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This patch updates http_websocket and its corresponding implementation
with a pre-session established callback. This callback allows for
WebSocket server consumers to be notified when a WebSocket connection is
attempted, but before we accept it. Consumers can choose to reject the
connection, if their application specific logic allows for it.
As a result, this patch pulls out the previously private
websocket_protocol struct and makes it public, as
ast_websocket_protocol. In order to preserve backwards compatibility
with existing modules, the existing APIs were left as-is, and new APIs
were added for the creation of the ast_websocket_protocol as well as for
adding a sub-protocol to a WebSocket server.
In particular, the following new API calls were added:
* ast_websocket_add_protocol2 - add a protocol to the core WebSocket
server
* ast_websocket_server_add_protocol2 - add a protocol to a specific
WebSocket server
* ast_websocket_sub_protocol_alloc - allocate a sub-protocol object.
Consumers can populate this with whatever callbacks they wish to
support, then add it to the core server or a specified server.
ASTERISK-24988
Reported by: Joshua Colp
Change-Id: Ibe0bbb30c17eec6b578071bdbd197c911b620ab2
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Add missing return -1 when no endpoint name is specified.
ASTERISK-25086 #close
Reported by: snuffy
Change-Id: I9de76c2935a1f4e3f0cffe97a670106f5605e89e
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The config wizard was always pulling the first occurrence of
a variable from an ast_variable list but this gets the template
value from the list instead of any overridden value. This patch
creates ast_variable_find_last_in_list() in config.c and updates
res_pjsip_config_wizard to use it instead of
ast_variable_find_in_list. Now the overridden values, where they
exist, are used instead of template variables.
Updated test_config to test the new API.
ASTERISK-25089 #close
Reported-by: George Joseph <george.joseph@fairview5.com>
Tested-by: George Joseph <george.joseph@fairview5.com>
Change-Id: Ifa7ddefc956a463923ee6839dd1ebe021c299de4
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When the new Bridging API was implemented, the workspace variable
changed to a malloc'd string, causing sizeof() to always be 8 (char).
Revert back to stored on stack string for workspace.
ASTERISK-25090 #close
Change-Id: I51e610ae87371df771ce7693a955510efb90f8f7
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list" into 13
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The message channel is a special channel that doesn't actually process frames.
However, certain actions can cause frames to be placed in the channel's read
queue including the Hangup application which is called on the channel after
each message is processed. Since the channel will continually be reused for
many messages, it's necessary to flush these frames at some point.
ASTERISK-25083 #close
Reported by: Jonathan Rose
Change-Id: Idf18df73ccd8c220be38743335b5c79c2a4c0d0f
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When completing voicemail playback of a message in the 'INBOX', the
message gets moved to the 'Old' messages folder. Without this patch, if
the 'Old' folder is already at its set limit, then the 'INBOX' message will
simply be deleted. With this patch, the flag to delete the message will be
removed if the save_to_folder function indicates that the message could
not be moved due to a full folder.
ASTERISK-25082 #close
Reported by: Jonathan Rose
Review: https://gerrit.asterisk.org/#/c/448/
Change-Id: I2be440a09f42e2d06d50975c40d1ad7f836ecb3f
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Fix the alphabetic order added on ast_manager_register_struct. The order
for struct manager_action added is not working, this change fixes the
problem.
Change-Id: I149da0cd06c3c4445d7516cc303358e9f26f8b4b
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MySQL configuration engine contains a bug in require_mysql(). This
function is used for column type checking in tables. This bug only
affects DATETIME, DATE and FLOAT types.
It came from mixing the first condition (switch-case-like
if/then/else), to check the expected column type, with the second
condition, to check the actual column type against the expected column
type. Both conditions must be checked separately in order to avoid the
execution of the wrong block.
ASTERISK-18252 #comment This patch might fix the issue
Reported by: Gareth Blades
ASTERISK-25041 #close
Reported by: Alexandre Fournier
Tested by: Alexandre Fournier
Change-Id: I0b8bf7e68ab938be8e6525a249260cb648cb0bfa
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First byte of DTLS packet shall be in range 20-63, not 20-64. Refer to RFC
https://tools.ietf.org/html/rfc5764#section-5.1.2 for correct values.
Change-Id: Iae6fa0d72b37c36a27fe40686e0ae6fba3afec31
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If an ISDN call is hungup by both sides at the same time a crash could
happen.
* Added missing NULL checks for the owner channel after calling
pri_queue_pvt_cause_data() in two places. Code after those calls need to
check the owner channel pointer for NULL before use because
pri_queue_pvt_cause_data() needs to do deadlock avoidance to lock the
owner and the owner may get hung up.
