The supplied hash function to a container must be idempotent given the
object's key value to figure out which container bucket the object belongs
in. Returning a random number or the current container count is not
idempotent. The "computed hash" value doesn't help find the object later
in those cases.
* Fixed the format_list container to actually be a list since that is how
the container is used. Conceptually, if more than 283 formats were added
to the format_list then odd things may have happened before the fix.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@415728 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Replaced a stray echo that should've been a message call in
safe_asterisk. I'm using the contents of the old message inside the
if $NOTIFY so peoples log parsing scripts won't get confused by new
messages. I'll clean that up in trunk.
(Note that a 'make install' still won't overwrite your old safe_asterisk
if it exists. See ASTERISK-21965.)
ASTERISK-23492 #close
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@415522 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The twisted logic determining if a config file should be reloaded was
mostly broken and disabled. The incorrect test that ASTERISK-23383 fixed
actually reenabled the broken logic. The incorrect test was causing the
timestamp to always be cleared which caused config files with includes to
always be reloaded.
* Made wildcard includes always cause a reload. Determining if a file was
deleted cannot be determined without restructuring the cache to determine
if any files are missing from the last files actually loaded. Also
without refactoring config_text_file_load(), the glob loop couldn't check
more than one file for changes anyway.
* Made remove the cache entry if the file no longer exists when trying to
get its timestamp or it is no longer a regular file. This fixes the
corner case where the file was loaded, then deleted, then the config
reloaded, then the file restored with the same timestamp, and then the
config reloaded again.
* Made remove the cache entry include list when actually loading the file.
This gets rid of any stale includes the file had from the last time the
file was loaded.
ASTERISK-23683 #close
Reported by: tootai
Review: https://reviewboard.asterisk.org/r/3575/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@415229 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Prior to this patch, users waiting to enter a ConfBridge were not considered
when muted via the CLI or via AMI. Instead, a confusing message would be
emitted stating that the channel did not exist.
This patch allows a user to be muted when waiting to enter a ConfBridge
conference. This is equivalent to start when muted, only toggled via the CLI
or AMI.
Review: https://reviewboard.asterisk.org/r/3582
ASTERISK-23824 #close
patches:
rb3582.patch uploaded by tm1000 (License 6524)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@415206 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Cleans up the safe_asterisk script and adds the ASTSAFE_FOREGROUND
option that allows the debian asterisk init script to capture the
right pid.
* Drop the vim #modeline which wasn't used. Use test consistently
without the odd configure xno syntax. Double quote all paths.
General cleanup.
* Don't output message()s to the console but only to TTY if set.
* Allow TTY to be "no" as well as empty (debian compatibility with
debian/patches/safe_asterisk-config).
* Add option to export ASTSAFE_FOREGROUND=1 from the init script
that calls this to disable backgrounding. Debian uses a similar
method in debian/patches/safe_asterisk-nobg).
ASTERISK-23492 #close
Review: https://reviewboard.asterisk.org/r/3574/
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Conference names were not checked for maximum length, allowing unexpected
behaviour. This change adds checking to ensure the maximum length is not
exceeded. The maximum length is also changed from 32 to AST_MAX_EXTENSION.
ASTERISK-23035 #close
Reported by: Iñaki Cívico
Tested by: Iñaki Cívico
Patches:
confbridge-enforce_max-1.8.patch uploaded by coreyfarrell (license 5909)
confbridge-enforce_max-11up.patch uploaded by coreyfarrell (license 5909)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@415066 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The change that removed the fixed size buffers in odbc-related code --
removing arbitrary column width limits -- was incomplete. This change
adds: no segfault on writesql without insertsql and return value checks
after strdup.
While I was in the vicinity I cleaned up the linefeeds in the odbc
function descriptions, moved some code for clarity, removed some blobs
and noted (but didn't fix) that the 'odbc write ... exec' CLI command
doesn't behave as the dialplan equivalent when insertsql= is used.
#ASTERISK-23582 #close
Review: https://reviewboard.asterisk.org/r/3579/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@414998 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When invoking UpdateConfig AMI action with Action set to EmptyCat, Asterisk
will make all categories empty in the config but the one requested with a
Cat variable. This is due to a bug in ast_category_empty (main/config.c)
that makes an incorrect comparison for a category name.
This patch corrects the comparison such that only the requested category
is cleared.
Review: https://reviewboard.asterisk.org/r/3573/
ASTERISK-23803 #close
Reported by: zvision
patches:
manager.c.diff uploaded by zvision (License 5755)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@414881 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Dynamic and pattern matching hints should not be checked for their last
known state until they are instantiated by subscribers.
