Kevin Harwell [Fri, 13 Dec 2013 16:32:53 +0000 (16:32 +0000)]
ARI: Adding a channel to a bridge while a live recording is active blocks
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.
realtime: Create extensions in alembic ast-db-manage contribution
When the alembic scripts were written for creating Asterisk
realtime databases the extensions table for dialplan wasn't
included. This update creates the extensions table.
(closes issue ASTERISK-22815)
Reported by: Zone Conkle
Review: https://reviewboard.asterisk.org/r/3064/
Jonathan Rose [Thu, 12 Dec 2013 19:12:00 +0000 (19:12 +0000)]
chan_pjsip: Revert r403587
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.
Kevin Harwell [Wed, 11 Dec 2013 20:11:50 +0000 (20:11 +0000)]
res_pjsip_messaging: send message to a default outbound endpoint
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.
Russell Bryant [Wed, 11 Dec 2013 19:18:31 +0000 (19:18 +0000)]
Reset peer outboundproxy on sip.conf reload
If you set a peer's outboundproxy and then removed it from the config,
this would not get picked up in a config reload. This patch fixes that
by resetting it in set_peer_defaults().
Matthew Jordan [Wed, 11 Dec 2013 13:05:12 +0000 (13:05 +0000)]
func_channel, chan_pjsip: Add CHANNEL read function support for chan_pjsip
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.
Matthew Jordan [Wed, 11 Dec 2013 12:29:49 +0000 (12:29 +0000)]
func_pjsip_endpoint: Add PJSIP_ENDPOINT function for querying endpoint details
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.
Richard Mudgett [Mon, 9 Dec 2013 19:23:49 +0000 (19:23 +0000)]
Reverting regex part of -r403545 at request of file.
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 '^'.
Richard Mudgett [Mon, 9 Dec 2013 18:49:28 +0000 (18:49 +0000)]
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 '^'.
* Fix off nominal memory leak in sorcery_astdb_retrieve_regex().
Joshua Colp [Mon, 9 Dec 2013 18:31:13 +0000 (18:31 +0000)]
endpoints: Keep a reference to channel ids when creating snapshot.
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.
Richard Mudgett [Mon, 9 Dec 2013 18:31:07 +0000 (18:31 +0000)]
sorcery: Eliminate shadowing a varaible that caused confusion.
* 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.
Joshua Colp [Mon, 9 Dec 2013 16:40:59 +0000 (16:40 +0000)]
res_pjsip_nat: Add NAT module to session dialogs.
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.
Matthew Jordan [Mon, 9 Dec 2013 03:19:39 +0000 (03:19 +0000)]
res_fax_spandsp: Always init T.38 session to avoid crashes during state change
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.
Matthew Jordan [Sun, 8 Dec 2013 05:54:53 +0000 (05:54 +0000)]
res_config_sqlite: Check for CDR unregistration failures
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.
Joshua Colp [Wed, 4 Dec 2013 20:53:32 +0000 (20:53 +0000)]
res_pjsip_t38: Don't pass T.38 control frames through to other hooks.
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".
Mark Michelson [Wed, 4 Dec 2013 18:40:23 +0000 (18:40 +0000)]
Initialize the hash value argument to pj_hash_get() to 0.
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.
David M. Lee [Tue, 3 Dec 2013 20:17:48 +0000 (20:17 +0000)]
ari: Fix deadlock problem with functions that use autoservice.
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.
Joshua Colp [Tue, 3 Dec 2013 17:59:41 +0000 (17:59 +0000)]
res_pjsip_session: Add support for PJMEDIA_SDP_NEG_ALLOW_MEDIA_CHANGE flag.
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.
Richard Mudgett [Tue, 3 Dec 2013 17:23:50 +0000 (17:23 +0000)]
sorcery, bucket: Change observer remove calls to take const callbacks struct.
* 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.
Joshua Colp [Tue, 3 Dec 2013 16:37:26 +0000 (16:37 +0000)]
media_index: Make media indexing tolerable of bad symlinks.