ASTERISK-21893 #close
Reported by: Alexandr Gordeev
Change-Id: Ica3e266ebc7a894b41d762326f08653e1904bb9a
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MAKE_MENUSELECT currently sets CC to CC, which is the compiler for the
target platform. But menuselect is to be run on the build system, so
BUILD_CC needs to be used instead - like it was in the past, before the
recent changes (https://reviewboard.asterisk.org/r/4370/). This is the
patch for ASTERISK-25074.
ASTERISK-25074 #close
Reported by: Sebastian Kemper
Tested by: Sebastian Kemper
Change-Id: I8a2b1fc5deb6ad2b80f49baca35b1b13d468ebf8
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Currently you can 'apply' a wizard to an object type but the wizard
always goes at the end of the object type's wizard list. This patch
adds a new ast_sorcery_insert_wizard_mapping function that allows
you to insert a wizard anyplace in the list. I.E. You could
add a caching wizard to an object type and place it before all
wizards.
ast_sorcery_get_wizard_mapping_count and
ast_sorcery_get_wizard_mapping were added to allow examination
of the mapping list.
ast_sorcery_remove_mapping was added to remove a mapping by name.
As part of this patch, the object type's wizard list was converted
from an ao2_container to an AST_VECTOR_RW.
A new test was added to test_sorcery for this capability.
ASTERISK-25044 #close
Change-Id: I9d2469a9296b2698082c0989e25e6848dc403b57
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The code which reads asterisk.conf supports processing the debug
option with ast_true, but ast_true returns -1. This causes debug
to still be off, convert to 1 so debug will be on as requested.
ASTERISK-25042
Reported by: Corey Farrell
Change-Id: I3c898b7d082d914b057e111b9357fde46bad9ed6
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Use function PQescapeStringConn for escaping the name
of the table and schema instead of doing it manually.
Change-Id: I6709165e2d00463e9c813d24f17830ad4910b599
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Based on feedback from Corey Farrell and Y Ateya, a few new
macros have been added...
AST_VECTOR_REMOVE which takes a parameter to indicate if
order should be preserved.
AST_VECTOR_ADD_SORTED which adds an element to
a sorted vector.
AST_VECTOR_RESET which cleans all elements from the vector
leaving the storage intact.
Change-Id: I41d32dbdf7137e0557134efeff9f9f1064b58d14
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pbx_spool used to delete/move the call file upon successful outgoing
call completion, but did not delete it from in-memory list of files
(dirlist, used only when compiled with inotify/kqueue support).
That resulted in an extra attempt to process that filename after
retrytime seconds.
Then, if a new file with the same name appears that is scheduled
in future further than the completed one plus its retrytime,
then it gets executed earlier than expected.
This patch fixes remove_from_queue function to also remove the entry
from the dirlist.
ASTERISK-17069 #close
Reported by: Jeremy Kister
ASTERISK-24442 #close
Reported by: tootai
Change-Id: If9ec9b88073661ce485d6b008fd0b2612e49a28b
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The main IVR was playing demo-congrats. I've switched it over to the
basic-pbx-ivr-main file that we added in core sounds 1.4.27. This prompt
has Allison prompting the user with the actual IVR menu.
ASTERISK-24892 #close
Change-Id: Ifb749616ff8e156a1031ddaddfcc9244767a095d
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While trying to get WebRTC working with chan_pjsip, I was running
into the following error:
Attempted to set an invalid DTLS-SRTP configuration on RTP
instance...
Josh helpfully pointed out that res_srtp.so might not be loaded, and
sure enough, it wasn't. This patch adds a ERROR indiciating as much
to hopefully help others having a similar problem.
Change-Id: I13aa477b47b299876728a21b130998a0ea6cd19f
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Added the necessary lines to the Makefile and sounds.xml so we'll have the
Swedish sounds in all available formats in menuselect.
See also: Swedish sounds were added into the core sounds release 1.4.27.
ASTERISK-24744 #close
Reported by: Tove Hjelm
Tested by: Rusty Newton
Change-Id: Ib6f4fd177afd1667b2402735034001d4d055a908
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ERR_remove_state was deprecated with OpenSSL 1.0.0 and was replaced by
ERR_remove_thread_state. ERR_load_SSL_strings and ERR_load_BIO_strings were
called by SSL_load_error_strings already and got removed. These changes allow
OpenSSL forks like BoringSSL to be used with Asterisk.
