branches/10
Richard Mudgett [Fri, 16 Dec 2011 23:58:44 +0000 (23:58 +0000)]
Clean-up on isle five for __ast_request_and_dial() and ast_call_forward().
* Add locking when a channel inherits variables and datastores in
__ast_request_and_dial() and ast_call_forward(). Note: The involved
channels are not active so there was minimal potential for problems.
* Remove calls to ast_set_callerid() in __ast_request_and_dial() and
ast_call_forward() because the set information is for the wrong direction.
* Don't use C++ keywords for variable names in ast_call_forward().
* Run the redirecting interception macro if defined when forwarding a call
in ast_call_forward(). Note: Currently will never execute because the
only callers that supply a calling channel supply a hungup or zombie
channel.
* Make feature_request_and_dial() put the transferee into autoservice when
it calls ast_call_forward() in case a redirection interception macro is
run. Note: Currently will never happen because the caller channel (Party
B) is always hungup at this time.
* Make feature_request_and_dial() ignore the AST_CONTROL_PROCEEDING frame
to silence a log message.
........
Merged revisions 348464 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 348465 from http://svn.asterisk.org/svn/asterisk/branches/10
Jonathan Rose [Fri, 16 Dec 2011 22:00:37 +0000 (22:00 +0000)]
Voicemail with the saycid option will now play a caller's name based on cid if available.
In order to check the availability of the caller's name, app_voicemail will check for an
audio file in <astspooldir>/recordings/callerids/
This change sets a precedent for where to put recordings of names. Currently the idea is
that recordings here could also be used for applications like confbridge and meetme to
find recorded names in this folder from callerid (when another recording isn't available)
(closes issue ASTERISK-18565)
Reporter: Russell Brown
Patches:
r uploaded by Russel Brown (license 6182)
Richard Mudgett [Fri, 16 Dec 2011 21:10:19 +0000 (21:10 +0000)]
Fix crash during CDR update.
The ast_cdr_setcid() and ast_cdr_update() were shown in ASTERISK-18836 to
be called by different threads for the same channel. The channel driver
thread and the PBX thread running dialplan.
* Add lock protection around CDR API calls that access an ast_channel
pointer.
Matthew Jordan [Wed, 14 Dec 2011 22:36:30 +0000 (22:36 +0000)]
Added support for all slin formats to app_originate
Previously, app_originate could not originate a call into a non-8kHz conference
bridge as the formats for non-8kHz slin codecs were not applied to the created
channel. This patch adds all of the formats by default, such that if a created
channel has a codec that supports a higher sampling rate, a translation path
can be built between it and other channels.
........
Merged revisions 348265 from http://svn.asterisk.org/svn/asterisk/branches/10
Matthew Jordan [Wed, 14 Dec 2011 22:08:55 +0000 (22:08 +0000)]
Fixed Asterisk crash when function QUEUE_MEMBER receives invalid input
The function QUEUE_MEMBER has two required parameters (queuename, option). It
was only checking for the presence of queuename. The patch checks for the
existence of the option parameter and provides better error logging when
invalid values are provided for the option parameter as well.
........
Merged revisions 348211 from http://svn.asterisk.org/svn/asterisk/branches/10
Jonathan Rose [Wed, 14 Dec 2011 21:08:20 +0000 (21:08 +0000)]
Add and document PARKEDCALL variable set during timeout
PARKEDCALL variable tracks which parking lot the call was last parked in. This can be
used afterwards for flow control when returntoorigin is set to off. I went ahead and
documented both this and the existing variable set during timeout (PARKINGSLOT) in
the sample features.conf since there was no prior mention of variables being set during
timeout.
Richard Mudgett [Tue, 13 Dec 2011 23:10:42 +0000 (23:10 +0000)]
Fix FollowMe CallerID on outgoing calls.
The addition of the Connected Line support changed how CallerID is passed
to outgoing calls. The FollowMe application was not updated to pass
CallerID to the outgoing calls.
* Fix FollowMe CallerID on outgoing calls.
* Restructured findmeexec() to fix several memory leaks and eliminate some
duplicated code.
* Made check the return value of create_followme_number(). Putting a NULL
into the numbers list is bad if create_followme_number() fails.
* Fixed a couple uses of ast_strdupa() inside loops.
* The changes to bridge_builtin_features.c fix a similar CallerID issue
with the bridging API attended and blind transfers. (Not used at this
time.)
Stefan Schmidt [Tue, 13 Dec 2011 15:22:48 +0000 (15:22 +0000)]
Fix possible misshandling of an incoming SIP response as a peer poke response.
Also make sure peer has even qualify enabled when handle a peer poke response.
Matthew Jordan [Mon, 12 Dec 2011 19:35:08 +0000 (19:35 +0000)]
Backed out core changes from r346391
During testing, it was discovered that there were a number of side effects
introduced by r346391 and subsequent check-ins related to it (r346429,
r346617, and r346655). This included the /main/stdtime/ test 'hanging',
as well as the remote console option failing to receive the appropriate output
after a period of time.
