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().
ASTERISK-22846 #close
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3414/
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ASTERISK-23391 caused a regression where the symbol 'defaultlanguage'
was used by app_voicemail but not exported by main/asterisk. This
change renames the variable to ast_defaultlanguage. The variable was
already renamed in Asterisk 12+.
(closes issue ASTERISK-23559)
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/3408/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@411633 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch works around a problem with the HTTP body
being dropped from the response to a specific client
and under specific circumstances:
a) Client request comes from node.js user agent
"Shred" via use of swagger-client library.
b) Asterisk and Client are *not* on the same
host or TCP/IP stack
In testing this problem, it has been determined that
the write of the HTTP body is lost, even if the data
is written using low level write function. The only
solution found is to instruct the TCP stack with the
shutdown function to flush the last write and finish
the transmission. See review for more details.
ASTERISK-23548 #close
(closes issue ASTERISK-23548)
Reported by: Sam Galarneau
Review: https://reviewboard.asterisk.org/r/3402/
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* Backport ast_register_cleanup from Asterisk 12.
* Use ast_register_cleanup for format_attr_shutdown.
ast_register_cleanup was originally commited in r390122 by dlee.
(closes issue ASTERISK-23103)
Reported by: JoshE
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@411310 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Callerid checksum-ing was being handled incorrectly here. When the checksum is
calculated to be 0x00, it will perform 0x100-0x00 which results in 0x100. This
value will then fail the otherwise correct callerid message.
This patch changes the logic to simply add the calculated checksum to the
transmitted 2's compliment checksum.
Review: https://reviewboard.asterisk.org/r/3356/
(closes issue ASTERISK-23488)
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The syncing thread sleeps for a second before waiting to be
told to attempt to sync again. If a signal were sent during this
sleeping period, we would end up having to wait until the next
sync signal occurred in order to sync up the astdb.
This code rearrangement also ensures that any pending transactions
will be synced prior to Asterisk shutting down.
Patches: db_sync.patch by John Hardin (License #6512)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@410556 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Sending a HTTP request that is handled by Asterisk with a large number of
Cookie headers could overflow the stack.
Another vulnerability along similar lines is any HTTP request with a
ridiculous number of headers in the request could exhaust system memory.
(closes issue ASTERISK-23340)
Reported by: Lucas Molas, researcher at Programa STIC, Fundacion; and Dr. Manuel Sadosky, Buenos Aires, Argentina
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The test of the result of the stat() call was inverted such that its
output was only used if the call failed. This inverts the test so that
the output of stat() is used correctly. This was causing full reloads
on unchanged files.
(closes issue ASTERISK-23383)
Reported by: David Woolley
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When nanosecond time resolution was added for identifying config file
changes, it didn't cover all of the myriad of ways that one might obtain
nanosecond time resolution off of struct stat.
Rather than complicate the #if even further figuring out one system from
the next, this patch directly tests for the three struct members I know
about today, and #ifdef's accordingly.
Review: https://reviewboard.asterisk.org/r/3273/
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Upon failure of an INVITE transaction meant to initiate a remote native
bridge, rtp_engine.c would not clean up non-reference-counted bridge
instance pointers leaving a dangling pointer which was being used to
perform a local native bridge after the other channel had hung up. This
lead to dereferencing into freed memory and plenty of AO2 errors. This
change allows the remote native bridge loop to clean up properly when
the bridge fails.
(closes issue ASTERISK-23310)
Reported by: Jeremy Laine
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When two RTP channels are in a remote bridge, the remote bridging loop in
rtp_engine will periodically check to see if the two channels can still be
bridged. One of the many things it checks is whether or not the codecs have
changed on the channel. If the codec has changed, it will break out of the
loop to re-determine which type of bridge is appropriate.
In order to perform this check, the ast_rtp_glue virtual table's get_codec
callback is called for each channel. The callback implementations assume
that the channel tech private is valid when called; as such, there has
always been some code in place to check whether or not the channel pvt is
NULL before calling. However, this check is insufficient.
The channels are unlocked during the remote bridging loop. It is possible
for a channel to get masqueraded between the check for the pvt being NULL and
the actual call to get_codec. When this occurs, the callback is called with a
ZOMBIE channel, which now has a NULL pvt. Crash.
While this has always been possible in Asterisk 1.8, it is much more likely to
occur in Asterisk 11 and later versions due to the timing changes that occur
when getting the codec from a channel. Note that this is much more likely to be
reproduced on slow, boggy hardware running Asterisk 11 - but fairly rarely
otherwise.
Also Note: This crash was also caught by the various SIP blind transfer tests,
in addition to the bug report Alec filed.
Review: https://reviewboard.asterisk.org/r/3247/
(closes issue ASTERISK-21737)
Reported by: Alec Davis
Tested by: Alec Davis
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Asterisk didn't support the dynamic payload change in rtp mapping in the 200
OK response.
Scenario:
Asterisk sends the INVITE proposing alaw and telephone-event, it proposes
rtpmap:101 for telephone-event. Peer responds with 2xx, it answers with
alaw and telephone-event also, but it proposes a different rtpmap number
(rtpmap:103) for telephone-event.
Expected Behaviour:
Asterisk should honour the rtpmapping in the response and send DTMF packets
using 103 as payload type for DTMF.
Actual Behaviour: Asterisk sends DTMF packets using payload type 101.
With this patch asterisk now supports changes that can occur in the rtp mapping
in the response.
