Corey Farrell [Wed, 7 Mar 2018 06:29:13 +0000 (01:29 -0500)]
Replace direct checks of option_debug with DEBUG_ATLEAST macro.
Checking option_debug directly is incorrect as it ignores file/module
specific debug settings. This system-wide change replaces nearly all
direct checks for option_debug with the DEBUG_ATLEAST macro.
Jean Aunis [Wed, 14 Feb 2018 13:33:18 +0000 (14:33 +0100)]
chan_sip: Fix improper RTP framing on outgoing calls
The "ptime" SDP parameter received in a SIP response was not honoured.
Moreover, in the abscence of this "ptime" parameter, locally configured
framing was lost during response processing.
This patch systematically stores the framing information in the
ast_rtp_codecs structure, taking it from the response or from the
configuration as appropriate.
Alexander Traud [Wed, 7 Mar 2018 15:32:05 +0000 (16:32 +0100)]
utils: In Solaris, avoid a warning about an unused variable.
When HAVE_GETHOSTBYNAME_R_5 was set by the script ./configure, GCC 7.3.0 found
an unused variable. Actually, the variable was used (set to a dummy value) but
the compiler optimization might have removed that. Instead, this change ensures
that the variable 'res' is only used when it is really required.
Richard Mudgett [Tue, 27 Feb 2018 21:40:18 +0000 (15:40 -0600)]
pjproject: Add cache_pools debugging option.
The pool cache gets in the way of finding use after free errors of memory
pool contents. Tools like valgrind and MALLOC_DEBUG don't know when a
pool is released because it gets put into the cache instead of being
freed.
* Added the "cache_pools" option to pjproject.conf. Disabling the option
helps track down pool content mismanagement when using valgrind or
MALLOC_DEBUG. The cache gets in the way of determining if the pool
contents are used after free and who freed it.
To disable the pool caching simply disable the cache_pools option in
pjproject.conf and restart Asterisk.
Michael Cargile [Sat, 24 Feb 2018 03:24:10 +0000 (22:24 -0500)]
apps/app_amd.c: Fixed total time and silence calculations
Between Asterisk 11 and Asterisk 13 there was a significant increase
in the number of AST_FRAME_NULL frames being processed by app_amd.c's
main loop. Each AST_FRAME_NULL frame was being counted as 100ms
towards the total time and silence. This may have been accurate
when app_amd.c was orginally added, but it is not in Asterisk 13.
As such the total analysis time and silence calculations were way
off effectively breaking app_amd.c
* Additional debug messages were added
* AST_FRAME_NULL are now ignored
Corey Farrell [Fri, 23 Feb 2018 16:09:46 +0000 (11:09 -0500)]
core: Fix handling of maximum length lines in config files.
When a line is the maximum length "\n" is found at sizeof(buf) - 2 since
the last character is actually the null terminator. In addition if a
line was exactly 8190 plus a multiple of 8192 characters long the config
parser would skip the following line.
Additionally fix comment in voicemail.conf sample config. It previously
stated that emailbody can only contain up to 512 characters which is
always wrong. The buffer is normally 8192 characters unless LOW_MEMORY
is enabled then it is 512 characters. The updated comment states that
the line can be up to 8190 or 510 characters since the line feed and
NULL terminator each use a character.
Richard Mudgett [Thu, 22 Feb 2018 19:53:20 +0000 (13:53 -0600)]
res_pjsip_refer.c: Fix attended transfer race condition crash.
The transferrer's session channel was destroyed by the transferrer's
serializer thread in a race condition with the transfer target's
serializer thread during an attended transfer. The transfer target's
serializer was attempting to clean up a deferred end status on behalf of
the transferrer's channel when it should have passed the action to the
transferrer's serializer. When the transfer target's serializer lost the
race then both threads wind up trying to end the transferrer's session.
* Push the ast_sip_session_end_if_deferred() call onto the transferrer's
serializer to avoid a race condition that results in a crash. The
session_end() function that could be called by
ast_sip_session_end_if_deferred() really must be executed by the
transferrer's serializer to avoid this kind of crash.