I only backed out the changes to main/ and utils/, as this was adequate
to reverse the behavior experienced.
Jonathan Rose [Fri, 9 Dec 2011 21:47:28 +0000 (21:47 +0000)]
Adds MixMonitor and StopMixMonitor AMI commands to the manager
These commands work much like the dialplan applications that would otherwise invoke them.
A nice benefit of these is that they can be invoked on a call remotely and at any time
during a call. They work much like the Monitor and StopMonitor ami commands.
Jonathan Rose [Fri, 9 Dec 2011 20:27:03 +0000 (20:27 +0000)]
Remove autojump extensions from SayUnixTime, make an option to perform automatic jumps.
When a caller sends DTMF while the SayUnixTime application is saying the time, The call
would jump to the next extension much like it does during Background(). This patch adds
option 'j' to SayUnixTime which when used employs the old behavior. Also, this patch
allows arguments to sayunixtime to not be used as empty strings in the case of something
like 'sayunixtime(,,,j)' or 'sayunixtime(,,pattern).
Richard Mudgett [Fri, 9 Dec 2011 01:33:29 +0000 (01:33 +0000)]
Fix some parsing issues in add_exten_to_pattern_tree().
* Simplify compare_char() and avoid potential sign extension issue.
* Fix infinite loop in add_exten_to_pattern_tree() handling of character
set escape handling.
* Added buffer overflow checks in add_exten_to_pattern_tree() character
set collection.
* Made ignore empty character sets.
* Added escape character handling to end-of-range character in character
sets. This has a slight change in behavior if the end-of-range character
is an escape character. You must now escape it.
* Fix potential sign extension issue when expanding character set ranges.
* Made remove duplicated characters from character sets. The duplicate
characters lower extension matching priority and prevent duplicate
extension detection.
* Fix escape character handling when the escape character is trying to
escape the end-of-string. We could have continued processing characters
after the end of the exten string. We could have added the previous
character to the pattern matching tree incorrectly.
Walter Doekes [Thu, 8 Dec 2011 21:32:36 +0000 (21:32 +0000)]
Fix regression when using tcpenable=no and tlsenable=yes.
The tlsenable settings are tucked away in main/tcptls.c, so I missed
them when resolving ASTERISK-18837. This should resolve the test suite
breakage of the sip tls tests.
Review: https://reviewboard.asterisk.org/r/1615
Reviewed by: Matt Jordan
........
Merged revisions 347718 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 347727 from http://svn.asterisk.org/svn/asterisk/branches/10
Jonathan Rose [Thu, 8 Dec 2011 20:55:19 +0000 (20:55 +0000)]
Fix regressed behavior of queue set penalty to work without specifying 'in <queuename>'
r325483 caused a regression in Asterisk 10+ that would make Asterisk segfault when
attempting to set penalty on an interface without specifying a queue in the queue set
penalty CLI command. In addition, no attempt would be made whatsoever to perform the
penalty setting on all the queues in the core list with either the cli command or the
non-segfaulting ami equivalent. This patch fixes that and also makes an attempt to
document and rename some functions required by this command to better represent what
they actually do. Oh yeah, and the use of this command without specifying a specific
queue actually works now.
Richard Mudgett [Thu, 8 Dec 2011 17:55:07 +0000 (17:55 +0000)]
Mark channel running the h exten with the soft-hangup flag.
When a bridge is broken, ast_bridge_call() might execute the h exten on
the calling channel. However, that channel may not have been the channel
that broke the bridge by hanging up. The channel executing the h exten
must be in a hung up state so things like AGI run in the correct mode.
* Make sure ast_bridge_call() marks the channel it is executing the h
exten on as hung up. (The AST_SOFTHANGUP_APPUNLOAD flag is used so as to
match the pbx.c main dialplan execution loop when it executes the h
exten.)
(closes issue ASTERISK-18811)
Reported by: David Hajek
Patches:
jira_asterisk_18811_v1.8.patch (license #5621) patch uploaded by rmudgett
Tested by: David Hajek, rmudgett
........
Merged revisions 347595 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 347600 from http://svn.asterisk.org/svn/asterisk/branches/10
Terry Wilson [Thu, 8 Dec 2011 16:24:29 +0000 (16:24 +0000)]
Don't crash on INFO automon request with no channel
AST-2011-014. When automon was enabled in features.conf, it was possible
to crash Asterisk by sending an INFO request if no channel had been
created yet.
(closes issue ASTERISK-18805)
........
Merged revisions 347530 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2
........
Merged revisions 347531 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 347532 from http://svn.asterisk.org/svn/asterisk/branches/10
Damien Wedhorn [Thu, 8 Dec 2011 06:59:01 +0000 (06:59 +0000)]
Fix segfault on answer.
Fix a segfault if an attempt to answer a call is made between when
the inbound call gives up (and the channel is removed) and when the
device is notified and removes the call from the device.