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.
Mark Michelson [Tue, 3 Dec 2013 16:33:49 +0000 (16:33 +0000)]
Add channel locking for channel snapshot creation.
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.
Alexandr Anikin [Mon, 2 Dec 2013 17:58:33 +0000 (17:58 +0000)]
Check and reject non-digits e164 values on peers and general sections in ooh323.conf
Regenerate e164 endpoint list on reload ooh323
(issue ASTERISK-22901)
Reported by: Cyril CONSTANTIN
Patches:
ASTERISK-22901.patch
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Merged revisions 403288 from http://svn.asterisk.org/svn/asterisk/branches/11
Joshua Colp [Sun, 1 Dec 2013 20:04:02 +0000 (20:04 +0000)]
res_pjsip_t38: Add the framehook to the channel only on first INVITE.
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.
Joshua Colp [Sun, 1 Dec 2013 19:56:38 +0000 (19:56 +0000)]
res_pjsip_transport_websocket: Fix security events and simplify implementation.
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.
Joshua Colp [Thu, 28 Nov 2013 02:12:04 +0000 (02:12 +0000)]
res_pjsip_sdp_rtp: Don't produce an invalid media stream with no formats.
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
Joshua Colp [Thu, 28 Nov 2013 01:56:02 +0000 (01:56 +0000)]
res_pjsip_header_funcs: Don't add headers to re-INVITEs.
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
Joshua Colp [Thu, 28 Nov 2013 00:53:43 +0000 (00:53 +0000)]
res_stasis_playback: Add 'number', 'digits', and 'characters' URI scheme implementations.
This change adds new URI scheme implementations for playing numbers, digits,
and characters. This is done as part of the normal playback mechanism and can
be used with queueing to create a combined sentence.
Joshua Colp [Thu, 28 Nov 2013 00:36:53 +0000 (00:36 +0000)]
res_pjsip_session: Add configurable behavior for redirects.
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
Joshua Colp [Wed, 27 Nov 2013 16:09:48 +0000 (16:09 +0000)]
res_pjsip: Fix crash when reloading certain configurations.
Certain options available that specify a SIP URI perform validation
on the provided URI using the PJSIP URI parser. This operation
requires that the thread executing it be registered with the PJLIB
library. During reloads this was done on a thread which was NOT
registered with it.
This fixes the problem by creating a task which reloads the
configuration on a PJSIP thread.
(closes issue ASTERISK-22923)
Reported by: Anthony Messina
David M. Lee [Wed, 27 Nov 2013 15:36:16 +0000 (15:36 +0000)]
ari:Add application/json parameter support
The patch allows ARI to parse request parameters from an incoming JSON
request body, instead of requiring the request to come in as query
parameters (which is just weird for POST and DELETE) or form
parameters (which is okay, but a bit asymmetric given that all of our
responses are JSON).
For any operation that does _not_ have a parameter defined of type
body (i.e. "paramType": "body" in the API declaration), if a request
provides a request body with a Content type of "application/json", the
provided JSON document is parsed and searched for parameters.
The expected fields in the provided JSON document should match the
query parameters defined for the operation. If the parameter has
'allowMultiple' set, then the field in the JSON document may
optionally be an array of values.
Joshua Colp [Wed, 27 Nov 2013 15:31:05 +0000 (15:31 +0000)]
res_pjsip: Update handling of some options to work with new option names.
Some options (such as call_group and pickup_group) share the same configuration
handler and decide what logic to use based on the name of the option. These
handlers were not updated to check for the new option names and were treating
the options as invalid.
This change simply updates the handlers with the proper names of the options.
