It is highly unlikely, but - at least in Asterisk 12 - theoretically possible
to load Asterisk with no dialplan whatsoever. If that occurs, and some other
module (that is not a pbx module) attempts to merge its contexts into the
dialplan, the existing merge routine will crash. This is because it is not
insane, and rightly believes that you provided some sort of dialplan,
somewhere.
This patch will gracefully merge the contexts in such a case. Note that this
is highly unlikely to occur in 1.8/11, as features will most likely provide
some dialplan via parking. However, in Asterisk 12, parking is now provided
by res_parking, and hence may create its dialplan later.
(closes issue ASTERISK-23297)
Reported by: CJ Oster
Review: https://reviewboard.asterisk.org/r/3222
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@408200 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In pbx.c ast_custom_function_unregister(), a list
of escalations being removed from the list wasn't
being free'd creating a leak. This patch corrects
that by freeing the records.
Review: https://reviewboard.asterisk.org/r/3213/
Reported by: Corey Farrell
Patches:
acf_escalating_leak.patch uploaded by coreyfarrell (license 5909)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@408142 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The "goodbye" parameter is not implemented in the source code, it does nothing.
(closes issue SWP-6518)
Reported By: Steve Pitts
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@408020 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When a DAHDI device is removed at run-time it sends the event
DAHDI_EVENT_REMOVED on each channel. This is intended to signal the
userspace program to close the respective file handle, as the driver of
the device will need all of them closed to properly clean-up.
This event has long since been handled in chan_dahdi (chan_zap at the
time). However the event that is sent on a D-Channel of a "PRI" (ISDN)
span simply gets ignored.
This commit adds handling for closing the file descriptor (and shutting
down the span, while we're at it).
It also adds a CLI command 'pri destroy span <N>' to destroy the span
and its DAHDI channels.
Backported from trunk/12.
Review: https://reviewboard.asterisk.org/r/726/
........
Merged revisions 394552 394567 from http://svn.asterisk.org/svn/asterisk/trunk
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@407817 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Establishing an IAX2 call between Asterisk v1.4 and v1.8 (or later)
results in an unexpected call disconnect. The problem happens because
newer values in the enum ast_control_frame_type are not consistent between
the branch versions of Asterisk.
For example:
1) v1.4 calls v1.8 (or later) using IAX2
2) v1.8 answers and sends a connected line update control frame. (on v1.8
AST_CONTROL_CONNECTED_LINE = 22)
3) v1.4 receives the control frame as an end-of-q (on v1.4
AST_CONTROL_END_OF_Q = 22)
4) v1.4 disconnects the call once the receive queue becomes empty.
Several things are done by this patch to fix the problem and attempt to
prevent it from happening again in the future:
* Added a warning at the definition of enum ast_control_frame_type about
how to add new control frame values.
* Made block sending and receiving control frames that have no reason to
go over the wire.
* Extended the connectedline iax.conf parameter to also include the
redirecting information updates.
* Updated the connectedline iax.conf parameter documentation to include a
notice that the parameter must be "no" when the peer is an Asterisk v1.4
instance.
(closes issue AST-1302)
Review: https://reviewboard.asterisk.org/r/3174/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@407678 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* If the "stutter" (voicemail indication) tone is indeed a stutter tone,
and it ends with a constant tone, make sure that it is the dial tone.
This was done for India (in), Mexico (mx) and the Philippines (ph).
* If no "stutter" tone exists for a country, provide one. This was done for
Spain (es), Malaysia (my) and Venezuela (ve).
Review: https://reviewboard.asterisk.org/r/3158/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@407622 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This fixes path handling for log files so that an extra / is not
appended to the file path when the path is absolute (begins with /).
This would previously result in different but functionally equivalent
paths in the output of 'logger show channels'.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@407455 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Thanks to Guillaume Martres for doing the necessary research to validate
the change.
