| Age | Commit message (Collapse) | Author |
|---|
|
|
|
|
|
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.
Sample pjproject.conf setting:
[startup]
cache_pools=no
* Made current users of the caching pool factory initialization and
destruction calls call common routines to create and destroy cached pools.
ASTERISK-27704
Change-Id: I64d5befbaeed2532f93aa027a51eb52347d2b828
|
|
|
|
Change-Id: I459dc320a8c9452a01eed6f403d786741587c890
|
|
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
ASTERISK-27610
Change-Id: I18aca01af98f87c1e168e6ae0d85c136d1df5ea9
|
|
* Fix --tarball-config so the option doesn't cause an error.
* Allow for missing /etc/os-release.
* Add a sleep between tarballing the coredump and removing the
output directory to allow the filesystem to settle.
Change-Id: I73e03b13087978bcc7f6bc9f45753990f82d9d77
|
|
A couple of additional properties are needed in rtp_engine to enable
support for packet retransmission: AST_RTP_PROPERTY_RETRANS_RECV and
AST_RTP_PROPERTY_RETRANS_SEND. These will both be enabled automatically
if an endpoint has the webrtc option enabled. While this adds no
functionality currently, it will serve as a building block for future
changes for RTP retransmission support.
For more information, refer to the wiki page:
https://wiki.asterisk.org/wiki/display/AST/WebRTC+User+Experience+Improvements
Change-Id: Ic598acd042a045f9d10e5bdccb66f4efc9e587cc
|
|
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.
ASTERISK-26688 #close
Change-Id: I80864a0d40d2e2d8cd79d72af52a8f0a3a99c015
|
|
|
|
|
|
|
|
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.
ASTERISK-27568
Change-Id: Iacda724e7cb24d7520e49b2fd7e504aa398d7238
|
|
|
|
deleted
autoconf gcc libsamplerate sqlite
changed
binutils to libbfd
freetds-devel to freetds
gmime2 to gmime26
mysql55-client to mysql57-client
added
alsa-lib bison bzip2 cclient corosync doxygen libedit flex graphviz
libhoard libical libilbc libltdl lua neon newt net-snmp
openldap-client openssl patch pkgconf portaudio postgresql10-client
python radcli speexdsp subversion uriparser xmlstarlet libzip
ASTERISK-27686
Change-Id: Ibe88c9b26e59c30d26cdb313a3ef01c9f37ac80d
|
|
Change-Id: Id52f719078a65c4b2eee7ab99d761eba6b6aed94
|
|
|
|
|
|
|
|
|
|
ASTERISK-27659
Change-Id: I04a2705d2cb7df250769967bc59e2b397a49b797
|
|
The handling of stream topologies was not protected by channel locks in
simple_bridge_request_stream_topology_change().
* Fixed topology handling to be protected by channel locks where needed in
simple_bridge_request_stream_topology_change().
ASTERISK-27692
Change-Id: Ica5d78a6c7ecf4f0b95fb16de28d3889b32c4776
|
|
|
|
|
|
|
|
passed up."
|
|
In ast_websocket_read() we were not adequately checking that the
payload_len was non-zero before passing it to ws_safe_read(). Calling
ws_safe_read with a len argument of 0 will result in a busy loop until
the underlying socket is closed.
ASTERISK-27658 #close
Change-Id: I9d59f83bc563f711df1a6197c57de473f6b0663a
|
|
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.
ASTERISK-27583 #close
Change-Id: I5cebe000ce2d846cae3af33b6d72c416e51caf2f
|
|
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.
ASTERISK-27582 #close
Change-Id: I883b3a4ef85b6972397f7b56bf46c5779c55fdd6
|
|
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.
ASTERISK-27618
Reported By: Sandro Gauci
Change-Id: I21a2d33d9dda001452ea040d350d7a075f9acf0d
|
|
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.
ASTERISK-27618
Reported By: Sandro Gauci
Change-Id: Iab33a6c7b6fba96148ed465b690ba8534ac961bf
|
|
It was discovered that there are some corner cases where a pjsip tsx
might have no last_tx so calling ast_sip_failover_request with
a NULL last_tx as its tdata would cause a crash.
ASTERISK-27618
Reported By: Sandro Gauci
Change-Id: Ic2b63f6d4ae617c4c19dcdec2a7a6156b54fd15b
|
|
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.
ASTERISK-27640
Reported By: Sandro Gauci
Change-Id: I99a814b10b554b13a6021ccf41111e5bc95e7301
|
|
When an RTP packet is received by an RTP engine it has to map the
payload into the Asterisk format. The code was incorrectly checking
our own static list for ALL payloads if it couldn't find a negotiated one.
This included dynamic payloads. If the payload mapped to a format
of a different type (for example receiving a video packet on an audio
RTP instance) then the core stream code could cause a crash if a legacy
channel driver was in use as no stream would be present.
To provide further protection the core stream code will no longer assume
that a video or audio frame will always have a stream for legacy channel
drivers. If no stream is present the frame is dropped.
ASTERISK-27488
Change-Id: I022556f524ad8379ee73f14037040af17ea3316a
|
|
|
|
|
|
Change-Id: I4eeedf89085697e81c354eb92d546686c67b0b5b
|
|
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.
ASTERISK-24488
Change-Id: I547a458fc59721c918cb48be060cbfc3c88bcf9c
|
|
|
|
|
|
|
|
|
|
|
|
OpenBSD."
|
|
This document is out of date and is superseded by content on the
Asterisk wiki.
ASTERISK-24386 #close
Change-Id: Idbf95b27b096c205251e1bbb560c79224ba81822
|
|
|
|
|
|
|
|
|
|
|