(closes issue ASTERISK-22922)
Reported by: Anthony Messina
Kevin Harwell [Sat, 23 Nov 2013 17:38:29 +0000 (17:38 +0000)]
ARI: Implement device state API
Created a data model and implemented functionality for an ARI device state
resource. The following operations have been added that allow a user to
manipulate an ARI controlled device:
Create/Change the state of an ARI controlled device
PUT /deviceStates/{deviceName}&{deviceState}
Retrieve all ARI controlled devices
GET /deviceStates
Retrieve the current state of a device
GET /deviceStates/{deviceName}
Destroy a device-state controlled by ARI
DELETE /deviceStates/{deviceName}
The ARI controlled device must begin with 'Stasis:'. An example controlled
device name would be Stasis:Example. A 'DeviceStateChanged' event has also
been added so that an application can subscribe and receive device change
events. Any device state, ARI controlled or not, can be subscribed to.
While adding the event, the underlying subscription control mechanism was
refactored so that all current and future resource subscriptions would be
the same. Each event resource must now register itself in order to be able
to properly handle [un]subscribes.
(issue ASTERISK-22838)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3025/
Kevin Harwell [Sat, 23 Nov 2013 17:14:22 +0000 (17:14 +0000)]
res_pjsip: AMI commands and events.
Created the following AMI commands and corresponding events for res_pjsip:
PJSIPShowEndpoints - Provides a listing of all pjsip endpoints and a few
select attributes on each.
Events:
EndpointList - for each endpoint a few attributes.
EndpointlistComplete - after all endpoints have been listed.
PJSIPShowEndpoint - Provides a detail list of attributes for a specified
endpoint.
Events:
EndpointDetail - attributes on an endpoint.
AorDetail - raised for each AOR on an endpoint.
AuthDetail - raised for each associated inbound and outbound auth
TransportDetail - transport attributes.
IdentifyDetail - attributes for the identify object associated with
the endpoint.
EndpointDetailComplete - last event raised after all detail events.
PJSIPShowRegistrationsInbound - Provides a detail listing of all inbound
registrations.
Events:
InboundRegistrationDetail - inbound registration attributes for each
registration.
InboundRegistrationDetailComplete - raised after all detail records have
been listed.
PJSIPShowRegistrationsOutbound - Provides a detail listing of all outbound
registrations.
Events:
OutboundRegistrationDetail - outbound registration attributes for each
registration.
OutboundRegistrationDetailComplete - raised after all detail records
have been listed.
PJSIPShowSubscriptionsInbound - A detail listing of all inbound subscriptions
and their attributes.
Events:
SubscriptionDetail - on each subscription detailed attributes
SubscriptionDetailComplete - raised after all detail records have
been listed.
PJSIPShowSubscriptionsOutbound - A detail listing of all outboundbound
subscriptions and their attributes.
Events:
SubscriptionDetail - on each subscription detailed attributes
SubscriptionDetailComplete - raised after all detail records have
been listed.
(issue ASTERISK-22609)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2959/
Joshua Colp [Sat, 23 Nov 2013 12:51:31 +0000 (12:51 +0000)]
ari: Add events for playback and recording.
While there were events defined for playback and recording
these were not actually sent. This change implements the
to_json handlers which produces them.
(closes issue ASTERISK-22710)
Reported by: Jonathan Rose
Joshua Colp [Sat, 23 Nov 2013 12:38:04 +0000 (12:38 +0000)]
ari: Add Snoop operation for spying/whispering on channels.
The Snoop operation can be invoked on a channel to spy or
whisper on it. It returns a channel that any channel operations
can then be invoked on (such as record to do monitoring).
(closes issue ASTERISK-22780)
Reported by: Matt Jordan
Kinsey Moore [Fri, 22 Nov 2013 23:44:39 +0000 (23:44 +0000)]
Make sure unit tests compile
This fixes the unit tests that were broken by r403069 and several
functions requiring a new parameter for sanitization of JSON messages
generated from object snapshots.
Kevin Harwell [Fri, 22 Nov 2013 22:24:43 +0000 (22:24 +0000)]
res_pjsip: convert configuration settings names to snake case some more
Updated the alembic script for pjsip. Also, the dtls config parsing stuff was
expecting strings with no underscores, so removed the underscores from the
option name before passing it to the parser.