(closes issue ASTERISK-17727)
Reported by: LN
Patches:
use_certificate_chain.patch (license #5864) patch uploaded by st
documente_certificate_chain.patch (license #6576) patch uploaded by Guillaume Martres
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@407272 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The code assumed that unregistering the alias would always succeed while in
practice this is not actually true. A common case is the "reload" command itself.
If the cli_aliases.conf configuration file was changed and reload executed the
command would fail to unregister and ultimately point to freed memory.
The reload process now checks whether unregistering succeeded or not and if not
the old CLI alias is retained.
(closes issue ASTERISK-19773)
Reported by: Joel Vandal
(closes issue ASTERISK-22757)
Reported by: Gareth Blades
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@407205 65c4cc65-6c06-0410-ace0-fbb531ad65f3
STACK_PEEK requires 2 parameters and LOCAL_PEEK requires 1 parameter. This
protects against situations where those parameters are blank or missing by
logging an error and returning.
(closes issue ASTERISK-23220)
Reported by: James Sharp
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@407100 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The parsing for the destination of the macro/gosub uses the '^' character to
separate out context, extension, and priority. However, the logic for the
macro/gosub execution was written such that it would only do the actual
macro/gosub jump if a '^' character existed. This doesn't apply when the
macro/gosub jump occurs in a priority/priority label. This patch changes
the logic so that the parsing still occurs, but the jump will occur even
for priorities/priority labels.
(issue ASTERISK-23164)
Review: https://reviewboard.asterisk.org/r/3154
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@407041 65c4cc65-6c06-0410-ace0-fbb531ad65f3
ast_bind to a port reserved for another program by SELinux causes
errno == EACCES. This caused random failures when binding rtp or
udptl sockets. Treat EACCES as a non-fatal error, try next port.
(closes issue ASTERISK-23134)
Reported by: Corey Farrell
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@406933 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Asterisk's RADIUS module currently build against libradiusclient-ng, but this
project has been superseeded by libfreeradius-client. The API is 99% compatible
except that the header name has changed, the library name has changed, and
the configuration file location has changed.
(closes issue ASTERISK-22980)
Reported by: Jeremy Lainé
Patches:
freeradius-client.patch uploaded by sharky (license 6561)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@406801 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In ast_rtp_instance_make_compatible(), after a failure of
channel tech call get_rtp_info() to return peer_instance,
the null pointer would be passed to ao2_ref, producing an
error that looked like a refernce counting problem but is
not. This patch corrects that and adds helpful LOG_ERROR
messages to indicate which failure path occurred.
(issue AST-1276)
Review: https://reviewboard.asterisk.org/r/3156/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@406721 65c4cc65-6c06-0410-ace0-fbb531ad65f3
extconfig.conf was hard-coded to not allow nested includes for some reason.
The code has been this way since a patch was merged for ASTERISK-3333 (revision
4889), which was a significant update to this code ("Merge config updates").
I can't figure out any good reason why this should be limited. This patch just
removes the limit and uses the default nesting depth limit.
Closes issue ASTERISK-17837
Review: https://reviewboard.asterisk.org/r/3159/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@406643 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The ast_filestream object gets tacked on to a channel via
chan->timingdata. It's a reference counted object, but the reference
count isn't used when putting it on a channel. It's theoretically
possible for another thread to interfere with the channel while it's
unlocked and cause the filestream to get destroyed.
Use the astobj2 reference count to make sure that as long as this code
path is holding on the ast_filestream and passing it into the file.c
playback code, that it knows it's valid.
Bug reported by Leif Madsen.
Review: https://reviewboard.asterisk.org/r/3135/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@406566 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The CEL data structures need to be protected during a configuration reload
and shutdown. Asterisk crashed during a shutdown because CEL events were
still in flight and the CEL data structures were already destroyed.
* Protected the appset and linkedids ao2 containers using the reload_lock.
* Added NULL checks before use of the appset and linkedids ao2 containers
in case the CEL module is already shutdown.
* Fixed overloading of the linkedids held objects reference count. During
shutdown any held objects would be leaked.