(closes issue AFS-56)
Reported by: John Hardin
Patch AFS-56-pbx.diff submitted by Matt Jordan (license 6283)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@414859 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Asterisk started counting the session timer at INVITE while the other
end correctly started at 200. This meant that for short session-expiries
(90 seconds) combined with long ringing times (e.g. 30 seconds), asterisk
would wrongly assume that the timer was hit before the other end thought
it was time to send a session refresh. This resulted in prematurely
ended calls.
This changes the session timer to start counting first at 200 like RFC
says it should.
(Also removed a few excess NULL checks that would never hit, because if
they did, asterisk would have crashed already.)
ASTERISK-22551 #close
Reported by: i2045
Review: https://reviewboard.asterisk.org/r/3562/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@414628 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The ODBC realtime driver uses ^NN parameter encoding to cope with the
special meaning of the semi-colon. A semi-colon in a field is
interpreted as if the key was supplied twice, something which isn't
otherwise possible with fixed database columns. E.g. allow=alaw;ulaw
is parsed as allow=alaw and allow=ulaw. A literal semi-colon is
rewritten to ^3B when stored in the database.
The module uses a stringfield to efficiently store the encoded
parameters. However, this stringfield wasn't always freed in some
off-nominal cases.
Commit r413241 fixed initialization so the encoding for INSERT and
DELETE queries wouldn't crash. (Only SELECTs and UPDATEs worked
apparently.) But that commit forgot the frees. This change cleans
that up.
Review: https://reviewboard.asterisk.org/r/3555/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@414565 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Occasionally, when the last marked user leaves the conference, waitmarked
users don't get MOH if MOH is supposed to be played while a waitmarked
user is waiting for another marked user.
* Made not interrupt MOH when the user is a waitmarked user. The
waitmarked user doesn't need to hear any leave announcements from the
conference as the user would have already heard different leave
announcements if they were enabled. Apparently DAHDI occasionally sends
unending non-silent streams to these users or a normal user still in the
conference has continuous high background noise. These non-silent streams
cause MOH to be suspended while the never ending "announcement" is played.
Issue caused by ASTERISK-13680.
AST-1349 #close
Reported by: Tyler Stewart
Review: https://reviewboard.asterisk.org/r/3543/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@414402 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Populate the CALLERID(ani2) value (and the special CALLINGANI2 channel
variable) with the ANI2 value in addition to the PRI specific ANI2 channel
variable.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@414050 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Starting a conference recording using the admin menu overwrites the DAHDI
conference data structure used to modify the admin user's conference mute
mode.
* Made no longer pass the user's DAHDI conference data structure into the
menu functions. The menu now uses its own DAHDI conference data
structure to start the recording channel.
* Moved the unlock conf->playlock to before playing the conf-full message.
No sense keeping the lock while that prompt is playing. The user is never
going to get into the conference at that point.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@413992 65c4cc65-6c06-0410-ace0-fbb531ad65f3
> Apparently this was already fixed in Asterisk 11.
> https://reviewboard.asterisk.org/r/1944/ (r368519, 2012-06-05 16:41:43 +0200)
........
chan_local+app_dial: Propagagate call answered elsewhere over local channels.
AST_FLAG_ANSWERED_ELSEWHERE was not propagated back from local channels.
It is now. That means that when a call is picked up from a callgroup of
local channels, the other channels will now properly see it as "picked up".
This occurs when you use a construct like Dial(Local/a@context&Local/b@context)
where a@context and b@context dial two chan_sip devices respectively. If one
device picks up, the other will not see "1 missed call" anymore. In this
respect, it now behaves the same as when doing Dial(SIP/a&SIP/b).
Review: https://reviewboard.asterisk.org/r/3540/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@413950 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When overlap dialing is enabled, the lack of inband audio available
information in the SETUP_ACKNOWLEDGE events causes an interoperability
problem with SIP. sig_pri doesn't know if there is dialtone present when
a SETUP_ACKNOWLEDGE is received so it assumes it is there and posts an
AST_CONTROL_PROGRESS frame. The SIP channel driver then sends out a 183
Session Progress and blocks the desired 180 Ringing message when the
ALERTING message comes in.
* Made the configure script detect if the installed version of libpri
supports the SETUP_ACKNOWLEDGE enhancements.
* Using the new API, made generate an AST_CONTROL_PROGRESS frame on an
incoming SETUP_ACKNOWLEDGE message when the message indicates inband audio
is present instead of assuming that dialtone is present.