ASTERISK-25043 #close
Reported by: Alexander Traud
patches:
asterisk_with_BoringSSL.patch uploaded by Alexander Traud (License 6520)
Change-Id: If1c0871ece21a7e0763fafbd2fa023ae49d4d629
(cherry picked from commit 247fef66537b59649e7571d64e2c574a106dbd65)
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Moved contrib/asterisk-ng-doxygen to doc/asterisk-ng-doxygen.in
Changed /Makefile to copy asterisk-ng-doxygen.in to
asterisk-ng-doxygen then modify it with version instead of
modifying asterisk-ng-doxygen directly. Updated clean
targets as well.
Updated /.gitignore and doc/.gitignore.
Change-Id: I38712d3e334fa4baec19d30d05de8c6f28137622
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termination" into 13
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* Added a lookbehind to one-line comment matcher to skip escaped
semicolons.
* Added support for block comments.
Change-Id: Id17dfaeda8ed4be572e8107a0c010066584aaee7
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The res_pjsip_exten_state module currently has a race condition between
processing the extension state callback from the PBX core and processing
the subscription shutdown callback from res_pjsip_pubsub. There is currently
no synchronization between the two. This can present a problem as while
the SIP subscription will remain valid the tree it points to may not.
This is in particular a problem as a task to send a NOTIFY may get queued
which will try to use the tree that may no longer be valid.
This change does the following to fix this problem:
1. All access to the subscription tree is done within the task that
sends the NOTIFY to ensure that no other thread is modifying or
destroying the tree. This task executes on the serializer for the
subscriptions.
2. A reference to the subscription serializer is kept to ensure it
remains valid for the lifetime of the extension state subscription.
3. The NOTIFY task has been changed so it will no longer attempt
to send a NOTIFY if the subscription has already been terminated.
ASTERISK-25057 #close
Reported by: Matt Jordan
Change-Id: I0b3cd2fac5be8d9b3dc5e693aaa79846eeaf5643
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description." into 13
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CPU" into 13
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transfer." into 13
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After using the new vector stuff for real I found...
A bug in AST_VECTOR_INSERT_AT that could cause a seg fault.
The callbacks needed to be closer to ao2_callback in behavior
WRT to CMP_MATCH and CMP_STOP behavior and the ability to return
a vector of matched entries.
A pre-existing issue with APPEND and REPLACE was also fixed.
I also added a new macro to test.h that acts like ast_test_validate
but also accepts a return code variable and a cleanup label. As well
as printing the error, it sets the rc variable to AST_TEST_FAIL and
does a goto to the specified label on error. I had a local version
of this in test_vector so I just moved it.
ASTERISK-25045
Change-Id: I05e5e47fd02f61964be13b7e8942bab5d61b29cc
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Creating a snoop channel in ARI and spying only on a single direction (in or
out) results in CPU utilization continually increasing until the CPU is fully
consumed. This occurs because frames are being put in the opposing direction's
slin factory queue, but not being removed.
Fixed the problem by always reading and disposing of frames from the opposite
queue of the direction selected.
ASTERISK-24938 #closes
Change-Id: I935bfd15f1db958f364d9d6b3b45582c0113dd60
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ASTERISK-25034
Reported by: Richard Mudgett
Change-Id: I1ff8f02124d2f4abd632a050da52c64285bb7f30
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into 13
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into 13
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A crash happens with this sequence of steps:
1) Party A is connected to party B.
2) Party B starts a DTMF attended transfer.
3) Party A hangs up while party B is dialing party C.
When party A hangs up the bridge that party A and party B are in is
dissolved and party B is kicked out of the bridge. When party B finishes
dialing party C he attempts to move to the new bridge with party C. Since
party B is no longer in a bridge the attempted move dereferences a NULL
bridge_channel pointer and crashes.
* Made the hold(), unhold(), ringing(), and the bridge_move() functions
tolerant of the channel not being in a bridge. The assertion that party B
is always in a bridge is not true if the bridged peer of party B hangs up
and dissolves the bridge. Being tolerant of not being in a bridge allows
the peer hangup stimulus to be processed by the FSM.
* Made the bridge_move() function return void since where the return value
for a failed move was checked generated a FSM coding ERROR message for a
normal off-nominal condition.
* Eliminated most uses of RAII_VAR in bridge_basic.c.
ASTERISK-25003 #close
Reported by: Artem Volodin
Change-Id: Ie2c1b14e5e647d4ea6de300bf56d69805d7bcada
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My 13 version of test_vector had an ASTERISK_REGISTER_FILE() macro
call at the top which is only supported in master. Once removed
builds are successful.
Change-Id: I7cac8b669bed6de543bbf4e2eec3cffc9741acdd
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This patch fixes EXITWITHTIMEOUT queue_log entry to always come with 3
parameters: position, original position and waiting time.
ASTERISK-25038 #close
Reported by: Etienne Lessard
Change-Id: I0c62045922e26bee2125e93aee1dee17eee79618
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