* Fixed memory leak of linkedids held objects if the LINKEDID_END is not
being tracked. The objects in the linkedids container were not removed if
the LINKEDID_END event is not used.
* Added access protection to the appset container during the CLI "cel show
status" command.
* Made CEL config reload not set defaults if the cel.conf file is invalid.
(closes issue AST-1253)
Reported by: Guenther Kelleter
Review: https://reviewboard.asterisk.org/r/3127/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@406417 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Made register atexit shutdown routine only once in __init_manager().
* Fixed some initial load failure conditions in __init_manager().
* Made reset options to defaults on reload when the reload will actually
happen.
* Fixed the order of unreferencing a session object in session_destroy().
* Removed unnecessary container traversals of the white/black filters
during session_destructor() and manager_free_user().
* ast_free() does not need a NULL check before calling.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@406359 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In ast_build_timing, initialize the timezone value to NULL
in order to avoid deferencing an uninitialized value later
when calling ast_destroy_timing. The timezone value could
be uninitialized if ast_build_timing were to fail due to a
zero length time string.
(closes issue ASTERISK-22861)
Reported by: Sebastian Murray-Roberts
Review: https://reviewboard.asterisk.org/r/3134/
Patches:
ast_build_timing-initialize-timezone.patch uploaded by coreyfarrell (license 5909)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@406241 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change allows chan_sip to decline individual image streams over
unsupported transports in the SDP of the 200 response. Previously,
an image stream offer with RTP/AVP as the transport would cause
chan_sip to respond with a 488.
(closes issue ASTERISK-22988)
Reported by: adomjan
Original patch by: adomjan
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@406170 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This restricts direct usage of global oseq so that all accesses are
locked and threads are not racing to get oseq values that they did not
claim.
This also fixes a build error in res_pktccops under dev mode.
(closes issue ASTERISK-23100)
Reported by: adomjan
Patch by: adomjan
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@406037 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Fixes typos of "transfered" instead of "transferred" in various code. Fixes incorrect gosub param help text for app_queue.
Fixes Asterisk man pages containing unquoted minus signs. Adds note about the "textsupport" option in sip.conf.sample.
(issue ASTERISK-23061)
(issue ASTERISK-23028)
(issue ASTERISK-23046)
(issue ASTERISK-23027)
(closes issue ASTERISK-23061)
(closes issue ASTERISK-23028)
(closes issue ASTERISK-23046)
(closes issue ASTERISK-23027)
Reported by: Eugene, Jeremy Laine, Denis Pantsyrev
Patches:
transferred.patch uploaded by Jeremy Laine (license 6561)
hyphen.patch uploaded by Jeremy Laine (license 6561)
sip.conf.sample.patch uploaded by Eugene (license 6360)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@405791 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Added some text to UPGRADES.txt about the V.27 mode rate changes in r405656.
(issue ASTERISK-22790)
Reported by: Paolo Compagnini
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@405692 65c4cc65-6c06-0410-ace0-fbb531ad65f3
According to the new standard for V.27 and V.32 they are able to transmit
at a bit rate of 4,800 or 9,600. The check_mode_rate function needed to be
updated to reflect this. Also, because of this change the default 'minrate'
value was updated to be 4800.
(closes issue ASTERISK-22790)
Reported by: Paolo Compagnini
Patches:
res_fax.txt uploaded by looserouting (license 6548)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@405656 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The cel_manager module did not properly handle the case where the
configuration file was invalid. The module will now output a warning
message and disable itself if this occurs.
Reported by: Bryan Walters
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@405581 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In ASTERISK-12117, an improvement to insure consistant local from tags
on outbound registrations resulted in an undesirable behavior - caused
by leftover unexpired sip_pvt dialogs (with the previous cseq number),
resulting in many uncessary REGISTER requests. Instead of significant
rework of transmit_register(), this change deletes the dialogs after a
200 OK response indiciating a successful registration, keeping the old
dialogs from interfering with normal operation.