Kevin Harwell [Thu, 22 Feb 2018 17:11:14 +0000 (11:11 -0600)]
AMI: Bumping AMI non-breaking number for Asterisk 13.20.0 release
A few changes were made to AMI:
* Fixed "(null):" header in AMI AsyncAGIEnd event
* A mute header was added to the ConfbridgeJoin AMI event
* ConfbridgeList action's ConfbridgeList events now output all
the standard channel snapshot headers
Kevin Harwell [Wed, 31 Jan 2018 19:37:54 +0000 (13:37 -0600)]
AST-2018-003: Crash with an invalid SDP fmtp attribute
pjproject's fmtp retrieval function failed to catch invalid fmtp attributes.
Because of this Asterisk would crash if given an SDP with an invalid fmtp
attribute.
When retrieving the format this patch now makes sure the fmtp attribute is
available. If not available it now returns an error status.
Kevin Harwell [Wed, 31 Jan 2018 19:33:16 +0000 (13:33 -0600)]
AST-2018-002: Crash with an invalid SDP media format description
pjproject's media format parsing algorithm failed to catch invalid values.
Because of this Asterisk would crash if given an SDP with a invalid media
format description.
When parsing the media format description this patch now properly parses the
value and returns an error status if it can't successfully parse/convert the
value.
George Joseph [Tue, 6 Feb 2018 18:07:18 +0000 (11:07 -0700)]
AST-2018-005: res_pjsip_transport_management: Move to core
Since res_pjsip_transport_management provides several attack
mitigation features, its functionality moved to res_pjsip and
this module has been removed. This way the features will always
be available if res_pjsip is loaded.
George Joseph [Tue, 6 Feb 2018 17:28:49 +0000 (10:28 -0700)]
AST-2018-005: Fix tdata leaks when calling pjsip_endpt_send_response(2)
pjsip_distributor:
authenticate() creates a tdata and uses it to send a challenge or
failure response. When pjsip_endpt_send_response2() succeeds, it
automatically decrements the tdata ref count but when it fails, it
doesn't. Since we weren't checking for a return status, we weren't
decrementing the count ourselves on error and were therefore leaking
tdatas.
res_pjsip_session:
session_reinvite_on_rx_request wasn't decrementing the ref count
if an error happened while sending a 491 response.
pre_session_setup wasn't decrementing the ref count if
while sending an error after a pjsip_inv_verify_request failure.
res_pjsip:
ast_sip_send_response wasn't decrementing the ref count on error.
Joshua Colp [Wed, 7 Feb 2018 14:09:14 +0000 (14:09 +0000)]
AST-2018-004: Restrict the number of Accept headers in a SUBSCRIBE.
When receiving a SUBSCRIBE request the Accept headers from it are
stored locally. This operation has a fixed limit of 32 Accept headers
but this limit was not enforced. As a result it was possible for
memory outside of the allocated space to get written to resulting
in a crash.
This change enforces the limit so only 32 Accept headers are
processed.
Joshua Colp [Tue, 20 Feb 2018 16:33:43 +0000 (12:33 -0400)]
chan_sip: Emit a second ringing event to ensure channel is found.
When constructing a dialog-info+xml NOTIFY message a ringing channel
is found if the state is ringing and further information is placed into
the message. Due to the migration to the Stasis message bus this did
not always work as expected.
This change raises a second ringing event in such a way to guarantee
that the event is received by chan_sip and another lookup is done to
find the ringing channel.
Thomas Guebels [Mon, 19 Feb 2018 10:21:30 +0000 (11:21 +0100)]
res_rtp_asterisk: Fix ICE candidate nomination
If the ICE role is not set right away, we might have a role conflict
that stays undetected and ICE finishing with successful tests and no
candidate nominated. This was introduced by ASTERISK-27088.
To avoid this, we set the role as soon as before but only if the ICE
state permits it: still checking and not yet nominating candidates or
completed.
Corey Farrell [Mon, 19 Feb 2018 09:57:45 +0000 (04:57 -0500)]
core: Rename sounds_index.c to sounds.c.
This will make the source filename match the 'module reload sounds'
command. This will allow conversion to a built-in module in Asterisk 16
without needing to redefine AST_MODULE.
Corey Farrell [Fri, 16 Feb 2018 23:58:35 +0000 (18:58 -0500)]
BuildSystem: Use single bootstrap.sh for Asterisk and menuselect.
This causes the root bootstrap.sh script to generate configure scripts
for both Asterisk and menuselect. This ensures that both configure
scripts are generated with the same version of autotools and avoids
situations where shared autoconf macros get modified without
regenerating the menuselect script.
projects / thirdparty / asterisk.git / log