Kinsey Moore [Fri, 22 Nov 2013 20:01:26 +0000 (20:01 +0000)]
ARI: Don't leak implementation details
This change prevents channels used as implementation details from
leaking out to ARI. It does this by preventing creation of JSON blobs
of channel snapshots created from those channels and sanitizing JSON
blobs of bridge snapshots as they are created. This introduces a
framework for excluding information from output targeted at Stasis
applications on a consumer-by-consumer basis using channel sanitization
callbacks which could be extended to bridges or endpoints if necessary.
This prevents unhelpful error messages from being generated by
ast_json_pack.
This also corrects a bug where BridgeCreated events would not be
created.
(closes issue ASTERISK-22744)
Review: https://reviewboard.asterisk.org/r/2987/
Reported by: David M. Lee
Kevin Harwell [Fri, 22 Nov 2013 17:19:18 +0000 (17:19 +0000)]
res_pjsip: convert configuration settings names to snake case
Renamed, where appropriate, the configuration options for chan/res_pjsip to use
snake case (compound words separated by an underscore). For example, faxdetect
will become fax_detect, recordofffeature will become record_off_feature, etc...
Joshua Colp [Fri, 22 Nov 2013 17:11:46 +0000 (17:11 +0000)]
translate: Move freeing of frame to after it is used.
When translating from one format to another it is possible
to inform the translation function that the source frame should
be freed. This was previously done immediately but shortly
afterwards the frame that was freed was accessed and used again.
This change moves code around a bit so that the frame is now
freed after it has been completely used.
Matthew Jordan [Thu, 21 Nov 2013 17:52:11 +0000 (17:52 +0000)]
res_pjsip_sdp_rtp: Fix use of uninitialized value in PJSIP
In PJMEDIA, pjmedia_sdp_rtpmap_to_attr will attempt to use the string
rtpmap.param regardless of its length value. Simply setting the length to 0
does not prevent the garbage on the stack in rtpmap.param.ptr from being
formatted in a sprintf call. This patch initializes the string to NULL so that
at the very least, something is provided to the function that is predictable.
Matthew Jordan [Thu, 21 Nov 2013 17:49:28 +0000 (17:49 +0000)]
res_pjsip_mwi: Fix memory leak of MWI subscriptions container
This patch fixes a reference counting memory leak on the ao2_container
created as part of create_mwi_subscriptions. When we create the container
in this routine, the intent is to hand lifetime ownership over to the global
container unsolicited_mwi. When ao2_global_obj_replace_unref is called, the
reference count on mwi_subscriptions (the container) will be bumped by 1;
however, the function does not decrement the reference count on
mwi_subscriptions when this occurs. This will prevent the container from being
fully disposed of when Asterisk exits (or on any subsequent call to this
operation, such as during a reload).
David M. Lee [Thu, 21 Nov 2013 15:55:33 +0000 (15:55 +0000)]
ari: Add silence generator controls
This patch adds the ability to start a silence generator on a channel
via ARI. This generator will play silence on the channel (avoiding audio
timeouts on the peer) until it is stopped, or some other media operation
is started (like playing media, starting music on hold, etc.).
Joshua Colp [Tue, 19 Nov 2013 23:17:18 +0000 (23:17 +0000)]
res_pjsip_caller_id: Don't overwrite user portion of the From header when fromuser is set.
The fromuser option is used to explicitly set the user within the From header. The
res_pjsip_caller_id module did not take this setting into account when determining
if the From header could be modified or not.
(closes issue ASTERISK-22866)
Reported by: Anthony Messina
Joshua Colp [Sat, 16 Nov 2013 13:44:47 +0000 (13:44 +0000)]
res_pjsip: Add support for building against pjproject with SIP transaction group lock support.
SIP transaction group lock support has been backported into our pjproject. Since the code
now internally uses a group lock the code is now changed to unlock it if present. Note
that the act of finding the transaction is what actually returns it locked.
