ast_sip_push_task_synchronous() did not necessarily execute the passed in
task under the specified serializer. If the current thread is any
registered pjsip thread then it would execute the task immediately instead
of under the specified serializer. Reentrancy issues could result if the
task does not execute with the right serializer.
The original reason ast_sip_push_task_synchronous() checked to see if the
current thread was a registered pjsip thread was because of a deadlock
with masquerades and the channel technology's fixup callback
(ASTERISK_22936). A subsequent masquerade deadlock fix (ASTERISK_24356)
involving call pickups avoided the original deadlock situation entirely.
The PJSIP channel technology's fixup callback no longer needed to call
ast_sip_push_task_synchronous().
However, there are a few places where this unexpected behavior is still
required to avoid deadlocks. The pjsip monitor thread executes callbacks
that do calls to ast_sip_push_task_synchronous() that would deadlock if
the task were actually pushed to the specified serializer. I ran into one
dealing with the pubsub subscriptions where an ao2 destructor called
ast_sip_push_task_synchronous().
* Split ast_sip_push_task_synchronous() into
ast_sip_push_task_wait_servant() and ast_sip_push_task_wait_serializer().
ast_sip_push_task_wait_servant() has the old behavior of
ast_sip_push_task_synchronous(). ast_sip_push_task_wait_serializer() has
the new behavior where the task is always executed by the specified
serializer or a picked serializer if one is not passed in. Both functions
behave the same if the current thread is not a SIP servant.
* Redirected ast_sip_push_task_synchronous() to
ast_sip_push_task_wait_servant() to preserve API for released branches.
ASTERISK_26806
Change-Id: Id040fa42c0e5972f4c8deef380921461d213b9f3
There is a problem when an INVITE-with-Replaces transfer targets a channel
in a ConfBridge. The transfer will unconditionally swap out the
ConfBridge channel. Unfortunately, the ConfBridge state will not be aware
of this change. Unexpected behavior will happen as a result since
ConfBridge channels currently can only be replaced by a masquerade and not
normal bridge channel moves.
* We just need to pretend that the channel isn't in a bridge (like other
transfer methods already do) so the transfer channel will masquerade into
the ConfBridge channel.
Change-Id: I209beb0e748fa4f4b92a576f36afa8f495ba4c82
Given the below call scenario:
A -> Ast1 -> B
C <- Ast2 <- B
1) A calls B through Ast1
2) B calls C through Ast2
3) B transfers A to C
When party B transfers A to C, B sends a REFER to Ast1 causing Ast1 to
send an INVITE with replaces to Ast2. Ast2 then leaks a channel ref of
the channel between Ast1 and Ast2.
Channel ref leaks are easily seen in the CLI "core show channels" output.
The leaked channels appear in the output but you can do nothing with them
and they never go away unless you restart Asterisk.
* Properly account for the channel refs when imparting a channel into a
bridge when handling an INVITE with replaces in handle_invite_replaces().
The ast_bridge_impart() function steals a channel ref but the code didn't
account for how many refs were held by the code at the time and which ref
was stolen.
* Eliminated RAII_VAR in handle_invite_replaces().
ASTERISK-27740
Change-Id: I7edbed774314b55acf0067b2762bfe984ecaa9a4
In the script ./configure, AST_EXT_LIB_CHECK checks for external libraries. Some
libraries do not specify all their dependencies and require additional shared
libraries. In AST_EXT_LIB_CHECK, this is the fifth parameter. However, if a
library is specified there, it must exist on the platform, because ./configure
tries to compile/link/execute a small app using those statements. For example,
the library libdl.so is Linux specific and does not exist on BSD-like platforms.
Furthermore, no supported platform/version was found, which still (ever?)
requires those additional libraries. Therefore, they were simply removed.
Finally, this change adds the error code ESTRPIPE to the channel driver
chan_alsa for those platforms which lack it, again for example NetBSD.
ASTERISK-27720
Change-Id: I3b21f2135f6cbfac7590ccdc2df753257f426e0b
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.
Change-Id: Ic342d4799a945dbc40ac085ac142681094a4ebf0
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.
ASTERISK-27674
Change-Id: I828a6a98d27a45a8afd07236a2bd0aa3cbd3fb2c
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
Check if initreq data string exists before using it when processing a
CANCEL request.
ASTERISK-27666
Change-Id: Id1d0f0fa4ec94e81b332b2973d93e5a14bb4cc97
It seems that the ALSA backend of PortAudio doesn't know how to both
read and write at the same time by adding a per-device mutex.
