This patch prevents an infinite loop overwriting memory when
a message is received into the unpacksms16() function, where
the length of the message is an odd number of bytes.
(closes issue ASTERISK-22590)
Reported by: Jan Juergens
Tested by: Jan Juergens
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@403856 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Objects which are involved in SIP request creation and sending
now allow an outbound proxy to be specified. For cases where
an endpoint is used the outbound proxy specified there will
be applied.
(closes issue ASTERISK-22673)
Reported by: Antti Yrjola
Review: https://reviewboard.asterisk.org/r/3022/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@403811 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change adds an event for when an originated call is redirected to
another target. This event contains the original channel and the newly
created channel. If a stasis subscription exists on the original originated
channel for a stasis application then a new subscription will also be
created on the stasis application to the redirected channel. This allows
the application to follow the call path completely.
(closes issue ASTERISK-22719)
Reported by: Joshua Colp
Review: https://reviewboard.asterisk.org/r/3054/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@403808 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Rerunning a failed unit test after loading any required modules should
allow the test to report a pass status if it now passes.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@403782 65c4cc65-6c06-0410-ace0-fbb531ad65f3
There were still a few cases in which ATTENDEDTRANSFER and BLINDTRANSFER
wouldn't be set on channels involved with blind and attended transfers.
This would happen with features that were initialized by channel driver
specific mechanisms in multiparty calls. This patch resolves those cases
while attempted to keep the behavior for setting those variables as
consistent as possible.
(closes issue AFS-24)
Review: https://reviewboard.asterisk.org/r/3040/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@403781 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* The voicemail registration/unregistration functions now take a struct of
callbacks instead of a lengthy parameter list of callbacks.
* The voicemail registration/unregistration functions now prevent a
competing module from interfering with an already registered callback
supplying module.
........
Merged revisions 403643 from http://svn.asterisk.org/svn/asterisk/trunk
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@403779 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The change contains a slightly adjusted patch that was on the issue
(submitted by kmoore). A fix was made by adding in a bridge lock
while calling bridge_start/stop from the framehook callback. Since
the framehook callback is not called from the bridging core the bridge
is not locked, but needs to be before calling bridge_start.
(closes issue ASTERISK-22749)
Reported by: Kinsey Moore
Review: https://reviewboard.asterisk.org/r/3066/
Patches:
lock_inversion.diff uploaded by kmoore (license 6273)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@403767 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Added the ability to specify channel variables when creating/originating a
channel in ARI. The variables are sent in the body of the request and should
be formatted as a single level JSON object. No nested objects allowed.
For example: {"variable1": "foo", "variable2": "bar"}.
(closes issue ASTERISK-22872)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3052/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@403752 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Added the ability to have rules that are checked when adding and/or removing
channels to/from a bridge. In this case, if a channel is currently recording
and someone attempts to add it to a bridge an "is recording" rule is checked,
fails, and a 409 conflict is returned.
Also command functions now return an integer value that can be descriptive of
what kind of problems, if any, occurred before or during execution.
(closes issue ASTERISK-22624)
Reported by: Joshua Colp
Review: https://reviewboard.asterisk.org/r/2947/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@403749 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch was intended to eliminate a deadlock that occurs when
masquerades occur in pjsip channels, but has some potential side
effects. Mark Michelson is currently working on addressing this
problem from another angle.
(issue ASTERISK-22936)
Reported by: Jonathan Rose
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@403705 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In some cases messages need to be sent to a direct URI (sip:<ip address>). This
patch adds in that support by using a default outbound endpoint. When sending
messages, if no endpoint can be found then the default one is used.
To facilitate this a new default_outbound_endpoint option was added to the
globals section for pjsip.conf.
Review: https://reviewboard.asterisk.org/r/2944/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@403680 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds CHANNEL read support for chan_pjsip. This allows the dialplan
to use the CHANNEL function on a chan_pjsip channel to obtain run-time
information about the channel from the PJSIP channel driver and the PJSIP
stack. This includes:
* RTP information, including source/destination media addresses, whether or
not the media is secure, held, and other properties.
* RTCP information. This includes sets of parseable information, as well as
individual statistic attriutes.
* PJSIP information. This includes URIs, local/remote signalling addresses,
whether or not the signalling is secure, and other properties.
* The endpoint name. This can be used in conjunction with the PJSIP_ENDPOINT
function to obtain more detailed endpoint information.
Review: https://reviewboard.asterisk.org/r/3038/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@403618 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds a new function, PJSIP_ENDPOINT, which lets the dialplan query,
for any endpoint, any property configured on an endpoint. This function is a
companion to the CHANNEL function, which can be used to extract the endpoint
name for a channel.
Review: https://reviewboard.asterisk.org/r/3035
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@403616 65c4cc65-6c06-0410-ace0-fbb531ad65f3
res_sorcery_astdb.c: Fix get multiple records by regex.
* Fix sorcery_astdb_retrieve_regex() pattern matching. Let the regexec()
function match the stored key values instead of having astdb prefilter
them. Previoiusly you could only use a simple regex pattern when the
pattern began with '^'.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@403559 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Fix sorcery_astdb_retrieve_regex() pattern matching. Let the regexec()
function match the stored key values instead of having astdb prefilter
them. Previoiusly you could only use a simple regex pattern when the
pattern began with '^'.
* Fix off nominal memory leak in sorcery_astdb_retrieve_regex().
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@403545 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The snapshot process for endpoints uses the channel ids present
on the endpoint itself. Without keeping a reference it was possible
for the strings to be freed underneath any consumer of an endpoint
snapshot.
A reference is now held by the snapshot to the channel ids and
released when the snapshot is destroyed.