For further information about group locks check out the wiki page at:
http://trac.pjsip.org/repos/wiki/Group_Lock
Kinsey Moore [Fri, 15 Nov 2013 14:35:58 +0000 (14:35 +0000)]
CEL: Fix crash when using CELGenUserEvent
This fixes a crash when CELGenUserEvent is called from the dialplan
while CEL is disabled. Currently, CEL does not create its topics and
forwards if it is not enabled and external entities may depend on
these topics blindly since they should always be available. This patch
breaks up route creation and topic/forward creation such that the CEL
topics and forwards will always exist while the router and its
associated routes will be torn down and recreated as necessary.
(closes issue ASTERISK-22799)
Review: https://reviewboard.asterisk.org/r/3010/
Reported by: Matt Jordan
Joshua Colp [Wed, 13 Nov 2013 23:09:38 +0000 (23:09 +0000)]
res_ari_channels: Add the ability to stop locally generated ringing on a channel.
Using the 'ring' operation it is possible to start locally generated ringback if
the channel is answered. This change adds the ability to stop it by using DELETE.
Kevin Harwell [Tue, 12 Nov 2013 23:16:58 +0000 (23:16 +0000)]
ari endpoints: GET /ari/endpoints/{invalid-tech} should return a 404
Was returning a 404 on a valid technology with an empty list of endpoints.
Now checking against the channel tech to make sure the tech itself is valid
and not just an empty list of endpoints.
Kevin Harwell [Tue, 12 Nov 2013 22:15:41 +0000 (22:15 +0000)]
ari endpoints: GET /ari/endpoints/{invalid-tech} should return a 404
Implementation listing endpoints by technology returned an empty array if no
matching endpoints were found. Fixed so a "404 Not Found" will be returned
instead.
(closes issue ASTERISK-22803)
Reported by: David M. Lee
Kevin Harwell [Tue, 12 Nov 2013 16:45:42 +0000 (16:45 +0000)]
pjsip_messaging, pjsip_header_funcs: Crashes due to NULL pointer dereferences
Both res_pjsip_messaging and res_pjsip_header_funcs were causing asterisk to
crash because they were trying to dereference a NULL pointer.
In the case of res_pjsip_messaging it was attempting to "print" a contact
header that did not exist. In fact contact headers should not be part of
a SIP MESSAGE, so the offending code was simply removed.
In the case of res_pjsip_header_funcs a null private channel tech was being
passed to the function and then later dereferenced. Added null checks (and
error logging) to the read/write function handlers to guard against crashing.
(closes issue ASTERISK-22821)
Reported by: Anthony Messina
Kinsey Moore [Tue, 12 Nov 2013 16:33:24 +0000 (16:33 +0000)]
CELGenUserEvent: Fix error message from ast_json_pack
This prevents NULL from being passed into an ast_json_pack call when no
extra information is passed to the application which prevents an error
message about NULL arguments from being generated.
Kinsey Moore [Tue, 12 Nov 2013 15:02:18 +0000 (15:02 +0000)]
chan_dahdi: Fix crash during caller ID read
Asterisk will sometimes core dump during caller id read on analog
channels due to a negative return value from the read() in
my_get_callerid that slips through as a negative length argument to
callerid_feed() if the errno returned by DAHDI is ELAST. This change
ensures that the negative return is treated properly even when it is
ELAST.
(closes issue ASTERISK-22746)
Reported by: Michael Walton
Patches:
chan_dahdi_cid_crash_fix.r401410.patch uploaded by Michael Walton (License 6502)
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Merged revisions 402708 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 402709 from http://svn.asterisk.org/svn/asterisk/branches/11
Kinsey Moore [Mon, 11 Nov 2013 15:36:23 +0000 (15:36 +0000)]
app_queue: Honor penalty limits of 0
In the current app_queue code from 1.8 up to trunk the upper and lower
penalties can be set to 0 but the value is interpreted to be disabled
instead of actually setting limits. This is especially evident if min
and max limits are set to 0 and members with penalties of 0 and 1 are
in the queue since the member with penalty 1 will still receive calls.
