Kinsey Moore [Fri, 13 Jun 2014 13:08:32 +0000 (13:08 +0000)]
MoH: Don't restart stream on repeated start calls
Currently, music on hold will stop and then start again from the
beginning if ast_moh_start() is called multiple times. This can happen
if a call is put on hold repeatedly (the channel receives multiple
HOLD control frames) and can be triggered from ARI by starting MoH on a
channel multiple times. This is fairly jarring/annoying to users.
This change prevents MoH from being restarted if the requested music
class is the same as the one currently playing.
Corey Farrell [Thu, 12 Jun 2014 17:20:05 +0000 (17:20 +0000)]
chan_sip: DEBUG messages in sdp_crypto.c display despite a DEBUG level of zero
Change debug level for messages in sdp_crypto.c from zero to one. This
ensures the messages are not displayed when debugging is disabled. Change
does not apply to 12+ as it was already fixed in those versions.
ASTERISK-23246 #close
Reported by: Rusty Newton
Review: https://reviewboard.asterisk.org/r/3605/
........
Merged revisions 415908 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Richard Mudgett [Thu, 12 Jun 2014 16:22:19 +0000 (16:22 +0000)]
AST-2014-007: Fix DOS by consuming the number of allowed HTTP connections.
Simply establishing a TCP connection and never sending anything to the
configured HTTP port in http.conf will tie up a HTTP connection. Since
there is a maximum number of open HTTP sessions allowed at a time you can
block legitimate connections.
A similar problem exists if a HTTP request is started but never finished.
* Added http.conf session_inactivity timer option to close HTTP
connections that aren't doing anything. Defaults to 30000 ms.
* Removed the undocumented manager.conf block-sockets option. It
interferes with TCP/TLS inactivity timeouts.
* AMI and SIP TLS connections now have better authentication timeout
protection. Though I didn't remove the bizzare TLS timeout polling code
from chan_sip.
* chan_sip can now handle SSL certificate renegotiations in the middle of
a session. It couldn't do that before because the socket was non-blocking
and the SSL calls were not restarted as documented by the OpenSSL
documentation.
* Fixed an off nominal leak of the ssl struct in
handle_tcptls_connection() if the FILE stream failed to open and the SSL
certificate negotiations failed.
The patch creates a custom FILE stream handler to give the created FILE
streams inactivity timeout and timeout after a specific moment in time
capability. This approach eliminates the need for code using the FILE
stream to be redesigned to deal with the timeouts.
This patch indirectly fixes most of ASTERISK-18345 by fixing the usage of
the SSL_read/SSL_write operations.
ASTERISK-23673 #close
Reported by: Richard Mudgett
........
Merged revisions 415841 from http://svn.asterisk.org/svn/asterisk/branches/1.8
app_queue: delayed state can cause early leavewhenempty ringing
In app_queue, device state changes arrive in event messages and
update the queue member status value. That value is checked in
get_member_status() to decide that the caller should leave when
there are no available members. Although event messages can be
delayed by other activity, there is no adverse affect by lagged
status except in one specific case: there is only one available
member, it was just rung, and leavewhenempty is enabled set for
ringing members. This change adds a direct check of the device
state only under this condition where the caller may be dropped
incorrectly, resolving this issue without affecting performance
of app_queue normally.
AST-1248 #close
Review: https://reviewboard.asterisk.org/r/3595/
Reported by: Thomas Arimont
........
Merged revisions 415833 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Jonathan Rose [Thu, 12 Jun 2014 15:22:02 +0000 (15:22 +0000)]
MixMonitor: Add class authorization requirements to MixMonitor AMI commands
MixMonitor AMI commands StartMixMonitor and StopMixMonitor lacked class
authorization. StopMixMonitor now requires that the manager user either have
the call or system class authorization. StartMixMonitor is a slightly larger
issue since it can execute shell commands if the right arguments are passed
into it, and we consider this a permission escalation. A security release
will be issued for problem this shortly.
Richard Mudgett [Wed, 11 Jun 2014 22:44:30 +0000 (22:44 +0000)]
format.c: Fix misuse of hash container function.
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.
Walter Doekes [Mon, 9 Jun 2014 11:57:09 +0000 (11:57 +0000)]
safe_asterisk: Cleanup additions to r415132.
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
........
Merged revisions 415521 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Corey Farrell [Mon, 9 Jun 2014 03:47:11 +0000 (03:47 +0000)]
autoservice: stop thread on graceful shutdown
This change adds thread shutdown to autoservice for graceful shutdowns only.
ast_register_cleanup is backported to 1.8 to allow this. The logger callid
is also released on shutdown in 11+.
