Michael L. Young [Fri, 21 Feb 2014 00:49:19 +0000 (00:49 +0000)]
app_chanspy: Documentation Update To Clarify "x" Option
When using the "x" option (specify a DTMF digit to exit the application), it is
not obvious in the documentation that this only works when spying on a channel.
If a channel being used to spy on other channels is waiting to connect to a
channel or is no longer attached to a channel, the DTMF is ignored.
As noted on the issue tracker, since there are workarounds available and this is
a rarely used option we are opting for a documentation change here.
(closes issue ASTERISK-22661)
Reported by: Chris Hillman
Patches:
asterisk-22661-doc-clarify-chan_spy.diff
uploaded by Michael L. Young (license 5026)
George Joseph [Thu, 20 Feb 2014 21:09:47 +0000 (21:09 +0000)]
pjsip_cli: Add pjsip commands 'show registrations' and 'show contacts'.
Added 'show registrations' and 'show contacts' to pjsip cli to make things
a little more consistent. The output is exactly the same as the list command.
Just needed to add entries to their respective ast_cli_entry structures.
George Joseph [Thu, 20 Feb 2014 20:42:36 +0000 (20:42 +0000)]
sorcery: Create sorcery instance registry.
In order to retrieve an arbitrary sorcery instance from a dialplan function
(or any place else) there needs to be a registry of sorcery instances.
ast_sorcery_init now creates a hashtab as a registry.
ast_sorcery_open now checks the hashtab for an existing sorcery instance
matching the caller's module name. If it finds one, it bumps the
refcount and returns it. If not, it creates a new sorcery instance,
adds it to the hashtab, then returns it.
ast_sorcery_retrieve_by_module_name is a new function that does a hashtab
lookup by module name. It can be called by the future dialplan function.
res_pjsip/config_system needed a small change to share the main res_pjsip
sorcery instance.
tests/test_sorcery was updated to include a test for the registry.
Matthew Jordan [Thu, 20 Feb 2014 19:02:13 +0000 (19:02 +0000)]
res_pjsip: Update documentation for 'use_avpf' option
When 'use_avpf' is set to True, inbound offers must use the AVPF/SAVPF RTP
profile. However, when 'use_avpf' is set to False, Asterisk will accept
both AVP/SAVP or AVPF/SAVPF RTP profiles in inbound offers. The documentation
previously implied that Asterisk would reject AVPF/SAVPF if 'use_avpf' was
set to False and a UA offered said profile in an INVITE request.
Richard Mudgett [Wed, 19 Feb 2014 19:07:42 +0000 (19:07 +0000)]
config: Add file size and nanosecond resolution fields to the cached modified config file information.
Repeatedly modifying config files and reloading too fast sometimes fails
to reload the configuration because the cached modification timestamp has
one second resolution.
* Added file size and nanosecond resolution fields to the cached config
file modification timestamp information. Now if the file size changes or
the file system supports nanosecond resolution the modified file has a
better chance of being detected for reload.
* Added a missing unlock in an off-nominal code path.
Richard Mudgett [Wed, 19 Feb 2014 18:26:17 +0000 (18:26 +0000)]
res_sorcery_astdb.c: Fix regex handling and keep simple prefix matching performance.
The sorcery astDB wizzard does not handle regex correctly if the pattern
begins with an anchor character.
This patch attempts to convert the anchored regex pattern to a prefix
pattern supported by astDB for performance reasons. If it is not able to
convert the pattern it falls back to getting all astDB members of the
family and doing a normal regex pattern matching on the retrieved records.
Alexandr Anikin [Wed, 19 Feb 2014 10:20:02 +0000 (10:20 +0000)]
Allow different socket and signalling ip on h.323 connection if gk mode is active
Reported by: Gabriele Odone
Patches:
ASTERISK-22738-1.patch
Tested by: Gabriele Odone
(closes issue ASTERISK-22738)
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Merged revisions 408312 from http://svn.asterisk.org/svn/asterisk/branches/11
Mark Michelson [Mon, 17 Feb 2014 15:21:59 +0000 (15:21 +0000)]
Store SIP User-Agent information in contacts.
When an endpoint sends a REGISTER request to Asterisk, we now will
associate the User-Agent header with all contacts that were bound in
that REGISTER request.
Matthew Jordan [Sun, 16 Feb 2014 03:23:14 +0000 (03:23 +0000)]
pbx: Handle a completely empty dialplan during a context merge
It is highly unlikely, but - at least in Asterisk 12 - theoretically possible
to load Asterisk with no dialplan whatsoever. If that occurs, and some other
module (that is not a pbx module) attempts to merge its contexts into the
dialplan, the existing merge routine will crash. This is because it is not
insane, and rightly believes that you provided some sort of dialplan,
somewhere.
This patch will gracefully merge the contexts in such a case. Note that this
is highly unlikely to occur in 1.8/11, as features will most likely provide
some dialplan via parking. However, in Asterisk 12, parking is now provided
by res_parking, and hence may create its dialplan later.