FIXME: currently only a draft version. Need to either auto-detect
we work with the ALSA backend or add an extra configuration option
to use this mutex.
ASTERISK-27426 #close
Change-Id: I635eacee45f5413faa18f5a3b606af03b926dacb
This patch fix chan_unistim hold functions to correctly support
hold function in different states possible in case of multiple lines
established on the phone
ASTERISK-26596 #close
Change-Id: Ib1e04e482e7c8939607a42d7fddacc07e26e14d4
A patch for sending in-dialog SIP NOTIFY message
with "SIPnotify" AMI action.
ASTERISK-27461
(created patch for 13 branch manually due to merge conflict)
Change-Id: I255067f02e2ce22c4b244f12134b9a48d210c22a
Per RFC 5245, the foundation specified with an ICE candidate can be up
to 32 characters but we are only allowing for 31.
ASTERISK-27498 #close
Reported by: Michele Prà
Change-Id: I05ce7a5952721a76a2b4c90366168022558dc7cf
Fix instances of:
* Retreive
* Recieve
* other then
* different then
* Repeated words ("the the", "an an", "and and", etc).
* othterwise, teh
ASTERISK-24198 #close
Change-Id: I3809a9c113b92fd9d0d9f9bac98e9c66dc8b2d31
In change_redirecting_information variables we use ast_strlen_zero to
see if a value should be saved. In the case where the value is not NULL
but is a zero length string we leaked.
handle_response_subscribe leaked a reference to the ccss monitor
instance.
Change-Id: Ib11444de69c3d5b2360a88ba2feb54d2c2e9f05f
chan_console supports multiple devices but the CLI only works on a
single device. 'console set active' selects this device.
Sadly that CLI picks the wrong command-line parameter and will only
work for a device called 'active'.
ASTERISK-27490 #close
Change-Id: I2f0e5fe63db19845bee862575b739360797dc73d
Some variables are set and never changed, making them constant. This
means that code in the 'false' block of the conditional is unreachable.
In chan_skinny and res_config_ldap I used preprocessor directive `#if 0`
as I'm unsure if the unreachable code could be enabled in the future.
Change-Id: I62e2aac353d739fb3c983cf768933120f5fba059
Attempting to dial PJSIP/endpoint when the endpoint doesn't exist and
disable_multi_domain=no results in a misleading empty endpoint name
message. The message should say the endpoint was not found.
* Added missing endpoint not found message.
* Added more information to the empty endpoint name msgs if available.
* Eliminated RAII_VAR in request().
Change-Id: I21da85ebd62dcc32115b2ffcb5157416ebae51e4
Log a message to security events when an INVITE is received to an
invalid extension.
ASTERISK-25869 #close
Change-Id: I0da40cd7c2206c825c2f0d4e172275df331fcc8f
Remove nearly all use of regex from ACO users. Still remaining:
* app_confbridge has a legitamate use of option name regex.
* ast_sorcery_object_fields_register is implemented with regex, all
callers use simple prefix based regex. I haven't decided the best
way to fix this in both 13/15 and master.
Change-Id: Ib5ed478218d8a661ace4d2eaaea98b59a897974b
Stripping the DNID in a SIP dial string can result in attempting to call
the argument parsing macros on an empty string, causing a crash.
ASTERISK-26131 #close
Reported by: Dwayne Hubbard
Patches:
dw-asterisk-master-dnid-crash.patch (license #6257) patch
uploaded by Dwayne Hubbard
Change-Id: Ib84c1f740a9ec0539d582b09d847fc85ddca1c5e
This patch does three things associated with the initial incoming INVITE
request URI.
1) Add access to the full initial incoming INVITE request URI.
2) We were not setting DNID on incoming PJSIP channels. The DNID is the
user portion of the initial incoming INVITE Request-URI. The value is
accessed by reading CALLERID(dnid).
3) Fix CHANNEL(pjsip,target_uri) documentation.
* The initial incoming INVITE request URI is now available using
CHANNEL(pjsip,request_uri).
* Set the DNID on PJSIP channel creation so CALLERID(dnid) can return the
initial incoming INVITE request URI user portion.
* CHANNEL(pjsip,target_uri) now correctly documents that the target URI is
the contact URI.
* Refactored print_escaped_uri() out of channel_read_pjsip() to handle
pjsip_uri_print() error condition when the buffer is too small.
ASTERISK-27478
Change-Id: I512e60d1f162395c946451becb37af3333337b33
There are many places in the code base where we ignore the return value
of fcntl() when getting/setting file descriptior flags. This patch
introduces a convenience function that allows setting or clearing file
descriptor flags and will also log an error on failure for later
analysis.