Richard Mudgett [Thu, 5 Jun 2014 17:45:24 +0000 (17:45 +0000)]
config: Fix config files not reloading when only an included file changes.
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.
Matthew Jordan [Thu, 5 Jun 2014 14:32:38 +0000 (14:32 +0000)]
app_confbridge: Allow muting of users waiting to enter a ConfBridge
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)
Walter Doekes [Wed, 4 Jun 2014 20:12:36 +0000 (20:12 +0000)]
safe_asterisk: Cleanup and debian compatibility.
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).
Corey Farrell [Wed, 4 Jun 2014 07:20:22 +0000 (07:20 +0000)]
app_confbridge: Correct verification of conference name length
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)
........
Merged revisions 415060 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Walter Doekes [Tue, 3 Jun 2014 07:32:30 +0000 (07:32 +0000)]
func_odbc: Fix fixed size buffers fix (r414968).
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.
Matthew Jordan [Fri, 30 May 2014 11:59:02 +0000 (11:59 +0000)]
main/config.c: AMI action UpdateConfig EmptyCat clears all categories
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.
Walter Doekes [Tue, 27 May 2014 21:19:26 +0000 (21:19 +0000)]
chan_sip: Start session timer at 200, not at INVITE.
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.)
Walter Doekes [Tue, 27 May 2014 19:46:48 +0000 (19:46 +0000)]
res_config_odbc: Fix old and new ast_string_field memory leaks.
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.
Richard Mudgett [Thu, 22 May 2014 15:50:38 +0000 (15:50 +0000)]
app_meetme: Don't interrupt MOH for waitmarked users.
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.
Richard Mudgett [Wed, 21 May 2014 22:05:53 +0000 (22:05 +0000)]
chan_local: Only block media frames when a generator is on both ends of a local channel.
The fix for ASTERISK-12292 was a bit too aggressive. You could have
generators pointed at each other on local channels but need to get other
kinds of frames such as DTMF or CONNECTED_LINE frames accross.
........
Merged revisions 414269 from http://svn.asterisk.org/svn/asterisk/branches/1.8
pbx.c: prevent potential crash from recursive replace()
Recurisve usage of replace() resulted in corruption of the
temporary string storage and potential crash. By changing
the string to be allocated separtely per instance, this is
eliminated.
ASTERISK-23650 #comment Reported by: Roel van Meer
ASTEIRSK-23650 #close
Richard Mudgett [Fri, 16 May 2014 17:23:42 +0000 (17:23 +0000)]
sig_pri.c: Pull the pri_dchannel() PRI_EVENT_RING case into its own function.
* Populate the CALLERID(ani2) value (and the special CALLINGANI2 channel
variable) with the ANI2 value in addition to the PRI specific ANI2 channel
variable.
Richard Mudgett [Thu, 15 May 2014 21:44:34 +0000 (21:44 +0000)]
app_meetme: Fix overwrite of DAHDI conference data structure.
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.
........
Merged revisions 413991 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Walter Doekes [Thu, 15 May 2014 15:51:42 +0000 (15:51 +0000)]
Blocked revisions 413949
> 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).
Walter Doekes [Wed, 14 May 2014 15:31:27 +0000 (15:31 +0000)]
res_musiconhold: Minor cleanup.
Fix a few free()'s that should be ast_free()'s. Reverted an old
workaround that isn't necessary. Reorder a tiny bit of code.
Remove a bit of commented-out code.
Walter Doekes [Tue, 13 May 2014 13:32:46 +0000 (13:32 +0000)]
rtp: Fix case typo in H263+ mime.
http://tools.ietf.org/html/rfc3555#section-4.2.6 says the canonical
mime subtype is "H263-1998", not "h263-1998". Original code was added
in r183101 on 2009-03-19 02:26:50 +0100.
This fixes issues with Polycom phones.
ASTERISK-23665 #close
ASTERISK-23665 #comment Patch r3529.patch uploaded by Guillaume Maudoux, backported by me.
Review: https://reviewboard.asterisk.org/r/3529/
........
Merged revisions 413787 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Richard Mudgett [Mon, 12 May 2014 23:48:13 +0000 (23:48 +0000)]
chan_dahdi/sig_pri: Prevent unnecessary PROGRESS events when overlap dialing is enabled.
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.
Kinsey Moore [Fri, 9 May 2014 22:28:40 +0000 (22:28 +0000)]
Allow Asterisk to compile under GCC 4.10
This resolves a large number of compiler warnings from GCC 4.10 which
cause the build to fail under dev mode. The vast majority are
signed/unsigned mismatches in printf-style format strings.
........