(closes issue ASTERISK-22946)
Reported by: Stephan Eisvogel
Review: https://reviewboard.asterisk.org/r/3109/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@405433 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When performing a SIP transfer to a Park extension, if the Park fails, chan_sip
will currently not hang up either the transferer or the transfer target. This
results in the channels being orphaned with no thread to service frames,
resulting in stuck channels.
This patch immediately hangs up the two channels if a Park fails.
(closes issue ASTERISK-22834)
Reported by: rsw686
(closes issue ASTERISK-23047)
Reported by: Tommy Thompson
Review: https://reviewboard.asterisk.org/r/3107
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@405379 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Add a note that the "retry on 403 response to REGISTER" for chan_sip is
non-functional in the versions in which it was first introduced.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@405088 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In r404674 the AST_TEST_DEFINE(test_REPLACE) test was added
that made use of a function that doesn't exist in 1.8. This
fixes that by reverting to directly accessing chan varshead.
Reported by: Tzafrir Cohen
(issue ASTERISK-22910)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@404951 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Even since the fixes of AST-2013-007, Asterisk prints the following
warning on startup if the user decided to live dangerously:
Privilege escalation protection disabled!
See https://wiki.asterisk.org/wiki/x/1gKfAQ for more details.
This message is intended for the logs and interactive startup. No need
for it to appear on a remote console. This commit removes it from there.
(closes issue ASTERISK-23084)
Review: https://reviewboard.asterisk.org/r/3101/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@404861 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Upon reload the module unconditionally "unloaded" the module (freeing memory
and setting pointers to NULL) and then when attempting a "load" if the config
file had not changed then nothing would be reinitialized.
By moving the "unload" to occur conditionally (reload only) after an attempted
configuration load, but before module "loading" alleviates the issue. The module
now loads/unloads/reloads correctly.
(closes issue ASTERISK-22871)
Reported by: Matteo
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@404857 65c4cc65-6c06-0410-ace0-fbb531ad65f3
dahdi show channels output slices the callerid (which is dnid copied over on
PRI channels). If the channel naming structures look like:
'DAHDI/i1/1408409XXXX-6'
then the output slices 1408409XXXX down to 1408409XXX. This patch just opens
it up to 15 chars so you can see the whole thing.
(closes issue ASTERISK-22918)
Reported by: outtolunc
Patches:
svn_chan_dahdi.c.format12_15.diff.txt uploaded by outtolunc (license 5198)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@404784 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When calling REPLACE() with an empty replace-char argument, strcpy
is used to overwrite the the matching <find-char>. However as the
src and dest arguments to strcpy must not overlap, it causes other
parts of the string to be overwritten with adjacent characters and
the result is mangled. Patch replaces call to strcpy with memmove
and adds a test suite case for REPLACE.
(closes issue ASTERISK-22910)
Reported by: Gareth Palmer
Review: https://reviewboard.asterisk.org/r/3083/
Patches:
func_strings.patch uploaded by Gareth Palmer (license 5169)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@404674 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A deadlock can happen between a thread unloading or reloading the cel_pgsql
module and the core_event_dispatcher taskprocessor thread. Description of
what is happening:
Thread 1 (for example, a netconsole thread):
a "module reload cel_pgsql" is launched
the thread enter the "my_unload_module" function (cel_pgsql.c)
the thread acquire the write lock on psql_columns
the thread enter the "ast_event_unsubscribe" function (event.c)
the thread try to acquire the write lock on ast_event_subs[sub->type]
Thread 2 (core_event_dispatcher taskprocessor thread):
the taskprocessor pop a CEL event
the thread enter the "handle_event" function (event.c)
the thread acquire the read lock on ast_event_subs[sub->type]
the thread callback the "pgsql_log" function (cel_pgsql.c), since it's a subscriber of CEL events
the thread try to acquire a read lock on psql_columns
(closes issue ASTERISK-22854)
Reported by: Etienne Lessard
Patches:
cel_pgsql_fix_deadlock_event.patch uploaded by hexanol (license 6394)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@404603 65c4cc65-6c06-0410-ace0-fbb531ad65f3