Change-Id: I8b81901e1b1bd537ca632567cdb408931c6eded7
The SuccessfulAuth using_password field was declared as a pointer to a
uint32_t when the field was later read as a uint32_t value. This resulted
in unnecessary casts and a non-portable field value reinterpret in
main/security_events.c:add_json_object(). i.e., It would work on a 32 bit
architecture but not on a 64 bit big endian architecture.
Change-Id: Ia08bc797613a62f07e5473425f9ccd8d77c80935
Previously, peers connected via TCP (or TLS) were matched by ignoring their
source port. One cannot say anything when protocol:IP:port match, yes (see
<http://stackoverflow.com/q/3329641>). However, when the ports do not match, the
peers do not match as well.
This change allows two peers connected to an Asterisk server via TCP (or TLS)
behind a NAT (= same source IP address) to be differentiated via their port as
well.
ASTERISK-27457
Reported by: Stephane Chazelas
Change-Id: Id190428bf1d931f2dbfd4b293f53ff8f20d98efa
chan_skinny creates a new thread for each new session. In trying
to be a good cleanup citizen, the threads are joinable and the
unload_module function does a pthread_cancel() and a pthread_join()
on any sessions that are active at that time. This has an
unintended side effect though. Since you can call pthread_join on a
thread that's already terminated, pthreads keeps the thread's
storage around until you explicitly call pthread_join (or
pthread_detach()). Since only the module_unload function was
calling pthread_join, and even then only on the ones active at the
tme, the storage for every thread/session ever created sticks
around until asterisk exits.
* A thread can detach itself so the session_destroy() function
now calls pthread_detach() just before it frees the session
memory allocation. The module_unload function still takes care
of the ones that are still active should the module be unloaded.
ASTERISK-27452
Reported by: Juan Sacco
Change-Id: I9af7268eba14bf76960566f891320f97b974e6dd
The media frame cache gets in the way of finding use after free errors of
media frames. Tools like valgrind and MALLOC_DEBUG don't know when a
frame is released because it gets put into the cache instead of being
freed.
* Added the "cache_media_frames" option to asterisk.conf. Disabling the
option helps track down media frame mismanagement when using valgrind or
MALLOC_DEBUG. The cache gets in the way of determining if the frame is
used after free and who freed it. NOTE: This option has no effect when
Asterisk is compiled with the LOW_MEMORY compile time option enabled
because the cache code does not exist.
To disable the media frame cache simply disable the cache_media_frames
option in asterisk.conf and restart Asterisk.
Sample asterisk.conf setting:
[options]
cache_media_frames=no
ASTERISK-27413
Change-Id: I0ab2ce0f4547cccf2eb214901835c2d951b78c00
When chan_sip receives a SUBSCRIBE request with no "Expires" header it
processes the request as an unsubscribe. This is incorrect, per RFC3264
when the "Expires" header is missing a default expiry should be used.
ASTERISK-18140
Change-Id: Ibf6dcd4fdd07a32c2bc38be1dd557981f08188b5
When sip.conf contains 'sipdebug=yes' it is impossible to disable it
using CLI 'sip set debug off'. This corrects the output of that CLI
command to instruct the user to turn sipdebug off in the configuration
file.
ASTERISK-23462 #close
Change-Id: I1cceade9caa9578e1b060feb832e3495ef5ad318
The sys/sysmacros.h include file does not exist in BSD systems and
is not required to build this module there.
Since an "#if defined(__NetBSD__) || defined(__FreeBSD__)" section
already exist I moved that include line inside it's #else branch.
ASTERISK-27343 #close
Change-Id: Ibfb64f4e9a0ce8b6eda7a7695cfe57916f175dc1
chan_vpb was trying to use sizeof(*p->play_dtmf), where
p->play_dtmf is defined as char[16], to get the length of the array
but since p->play_dtmf is an actual array, sizeof(*p->play_dtmf)
returns the size of the first array element, which is 1. gcc7
validly complains because the context in which it's used could
cause an out-of-bounds condition.
Change-Id: If9c4bfdb6b02fa72d39e0c09bf88900663c000ba
Currently privacy requests are only granted if the Privacy header
value is exactly "id" (defined in RFC 3325). It ignores any other
possible value (or a combination there of). This patch reverses the
logic from testing for "id" to grant privacy, to testing for "none" and
granting privacy for any other value. "none" must not be used in
combination with any other value (RFC 3323 section 4.2).
ASTERISK-27284 #close
Change-Id: If438a21f31a962da32d7a33ff33bdeb1e776fe56