Merged revisions 413586 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Jonathan Rose [Fri, 9 May 2014 16:10:14 +0000 (16:10 +0000)]
app_chanspy: Fix a bug where Barge mode could fail
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/
Mark Michelson [Fri, 2 May 2014 19:50:07 +0000 (19:50 +0000)]
Return the number of rows affected by a SQL insert, rather than an object ID.
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
........
Merged revisions 413224 from http://svn.asterisk.org/svn/asterisk/branches/1.8
* 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.
Kinsey Moore [Wed, 30 Apr 2014 13:04:14 +0000 (13:04 +0000)]
Websocket: Add session locking and delay close
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
Matthew Jordan [Fri, 25 Apr 2014 17:47:21 +0000 (17:47 +0000)]
res_rtp_asterisk: Add support for DTLS handshake retransmissions
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.
Richard Mudgett [Wed, 23 Apr 2014 17:51:19 +0000 (17:51 +0000)]
http: Fix spurious ERROR message in responses with no content.
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:
Kinsey Moore [Mon, 21 Apr 2014 16:13:36 +0000 (16:13 +0000)]
HTTP: Add TCP_NODELAY to accepted connections
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.
Jonathan Rose [Mon, 21 Apr 2014 15:51:40 +0000 (15:51 +0000)]
chan_sip: Add sendrpid trust options
In r411189, some behavior was changed which made sendrpid behavior
act in a more trusting manner by sending full user data for peers
set with private caller presence in P-Asserted-Identity headers.
Since this changed long time expected behaviors, we decided to pull
that patch when that was pointed out by the community. Instead, this
patch provides a trust_id_outbound setting which will expose the data
per RFC-3325 if set to 'yes' and simply not send the PAI/RPID headers
at all if set to 'no'. By default trust_id_outbound will be set to
'legacy' which will preserve the behavior prior to these patches.
Extra special thanks to Walter Doekes for providing advice and
feedback.
(closes issue AST-1301)
(closes issue ASTERISK-19465)
Reported by: Krzysztof Chmielewski
Fix wrong dialtone. The "modulation" should not be referenced for tone+tone as it refers to the on-off characteristic - this often resulted in a single tone rather than the multitone as in the UK.
Matthew Jordan [Sat, 19 Apr 2014 01:02:08 +0000 (01:02 +0000)]
app_sms: Fix uninitialized values; hangup channel when REL is sent successfully
This patch fixes two issues in app_sms:
(1) Firstly, the 'flags' field on the stack in sms_exec() is uninitialised,
causing it to use the wrong protocol in some cases. This patch correctly
initializes the flags fields.
(2) Secondly, when disconnect supervision is not working or
inbanddisconnect=yes is set in chan_dahdi.conf, app_sms was failing to
terminate the call after it sent the REL(ease) message and the peer stopped
talking to it. This patch fixes the code to handle the 'bad stop bit'
message more gracefully in that case, and hang up the call.
Review: https://reviewboard.asterisk.org/r/1392/
ASTERISK-18331 #close
Reported by: David Woodhouse
patches:
asterisk-fix-sms.patch uploaded by David Woodhouse (License 5754)
........
Merged revisions 412655 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Rusty Newton [Fri, 18 Apr 2014 17:15:27 +0000 (17:15 +0000)]
sounds: Fix Sounds Makefile and XML that didn't support new sound prompt sets
In sounds/Makefile
1 Adds and moves some lines necessary for the en_GB core set. I'm just following how the other sets are defined here.
2 removes the ES extra sounds related lines as we don't have ES extra sound sets.
In sounds/sounds.xml
3 Adds member definitons for EN_AU, EN_GB, IT for core sound sets, and EN_GB in extra sound sets
Matthew Jordan [Thu, 17 Apr 2014 20:06:11 +0000 (20:06 +0000)]
main/Makefile: Fix build failure on SmartOS/Illumos/SunOS
This patch fixes two issues when building on SmartOS:
- channels/chan_oss.c: it makes sure soundcard.h is found
- main/Makefile: only use "-Wl,--version-script" when GNU LD is used as the Sun
Linker doesn't support that. Similar checks are already used elswhere in the
Makefile
Review: https://reviewboard.asterisk.org/r/3426
ASTERISK-23576 #close
Reported by: Sebastian Wiedenroth
patches:
fix-sunos.diff uploaded by Sebastian Wiedenroth (License 6597)
chan_sip: Send real CallerID information with P-Assserted-Identity (RFC-3325)
Prior to this patch, the P-Asserted-Identity header would include anonymous
caller id information which seems to go against the point of the
P-Asserted-Identity header. Now the real caller ID information will be
included in this header. Also, no privacy header would be included.