Jonathan Rose [Wed, 7 Dec 2011 20:34:23 +0000 (20:34 +0000)]
Fix: Meetme recording variables from realtime DB use null entries over channel variables
Meetme would attempt to substitute the realtime values of RECORDING_FILE and
RECORDING_FORMAT from the meetme db entry instead of using the channel variable set
for those variables in spite of those database entries being NULL or even lacking
a column to represent them.
(closes issue ASTERISK-18873)
Reported by: Byron Clark
Patches:
ASTERISK-18873-1.patch uploaded by Byron Clark (license 6157)
........
Merged revisions 347369 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 347383 from http://svn.asterisk.org/svn/asterisk/branches/10
Jonathan Rose [Tue, 6 Dec 2011 22:01:00 +0000 (22:01 +0000)]
Documents CHANNEL(musicclass) taking priority over m([x]) in waitExten
If waitExten specifies a music class to use with its music on hold option, it will use
CHANNEL(musicclass) instead if that channel variable has been set on the initiating
channel. This documents that behavior in the waitExten app so that this can be known
without checking the documentation of the code in function local_ast_moh_start.
(closes issue ASTERISK-18804)
........
Merged revisions 347239 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 347240 from http://svn.asterisk.org/svn/asterisk/branches/10
Walter Doekes [Tue, 6 Dec 2011 19:44:27 +0000 (19:44 +0000)]
Don't allow transport=tcp when tcpenable=no.
When tcpenable=no, sending to transport=tcp hosts was still allowed.
Resolving the source address wasn't possible and yielded the string
"(null)" in SIP messages. Fixed that and a couple of not-so-correct
log messages.
(closes issue ASTERISK-18837)
Reported by: Andreas Topp
Review: https://reviewboard.asterisk.org/r/1585
Reviewed by: Matt Jordan
........
Merged revisions 347166 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 347167 from http://svn.asterisk.org/svn/asterisk/branches/10
Walter Doekes [Tue, 6 Dec 2011 19:28:18 +0000 (19:28 +0000)]
The voicemail [general] zonetag and locale variables weren't loaded
until after the mailboxes were initialized. This caused the settings to
be unset for those mailboxes until a reload was performed.
(closes issue ASTERISK-18838)
Review: https://reviewboard.asterisk.org/r/1570
Reviewed by: Matt Jordan
........
Merged revisions 347111 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 347124 from http://svn.asterisk.org/svn/asterisk/branches/10
Richard Mudgett [Tue, 6 Dec 2011 19:09:56 +0000 (19:09 +0000)]
Doubly linked lists unit test and update to implementation.
Update the doubly linked list implementation. Now safe traversing can
insert before and after the current node when traversing in either
direction.
Updated the linked lists unit test test_linkedlist to also test doubly
linked lists. The old test_dlinkedlist requires a manual check of results
and probably should be removed.
Matthew Jordan [Tue, 6 Dec 2011 17:34:35 +0000 (17:34 +0000)]
Fixed crash from orphaned MWI subscriptions in chan_sip
This patch resolves the issue where MWI subscriptions are orphaned
by subsequent SIP SUBSCRIBE messages. When a peer is removed, either
by pruning realtime SIP peers or by unloading / loading chan_sip, the
MWI subscriptions that were orphaned would still be on the event engine
list of valid subscriptions but have a pointer to a peer that no longer
was valid. When an MWI event would occur, this would cause a seg fault.
(closes issue ASTERISK-18663)
Reported by: Ross Beer
Tested by: Ross Beer, Matt Jordan
Patches:
blf_mwi_diff_12_06_11.txt uploaded by Matt Jordan (license 6283)
Richard Mudgett [Mon, 5 Dec 2011 17:44:15 +0000 (17:44 +0000)]
Restore call progress code for analog ports.
Extracting sig_analog from chan_dahdi lost call progress detection
functionality.
* Fix analog ports from considering a call answered immediately after
dialing has completed if the callprogress option is enabled.
(closes issue ASTERISK-18841)
Reported by: Richard Miller
Patches:
chan_dahdi.diff (license #5685) patch uploaded by Richard Miller (Modified by me)
sig_analog.c.diff (license #5685) patch uploaded by Richard Miller (Modified by me)
sig_analog.h.diff (license #5685) patch uploaded by Richard Miller
........
Merged revisions 347006 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 347007 from http://svn.asterisk.org/svn/asterisk/branches/10
Jonathan Rose [Mon, 5 Dec 2011 15:04:12 +0000 (15:04 +0000)]
Resolve duplicate label used in multiple priorities for the same extension.
Prior to this patch, if labels with the same name were used for different priorities in
the same extension, the new label would be accepted, but it would be unusable since
attempts to reach that label would just go to the first one. Now pbx.c detects this,
generates a warning in logs, and culls the label before adding it to the dialplan.
projects / thirdparty / asterisk.git / log