This patch adjusts the special disabled value to be INT_MAX instead of
0.
(closes issue ASTERISK-20862)
Review: https://reviewboard.asterisk.org/r/2995/
Reported by: Schmooze Com
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Merged revisions 402645 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 402646 from http://svn.asterisk.org/svn/asterisk/branches/11
chan_sip: keep same local (from) tag for outgoing register requests
For outbound register requests the tag on the From line was
updated every 20 seconds prior to a successful registration
and also once for each registration renewal. That behavior
can possibly cause the registration to be denied because of
the different tag, and is not aligned with the intention of
RFC 3261 8.1.3.5 "... request constitutes a new transaction
and SHOULD have the same value of the Call-ID, To, and From
of the previous request...". This updates chan_sip to have
a field to keep the local tag in the registration structure
and use that tag for registration requests where the callid
is also unchanged.
David M. Lee [Fri, 8 Nov 2013 17:28:40 +0000 (17:28 +0000)]
ari: Add application/x-www-form-urlencoded parameter support
ARI POST calls only accept parameters via the URL's query string.
While this works, it's atypical for HTTP API's in general, and
specifically frowned upon with RESTful API's.
This patch adds parsing for application/x-www-form-urlencoded request
bodies if they are sent in with the request. Any variables parsed this
way are prepended to the variable list supplied by the query string.
Jonathan Rose [Thu, 7 Nov 2013 23:16:30 +0000 (23:16 +0000)]
PJSIP: Improve error handling in digest authenticator
Previously, regardless of whether failure to authenticate was due to
lacking any authentication or actually failing authentication, the
Digest Authenticator would simply return that a challenge was still
needed. It will continue to do that when no authentication information
is in the received SIP digest, but when authentication information
is present and does not pass authentication, that will be treated as
an authentication error. This is to ensure that PJSIP will issue
security events indicated failed auths.
Etc. We put the object name in the nickname (since we were generating C
code), but it makes OO generators redundant.
This patch makes the nicknames more OO friendly. This resulted in a lot
of name changing within the res_ari_*.so modules, but not much else.
There were a couple of other fixed I made in the process.
* When reversible operations (POST /hold, POST /unhold) were made more
RESTful (POST /hold, DELETE /unhold), the path for the second operation
was left in the API declaration. This worked, but really the two
operations should have been on the same API.
* The POST /unmute operation had still not been REST-ified.
Kevin Harwell [Wed, 6 Nov 2013 21:57:04 +0000 (21:57 +0000)]
app_queue: crash if first agent is "busy"
If the first agent/member (via CLI "queue show") in a queue is "busy" (dnd,
circuit busy, etc...) and no agents answered then app_queue would crash.
This occurred because while the calling of agent(s) remained valid the channel
on "busy" agent would be set to NULL and then later dereferenced upon a second
"rna" function call. The original intention of the code is to have only valid
"call attempt" objects (channels != NULL) checked while attempting to call
agent(s). It does this by building a "call_next" list of valid "call attempt"
objects. In the case of the "busy" agent subsequent builds of the valid "call
attempt" list would sometimes include (the case mentioned above) an invalid
"call attempt" object.
The fix was to make sure the "call attempt" list was appropriately built on
every iteration. A NULL sanity check was also added at the original offending
spot of the crash just in case another one slipped by somehow.
(closes issue ASTERISK-22644)
Reported by: Marco Signorini
Review: https://reviewboard.asterisk.org/r/2983/
Matthew Jordan [Tue, 5 Nov 2013 21:16:28 +0000 (21:16 +0000)]
chan_sip: Use AST_AF* defined constant when calling ast_get_ip
While the structure passed to ast_get_ip should be set memset to 0, thus
initializing the ss_family member to 0, explicitly setting it to AST_AF_UNSPEC
is more portable.