(closes issue ASTERISK-23279)
Reported by: NITESH BANSAL
Review: https://reviewboard.asterisk.org/r/3225/
Patches:
dynamic_payload_change.patch uploaded by nbansal (license 6418)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@408729 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Updated the code to check to see if MOH is playing on the transferor and if
so then start it on the channel that replaces it during a masquerade.
Example scenario of the problem:
Alice calls Bob and then Bob begins the attended transfer process into a queue.
Upon going on hold Alice hears music and so does Bob once he is in the queue.
Bob then transfers Alice into the queue and then music for Alice stops even
though she should be hearing it since has now replaced Bob in the queue.
The problem that was occurring is that once the channel was masqueraded the app
(queues, confbridge, etc...) had no way of knowing that the channel had just
been swapped out thus it did not start music for the present channel.
Credit to Olle Johansson for pointing me in the right direction on this issue.
(closes issue ASTERISK-19499)
Reported by: Timo Teräs
Review: https://reviewboard.asterisk.org/r/3226/
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Repeatedly modifying config files and reloading too fast sometimes fails
to reload the configuration because the cached modification timestamp has
one second resolution.
* Added file size and nanosecond resolution fields to the cached config
file modification timestamp information. Now if the file size changes or
the file system supports nanosecond resolution the modified file has a
better chance of being detected for reload.
* Added a missing unlock in an off-nominal code path.
(closes issue AST-1303)
Review: https://reviewboard.asterisk.org/r/3235/
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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
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In ast_format_sdp_parse and ast_format_sdp_generate
the check checks for a valid interface and function
were potentially confusing, and hid an error in the
test of the presence of the function that is called
later. This patch clears up and corrects the test.
Review: https://reviewboard.asterisk.org/r/3208/
(closes issue ASTERISK-23098)
Reported by: marcelloceschia
Patches:
main_format.patch uploaded by marcelloceschia (license 6036)
ASTERISK-23098.patch uploaded by coreyfarrell (license 5909)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@408137 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'.
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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
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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
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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/
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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/
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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/
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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.
As a result appset, linkedids, and held objects don't need a 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/
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* 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.
* Removed unnecessary container traversals of the white/black filters
during manager_free_user().
* ast_free() does not need a NULL check before calling.
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Occasionally, the manager module would get an "INTERNAL_OBJ: bad magic
number" error on a "core restart gracefully" command if an AMI connection
is established.
* Added ao2_global_obj protection to the sessions global container.
* Fixed the order of unreferencing a session object in session_destroy().
* Removed unnecessary container traversals of the white/black filters
during session_destructor().
(closes issue AST-1242)
Reported by: Guenther Kelleter
Review: https://reviewboard.asterisk.org/r/3144/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@406341 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)
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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)
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action_originate responds to the remote system with an error when cap==NULL,
but doesn't return (abort the originate). Patched to return.
(closes issue ASTERISK-23034)
Reported by: Corey Farrell
Patches:
ASTERISK-23034.patch uploaded by coreyfarrell (license 5909)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@405745 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The per console verbose level feature as previously implemented caused a
large performance penalty. The fix required some minor incompatibilities
if the new rasterisk is used to connect to an earlier version. If the new
rasterisk connects to an older Asterisk version then the root console
verbose level is always affected by the "core set verbose" command of the
remote console even though it may appear to only affect the current
console. If an older version of rasterisk connects to the new version
then the "core set verbose" command will have no effect.
* Fixed the verbose performance by not generating a verbose message if
nothing is going to use it and then filtered any generated verbose
messages before actually sending them to the remote consoles.
* Split the "core set debug" and "core set verbose" CLI commands to remove
the per module verbose support that cannot work with the per console
verbose level.
* Added a silent option to the "core set verbose" command.
* Fixed "core set debug off" tab completion.
* Made "core show settings" list the current console verbosity in addition
to the root console verbosity.
* Changed the default verbose level of the 'verbose' setting in the
logger.conf [logfiles] section. The default is now to once again follow
the current root console level. As a result, using the AMI Command action
with "core set verbose" could again set the root console verbose level and
affect the verbose level logged.
(closes issue AST-1252)
Reported by: Guenther Kelleter
Review: https://reviewboard.asterisk.org/r/3114/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@405431 65c4cc65-6c06-0410-ace0-fbb531ad65f3
'core show channeltypes' type column is being sliced, resulting in incomplete
type names.
(closes issue ASTERISK-22919)
Reported by: outtolunc
Patches:
svn_channel.c.format_15.diff.txt uploaded by outtolunc (license 5198)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@404579 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When Asterisk is shut down, the astdb_atexit() function releases
(finalize) the previously initiated (prepared) SQL statements in
sqlite3. Another thread making a subsequent request can cause a
crash in sqlite3. This patch eliminates that issue by resetting
the statement pointer after it is released/cleared. The sqlite3
code detects the null pointer, and aborts the operation cleanly.
(closes issue AST-1265)
Reported by: Alexander Hömig
(closes issue ASTERISK-22350)
Reported by: Birger "WIMPy" Harzenetter
Review: https://reviewboard.asterisk.org/r/3078/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@404344 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Richard Mudgett
Corey Farrell
Scott Griepentrog
Joshua Colp
Russ Meyerriecks
Jonathan Rose
Mark Michelson
Kinsey Moore
David M. Lee
Matthew Jordan
Kevin Harwell
Sean Bright
Russell Bryant
Walter Doekes
Rusty Newton
Tzafrir Cohen
Jason Parker