This patch adds 'Privacy: id' to outgoing SIP messages that include the
P-Asserted-Identity header.
(closes issue AST-1301)
---
........
Merged revisions 412328 from http://svn.asterisk.org/svn/asterisk/branches/1.8
autoservice acquires a local reference to the logger callid of each channel
in a loop. This local reference was not released, causing the callid of
every channel in autoservice to leak. This change moves the callid unref
inside the loop.
Matthew Jordan [Fri, 11 Apr 2014 02:10:22 +0000 (02:10 +0000)]
main/astobj2: Make REF_DEBUG a menuselect item; improve REF_DEBUG output
This patch does the following:
(1) It makes REF_DEBUG a meneselect item. Enabling REF_DEBUG now enables
REF_DEBUG globally throughout Asterisk.
(2) The ref debug log file is now created in the AST_LOG_DIR directory.
Every run will now blow away the previous run (as large ref files
sometimes caused issues). We now also no longer open/close the file
on each write, instead relying on fflush to make sure data gets written
to the file (in case the ao2 call being performed is about to cause a
crash)
(3) It goes with a comma delineated format for the ref debug file. This
makes parsing much easier. This also now includes the thread ID of the
thread that caused ref change.
(4) A new python script instead for refcounting has been added in the
contrib/scripts folder.
Richard Mudgett [Tue, 8 Apr 2014 21:20:09 +0000 (21:20 +0000)]
Internal timing: Add notice that the -I and internal_timing option are no longer needed.
Add notice messages during execution that the -I command line option and
the astersik.conf internal_timing option are no longer needed. The
internal timing functionality is now always enabled if there is a timing
module loaded.
NOTE: Since the command line options and the asterisk.conf config file are
processed before the logging system is initialized, the messages are
output to stderr.
Change requested as a result of asterisk-dev list comments about the
commit for ASTERISK-22846 that removed the -I and internal_timing options.
Richard Mudgett [Fri, 4 Apr 2014 18:46:18 +0000 (18:46 +0000)]
internal_timing: Remove the option and always make it enabled if a timing module is loaded.
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-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+.
app_queue: Fix a bug where realtime members would be deleted during reload causing waiting callers to get ejected.
This patch causes realtime queue members to remain in queues during the reload process. Previously these
members would be removed causing any waiting callers to be ejected from the queue with a reason of "EXITEMPTY".
Alexandr Anikin [Fri, 28 Mar 2014 17:44:11 +0000 (17:44 +0000)]
process stack command even if gatekeeper client isn't register
don't destroy gatekeeper client if it is not started
don't destroy gatekeeper client in some sort of gatekeeper errors
signal rtp create condition when call cleared before rtp structure created
http: response body often missing after specific request
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.
Matthew Jordan [Fri, 28 Mar 2014 04:27:02 +0000 (04:27 +0000)]
res_config_odbc/res_odbc: Fix handling of non-text columns updates with empty values.
This patch fixes setting nullable integer columns to NULL instead of an empty
string, which fails for PostgreSQL, for example. The current code is supposed
to do so, but the check is broken. The patch also allows the first column in
the list to be a nullable integer.
This patch also adds a compatibility setting in res_odbc.conf,
allow_empty_string_in_nontext. It is enabled by default. It should be disabled
for database backends (such as PostgreSQL) that require NULL instead of an
empty string for Integer columns.
Matthew Jordan [Fri, 28 Mar 2014 03:51:34 +0000 (03:51 +0000)]
chan_sip: Add MESSAGE request to allowed methods
The allowed methods advertised by chan_sip did not previously note the MESSAGE
request. Even in Asterisk 1.8, we do accept in-dialog MESSAGE requests; we
should advertise that we support MESSAGE requests.
ASTERISK-23504 #close
ASTERISK-23504 #comment Reported by: Martin Kontsek
ASTERISK-23504 #comment Patch sip.h_patch.diff uploaded by Martin Kontsek (license 6587)
Joshua Colp [Wed, 26 Mar 2014 22:44:11 +0000 (22:44 +0000)]
say: Fix a bug where SayNumber in Polish tries to play incorrect sound.
This change fixes a bug where calling SayNumber with a number divisible by
100 using the Polish language would cause the code to attempt to play a
sound file with an empty name.
Jonathan Rose [Wed, 26 Mar 2014 15:57:36 +0000 (15:57 +0000)]
chan_sip: Send real CallerID information with P-Assserted-Identity (RFC-3325)
Prior too this patch, the P-Asserted-Identity header would include anonymous
caller id information which seems to go against the point of the
P-Asserted-Identity header. Now the real caller ID information will be
included in this header. Also, no privacy header would be included.