Matthew Jordan [Tue, 5 Nov 2013 21:10:20 +0000 (21:10 +0000)]
chan_iax2: Fix incorrect usage of ast_get_ip involving uninitialized struct
This started off as a fix for the failing IAX2 acl_call test in the Asterisk
Test Suite. When inspecting why that test was failing, it became clear that all
attempts to bind to any local loopback address was failing:
[Nov 2 15:56:28] VERBOSE[15787] chan_iax2.c: == Binding IAX2 to address
127.0.0.1:4569
[Nov 2 15:56:28] DEBUG[15787] netsock2.c: Splitting '127.0.0.1' into...
[Nov 2 15:56:28] DEBUG[15787] netsock2.c: ...host '127.0.0.1' and port ''.
[Nov 2 15:56:28] ERROR[15787] netsock2.c: getaddrinfo("127.0.0.1", "(null)",
...): ai_family not supported
[Nov 2 15:56:28] WARNING[15787] acl.c: Unable to lookup '127.0.0.1'
While there's conceivably other ways for getaddrino to return EAI_FAMILY, the
most common way is if AF_INET, AF_INET6, or AF_UNSPEC is not provided as the
desired family. The culprit was the call to ast_get_ip, defined in acl.h. This
function uses the family from the passed in addr object (which it will also
populate when it returns!) when it eventually calls getaddrinfo.
This patch fixes the use of ast_get_ip that were not specifying the family in
chan_iax2. This prevents uninitialized use of the structure, so that the
addresses resolve correctly.
Matthew Jordan [Tue, 5 Nov 2013 21:06:03 +0000 (21:06 +0000)]
netsock2: Define AST_AF_* enum constants to their AF_* equivalents
This patch explicitly defines AST_AF_* enum constants to their sys/socket.h
defined equivalents. It is certainly unclear why these constants actually have
to exist, given that netsock2.h includes sys/socket.h; however, since the code
base is already liberally sprinkled with the usage of AST_AF_* (as well as with
direct calls to AF_*), this will at least keep the semantics consistent between
their usage across systems.
Matthew Jordan [Tue, 5 Nov 2013 20:53:27 +0000 (20:53 +0000)]
stasis_channels: Don't give preference to ANI info in channel snapshots
When publishing channel snapshots, we currently compute the caller ID name and
number by giving preference first to ani.{name|number}, then to
id.{name|number}. However, when a channel driver (such as chan_sip) updates the
caller ID, it typically only updates the caller ID stored in id.{name|number}.
This means that we are currently giving preference to stale information.
When looking at the rest of the code base, the only other place where we appear
to use this same logic is in app_amd. Everywhere else, we treat the party
information in ani as being separate to the party information in id.
This patch publishes only the caller ID name and number in the snapshot field
for caller_name and caller_num. Note that the information in ANI is still
available in caller_ani.
Kevin Harwell [Mon, 4 Nov 2013 20:56:16 +0000 (20:56 +0000)]
chan_sip: notify dialog info ignores presentation indicator in callerid
The presentation indicator in a callerid (e.g. set by dialplan function
Set(CALLERID(name-pres)= ...)) is not checked when SIP Dialog Info Notifies
are generated during extension monitoring. Added a check to make sure the
name and/or number presentations on the callee (remote identity) are set to
allow. If they are restricted then "anonymous" is used instead.
(closes issue AST-1175)
Reported by: Thomas Arimont
Review: https://reviewboard.asterisk.org/r/2976/
........
Merged revisions 402450 from http://svn.asterisk.org/svn/asterisk/branches/11
Richard Mudgett [Sat, 2 Nov 2013 04:05:24 +0000 (04:05 +0000)]
vector: Update API to be more flexible.
Made the vector macro API be more like linked lists.
1) Added a name parameter to ast_vector() to name the vector struct.
2) Made the API take a pointer to the vector struct instead of the struct
itself.
3) Added an element cleanup macro/function parameter when removing an
element from the vector for ast_vector_remove_cmp_unordered() and
ast_vector_remove_elem_unordered().