(issue ASTERISK-22801)
Reported by: Matt Jordan
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@403542 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Eliminated shadowing of the __ast_sorcery_apply_config() name parameter
causing confusion.
* Fix potential crash from sorcery.conf user input in
__ast_sorcery_apply_config() if the user supplied a malformed config line
that is missing the sorcery object type name.
* Remove redundant test in __ast_sorcery_apply_config(). !config and
config == CONFIGS_STATUS_FILEMISSING are identical.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@403541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Due to the way pjproject internally works it was possible for the
NAT module to not be invoked on messages with-in a session dialog.
This means that the various parts of the message would not get rewritten
with the source IP address and port.
This change uses a session supplement to add the NAT module
to the dialog on the first incoming or outgoing INVITE.
(closes issue ASTERISK-22941)
Reported by: Leif Madsen
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@403510 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Prior to this patch, res_fax_spandsp was conservative with how it initialized
the spandsp T.38 context. It would only initialize it if the driver thought
the current state was a T.38 fax. While this works fine in nominal situations,
in certain off nominal situations, res_fax_spandsp can believe that a T.38
fax will not occur when in fact one has started. In particular, this was
discovered when res_fax would fall back to audio after timing out on a T.38
upgrade. The SIP channel driver would continue to retry the re-INVITE and -
if the remote end responded after res_fax timed out with a 200 OK - a T.38
frame would be delivered to the res_fax stack when it no longer expected it.
As it turns out, there does not appear to be any downside to always
initializing the T.38 context, other than the actual memory allocation.
Since that avoids this off nominal situation (and others which are equally
likely hard to predict), this is the safest way to avoid this problem.
Much thanks to Torrey as well for providing a scenario that reproduces this
issue.
(closes issue ASTERISK-21242)
Reported by: Ashley Winters
Tested by: Torrey Searle
patches:
always-init-t38.patch uploaded by awinters (License 6477)
A_PARTY.xml uploaded by tsearle (License 5334)
........
Merged revisions 403449 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 403450 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@403458 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If the CDR unregistration fails due to an inflight CDR, the
res_config_sqlite module needs to bail on unloading itself. Otherwise,
the config could be unloaded (including the CDR table name) while the
CDR engine posts a CDR to the still registered backend, resulting in
a crash.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@403435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This crept up during gateway testing where the gateway would receive
the request to negotiate and assume it came from the remote side, causing
the gateway state machine to go a little, to a use a technical term,
"wonky".
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@403364 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Passing a non-zero value causes PJLIB to use the given input as the
hash value. Passing zero causes the parameter to become an output parameter
that receives the hash value that was computed based on the given key.
This change essentially makes ast_sip_dict_get() properly retrieve the
desired value.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@403349 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The code for getting channel variables from ARI assumed that you needed
to lock the channel in order to properly execute functions and read
channel variables. Apparently, this is not the case, since any dialplan
function that puts the channel into autoservice deadlocks when
attempting to remove the channel from autoservice.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@403342 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Newer versions of PJSIP have changed to using a flag for the
PJMEDIA_SDP_NEG_ALLOW_MEDIA_CHANGE instead of a define. This adds a
configure check to detect the presence of the flag and use it if found.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@403329 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Make ast_sorcery_observer_remove() accept a const callbacks struct.
* Make ast_sorcery_observer_remove() tolerant of the sorcery parameter
being NULL. Now it can be called within a module unload routine if the
sorcery initialization fails.
* Fix ast_sorcery_observer_add() to fail if the container link fails.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@403324 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Media indexing will now skip over files and directories that stat
will not return information about. This can occur under normal
conditions when a symbolic link points to a location that no longer
exists.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@403312 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This adds channel locks around calls to create channel snapshots as well
as other functions which operate on a channel and then end up
creating a channel snapshot. Functions that expect the channel to be
locked prior to being called have had their documentation updated to
indicate such.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@403311 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The changed value refers to the name of the module. The name of the
configuration file is specified in the configFile section.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@403310 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The check for determining whether the T.38 framehook should be added to
the channel or not has now been changed to guarantee adding only occurs
on the first incoming or outgoing INVITE.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@403258 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Transport type determination for security events has been simplified to use
the type present on the message itself instead of searching through configured
transports to find the transport used.
The actual WebSocket transport has also been simplified. It now leverages the
existing PJSIP transport manager for finding the active WebSocket transport
for outgoing messages. This removes the need for res_pjsip_transport_websocket
to store a mapping itself.
(closes issue ASTERISK-22897)
Reported by: Max E. Reyes Vera J.
Review: https://reviewboard.asterisk.org/r/3036/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@403256 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Depending on configuration it was possible for a media stream to be
created without any media formats. The produced SDP would fail internal
validation and cause a crash.
The code will now no longer add media streams with no formats to the SDP,
allowing it to pass validation and work.
(closes issue ASTERISK-22858)
Reported by: Anthony Messina
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@403223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When sending a re-INVITE to an endpoint it was possible for received
headers to be added as well (since they are stored for retrieval using
the PJSIP_HEADER dialplan function). This caused a broken (and
potentially large) SIP INVITE to be produced and sent.
This changes the module so it will no longer add headers to
re-INVITEs.
(closes issue ASTERISK-22882)
Reported by: David M. Lee
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@403221 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The action taken when a redirect occurs is now configurable on a
per-endpoint basis. The redirect can either be treated as a redirect
to a local extension, to a URI that is dialed through the Asterisk
core, or to a URI that is dialed within PJSIP itself.
(closes issue ASTERISK-21710)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2963/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@403207 65c4cc65-6c06-0410-ace0-fbb531ad65f3