* Using the new API, made SETUP_ACKNOWLEDGE send out an inband audio
available indication only if dialtone is expected. The change also makes
the fallback behaviour of sending the PROGRESS message better by sending
it only if dialtone is expected.
* Changed receiving a PROCEEDING message to not generate an
AST_CONTROL_PROGRESS frame if the progress indication ie indicates
non-end-to-end-ISDN. This helps interoperability with SIP.
* Changed sending a PROCEEDING message in response to an
AST_CONTROL_PROCEEDING frame to not indicate inband audio available. It
was silly to do so anyway because the channel driver doesn't know if
inband audio is even available. This helps interoperability with SIP.
This patch and a corresponding change in libpri work together to allow
Asterisk to control the inband audio available progress indication ie on
the SETUP_ACKNOWLEDGE message when dialtone is present.
AST-1338 #close
Reported by: Tyler Stewart
Review: https://reviewboard.asterisk.org/r/3521/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@413765 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If the barge audiohook was attached prior to the spyee and its peer
actually being bridged, the audiohook would not be applied and the
connected peer would not be able to hear audio from the spy when the
spy is in barge mode.
(closes issue ASTERISK-23381)
Reported by: Robert Moss
Review: https://reviewboard.asterisk.org/r/3505/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@413551 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The realtime API specifies that the store callback is supposed to return the number
of rows affected. res_config_pgsql was instead returning an Oid cast as an int, which
during any nominal execution would be cast to 0. Returning 0 when more than 0 rows were
inserted causes problems to the function's callers.
To give an idea of how strange code can be, this is the necessary code change to fix
a device state issue reported against chan_pjsip in Asterisk 12+. The issue was that
the registrar would attempt to insert contacts into the database. Because of the 0
return from res_config_pgsql, the registrar would think that the contact was not successfully
inserted, even though it actually was. As such, even though the contact was query-able
and it was possible to call the endpoint, Asterisk would "think" the endpoint was unregistered,
meaning it would report the device state as UNAVAILABLE instead of NOT_INUSE.
The necessary fix applies to all versions of Asterisk, so even though the bug reported
only applies to Asterisk 12+, the code correction is being inserted into 1.8+.
Closes issue ASTERISK-23707
Reported by Mark Michelson
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@413225 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Fixed early exit in sip_msg_send() not destroying the message iterator.
* Made ast_msg_var_iterator_next() and ast_msg_var_iterator_destroy()
tolerant of a NULL iter parameter in case ast_msg_var_iterator_init()
fails.
* Made ast_msg_var_iterator_destroy() clean up any current message data
ref.
* Made struct ast_msg_var_iterator, ast_msg_var_iterator_init(),
ast_msg_var_iterator_next(), ast_msg_var_unref_current(), and
ast_msg_var_iterator_destroy() use iter instead of i.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@413139 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This resolves a race condition where data could be written to a NULL
FILE pointer causing a crash as a websocket connection was in the
process of shutting down by adding locking to websocket session writes
and by deferring session teardown until session destruction.
(closes issue ASTERISK-23605)
Review: https://reviewboard.asterisk.org/r/3481/
Reported by: Matt Jordan
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@413123 65c4cc65-6c06-0410-ace0-fbb531ad65f3
On congested networks, it is possible for the DTLS handshake messages to get
lost. This patch adds a timer to res_rtp_asterisk that will periodically
check to see if the handshake has succeeded. If not, it will retransmit the
DTLS handshake.
Review: https://reviewboard.asterisk.org/r/3337
ASTERISK-23649 #close
Reported by: Nitesh Bansal
patches:
dtls_retransmission.patch uploaded by Nitesh Bansal (License 6418)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@413008 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Backport -r411687 and fix the fix because content_length is the length of
out plus the length of the file controlled by fd.
When a response has an out content length of 0, fwrite would be called to
write a buffer with no data in it. This resulted in the following classic
error message:
[Apr 3 11:49:17] ERROR[26421] http.c: fwrite() failed: Success
This patch makes it so that we only attempt to write the content of out if
the out string is non-zero.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@412923 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This adds the TCP_NODELAY option to accepted connections on the HTTP
server built into Asterisk. This option disables the Nagle algorithm
which controls queueing of outbound data and in some cases can cause
delays on receipt of response by the client due to how the Nagle
algorithm interacts with TCP delayed ACK. This option is already set on
all non-HTTP AMI connections and this change would cover standard HTTP
requests, manager HTTP connections, and ARI HTTP requests and
websockets in Asterisk 12+ along with any future use of the HTTP
server.
Review: https://reviewboard.asterisk.org/r/3466/
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