4) Added ast_vector_get_addr() in case the vector element is not a simple
pointer.
* Converted an inline vector usage in stasis_message_router to use the
vector API. It needed the API improvements so it could be converted.
* Fixed topic reference leak in router_dtor() when the
stasis_message_router is destroyed.
* Fixed deadlock potential in stasis_forward_all() and
stasis_forward_cancel(). Locking two topics at the same time requires
deadlock avoidance.
* Made internal_stasis_subscribe() tolerant of a NULL topic.
* Made stasis_message_router_add(),
stasis_message_router_add_cache_update(), stasis_message_router_remove(),
and stasis_message_router_remove_cache_update() tolerant of a NULL
message_type.
* Promoted a LOG_DEBUG message to LOG_ERROR as intended in
dispatch_message().
Richard Mudgett [Sat, 2 Nov 2013 03:21:17 +0000 (03:21 +0000)]
confbridge: Separate user muting from system muting overrides.
The system overrides the user muting requests when MOH is playing or a
waitmarked user is waiting for a marked user to join. System muting
overrides interfere with what the user may wish the muting to be when the
system override ends.
* User muting requests are now independent of the system muting overrides.
The effective muting is now the logical or of the user request and system
override.
* Added a Muted flag to the CLI "confbridge list <conference>" command.
* Added a Muted header to the AMI ConfbridgeList action ConfbridgeList
event.
Richard Mudgett [Sat, 2 Nov 2013 01:11:16 +0000 (01:11 +0000)]
config: Allow ConfBridge DTMF menus to have '#' as the first digit.
ConfBridge allows custom DTMF menus to be created in the confbridge.conf
file by assigning a DTMF key sequence to a sequence of actions as follows:
DTMF-sequence = action,action...
Unfortunately, the normal config file processing code interprets an
initial '#' character as starting a directive such as #include.
* Add the ability to escape the first non-blank character in a config line
so the '#' character can be used without triggering the directive
processing code.
(closes issue AFS-2)
(closes issue ASTERISK-22478)
Reported by: Nicolas Tanski
Patches:
jira_asterisk_22478_v11.patch (license #5621) patch uploaded by rmudgett (modified)
BridgeDestroy kicks an entire bridge, where BridgeKick kicks just one
channel off the bridge. When kicking a channel, specifying the bridge
also (optional) insures it is not removed from the wrong bridge. The
BridgeTechnology events allow viewing and changing suspension status,
which affects only subsequent not active bridging.
(closes ASTERISK-22356)
Reported by: Richard Mudgett
Review: https://reviewboard.asterisk.org/r/2973/
Joshua Colp [Fri, 1 Nov 2013 14:37:00 +0000 (14:37 +0000)]
res_ari_channels: Add ring operation, dtmf operation, hangup reasons, and tweak early media.
The ring operation sends ringing to the specified channel it is invoked on.
The dtmf operation can be used to send DTMF digits to the specified channel
of a specific length with a wait time in between. Finally hangup reasons
allow you to specify why a channel is being hung up (busy, congestion).
Early media behavior has also been tweaked slightly. When playing media to a channel
it will no longer automatically answer. If it has not been answered a progress indication
is sent instead.
(closes issue ASTERISK-22701)
Reported by: Matt Jordan
Kinsey Moore [Fri, 1 Nov 2013 12:38:22 +0000 (12:38 +0000)]
chan_sip: Fix RTCP port for SRFLX ICE candidates
This corrects one-way audio between Asterisk and Chrome/jssip as a
result of Asterisk inserting the incorrect RTCP port into RTCP SRFLX
ICE candidates. This also exposes an ICE component enumeration to
extract further details from candidates.
(closes issue ASTERISK-21383)
Reported by: Shaun Clark
Review: https://reviewboard.asterisk.org/r/2967/
........
Merged revisions 402345 from http://svn.asterisk.org/svn/asterisk/branches/11
projects / thirdparty / asterisk.git / log