This patch adds 'Privacy: id' to outgoing SIP messages that include the
P-Asserted-Identity header.
(closes issue AST-1301)
........
Merged revisions 411189 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Kinsey Moore [Tue, 25 Mar 2014 15:52:55 +0000 (15:52 +0000)]
chan_sip: Fix incorrect use of timers
If update_provisional_keepalive() is called while
send_provisional_keepalive_full() is waiting on the PVT lock, then
pvt->provisional_keepalive_sched_id will be changed to a new sched_id
value by update_provisional_keepalive(), but that new sched_id then may
be overwritten with -1 by send_provisional_keepalive_full(), killing
the pvt's reference to a schedule and "leaking" the reference.
(closes issue ASTERISK-22079)
Review: https://reviewboard.asterisk.org/r/3368/
Reported by: Jamuel Starkey, Matteo, Leif Madsen, Steve Davies
Patches:
provisional_keepalive_fix.diff uploaded by Steve Davies (license 5012)
........
Merged revisions 411088 from http://svn.asterisk.org/svn/asterisk/branches/1.8
g711_free() was introduced in spandsp 0.0.6pre4 and g711_release() became a
noop. I opted not to remove the call to g711_release() since it is harmless
and to call g711_free() if we have a sufficiently recent version of spandsp.
(issue ASTERISK-20149)
Reported by: Alexandr Gordeev
Russ Meyerriecks [Mon, 17 Mar 2014 21:19:49 +0000 (21:19 +0000)]
callerid: Logic error in checksum processing
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.
Mark Michelson [Fri, 14 Mar 2014 15:56:43 +0000 (15:56 +0000)]
Prevent delayed astdb syncs.
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)
Richard Mudgett [Wed, 12 Mar 2014 18:35:14 +0000 (18:35 +0000)]
app_confbridge: Make explicitly stop MOH if a user is kicked or hangs up while MOH is playing.
When MOH is playing to a user in a conference and the user is kicked or
hangs up from the conference then the AMI MusicOnHoldStop events didn't
happen. (Asterisk v11 AMI event: MusicOnHold, state:Stop)
(closes issue ASTERISK-23311)
Reported by: Benjamin Keith Ford
Richard Mudgett [Mon, 10 Mar 2014 17:09:42 +0000 (17:09 +0000)]
AST-2014-001: Stack overflow in HTTP processing of Cookie headers.
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
........
Merged revisions 410380 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Kinsey Moore [Mon, 10 Mar 2014 13:18:55 +0000 (13:18 +0000)]
AST-2014-002: chan_sip: Exit early on bad session timers request
This change allows chan_sip to avoid creation of the channel and
consumption of associated file descriptors altogether if the inbound
request is going to be rejected anyway.
Matthew Jordan [Fri, 7 Mar 2014 04:38:47 +0000 (04:38 +0000)]
chan_sip: Allow static realtime members to be qualified during module load.
When a static realtime peer with qualify=yes is loaded, Asterisk will fail to
send an OPTIONS request due to the lastms being equal to 0. This results in
the peer being unable to receive calls from Asterisk because the status is
permanently UNKNOWN.
This patch allows an OPTIONS request to be sent during module load by
ignoring the lastms value on startup only.
Review: https://reviewboard.asterisk.org/r/3294/
(closes issue ASTERISK-17523)
Reported by: Maciej Krajewski
Tested by: wushumasters
patches:
realtime_fix_11.7.0.txt uploaded by Trevor Peirce (license 6112)
........
Merged revisions 410105 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Russell Bryant [Thu, 6 Mar 2014 23:15:42 +0000 (23:15 +0000)]
moh: fix a refcount error with realtime MOH
I observed a crash in res_musiconhold on an Asterisk 11 system using realtime
MOH. Investigation of the backtrace showed a corrupt mohclass, implying that
it got destroyed before the code expected it to. I went looking for reference
counting errors that could have caused this crash and this patch this result.
It contains 2 changes.
1) Remove a usless block of code that was impossible to reach. There was even
a comment indicating that it was impossible to reach. The conditional includes
"!ast_test_flag(global_flags, MOH_CACHERTCLASSES)" and it's inside of an if
block with the opposite check "ast_test_flag(global_flags,
MOH_CACHERTCLASSES)". There's no good reason to keep it around.
2) A similar block to #1 contained a reference counting error. It stores
state->class in the local variable mohclass without increasing its reference
count. The reference count on mohclass is decremented at the end of the
function. This block of code probably very rarely runs, which would help
explain why this system was working fine for many months before experiencing a
crash.
projects / thirdparty / asterisk.git / log