(closes issue ASTERISK-23297)
Reported by: CJ Oster
Matthew Jordan [Sun, 16 Feb 2014 01:49:05 +0000 (01:49 +0000)]
buildsystem: Unbreak the build (infloop) on Asterisk 11+
Apparently r408084 ( https://reviewboard.asterisk.org/r/3212/ ) broke the
build. This patch fixes it by ignoring the .lastclean dependencies if the
MENUSELECT_EMBED variable is not defined.
patches:
tmp.diff uploaded by wdoekes (License 5674)
URI's are supposed to be case sensitive and all
lower case. In practice some portions of URI's
in ARI are case insensitive and others are not,
such as TECH, which in one instance would match
a lower case name and in another would not. In
this patch, the ast_endpoint_lastest_snapshot()
function is modified to change the TECH portion
to full upper case before lookup. This resolves
the discrepancy noted by the reporter. However
I chose to avoid forcing the /ari prefix of the
URI's to be lower case for now. Except for the
two cases here, all URI's should be lower case,
unless they are part of a resource name or id.
format.c: correct possible null pointer dereference
In ast_format_sdp_parse and ast_format_sdp_generate
the check checks for a valid interface and function
were potentially confusing, and hid an error in the
test of the presence of the function that is called
later. This patch clears up and corrects the test.
Matthew Jordan [Fri, 14 Feb 2014 12:39:11 +0000 (12:39 +0000)]
chnan_sip: Set SIP_DEFER_BYE_ON_TRANSFER prior to calling bridge blind transfer
This patch moves setting SIP_DEFER_BY_ON_TRANSFER prior to calling
ast_bridge_transfer_blind. This prevents a BYE from being sent prior to the
NOTIFY request that informs the transferor if the transfer succeeded or failed.
This patch also clears said flag from the off nominal NOTIFY paths in the
local_attended_transfer code, as once we've sent the NOTIFY request it is safe
to send by the BYE request.
This was caught by the blind-transfer-accountcode test in the Asterisk Test
Suite.
(closes issue ASTERISK-23290)
Reported by: Matt Jordan
Mark Michelson [Thu, 13 Feb 2014 18:50:30 +0000 (18:50 +0000)]
Remove all PJSIP MWI-specific use from our MWI code.
PJSIP has built-in MWI code that could be useful to some
degree, but our utilization of the API actually made our
code a bit more cluttered since we had to have special
cases peppered throughout.
With this change, we move to using the pjsip_evsub API
instead, which streamlines the code by removing special
cases.
Walter Doekes [Wed, 12 Feb 2014 08:18:37 +0000 (08:18 +0000)]
realtime: Fix ast_update2_realtime() on raspberry pi.
The old code depended on undefined va_arg behaviour: calling a function
twice with the same va_list parameter and expecting it to continue where
it left off. The changed code behaves like the manpage says it should.
Also added a bunch of early returns to trap errors (e.g. OOM) instead of
crashing.
The problem was found by Julian Lyndon-Smith. The deviant behaviour on
the raspberry PI also uncovered another bug (fixed in r407875) in the
res_config_pgsql.so driver.
Matthew Jordan [Tue, 11 Feb 2014 03:16:18 +0000 (03:16 +0000)]
ari/resource_channels: Add channel variables earlier in the creation process
This patch tweaks the behaviour of POST /channels with channel variables such
that the variables are passed into the pbx.c routines that perform the
origination. This allows the variables to be assigned to the newly created
channels immediately upon their construction, as opposed to be assigned after
the originate has completed.
The upshot of this is that the variables are available on the channels if
they execute in the dialplan, as opposed to only being available once the
channels are answered.
Walter Doekes [Mon, 10 Feb 2014 16:43:29 +0000 (16:43 +0000)]
res_config_pgsql: Fix ast_update2_realtime calls.
Fix so multiple updates from a single call works (add missing ',').
Remove bogus ast_free's that weren't supposed to be there.
Moved a few spaces for readability.
Kinsey Moore [Mon, 10 Feb 2014 15:54:45 +0000 (15:54 +0000)]
ConfBridge: Correct prompt playback target
Currently, when the first marked user enters the conference that
contains waitmarked users, a prompt is played indicating that the user
is being placed into the conference. Unfortunately, this prompt is
played to the marked user and not the waitmarked users which is not
very helpful.
This patch changes that behavior to play a prompt stating
"The conference will now begin" to the entire conference after adding
and unmuting the waitmarked users since the design of confbridge is not
conducive to playing a prompt to a subset of users in a conference in
an asynchronous manner.
(closes issue PQ-1396)
Review: https://reviewboard.asterisk.org/r/3155/
Reported by: Steve Pitts
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Matthew Jordan [Fri, 7 Feb 2014 20:09:01 +0000 (20:09 +0000)]
security_events: Fix assertion failure in dev-mode on optional IE parsing
When formatting an optional IE, the value is, of course, optional. As such, it
is entirely appropriate for ast_json_object_get to return NULL. If that occurs,
we now simply skip the IE that was requested, as it was not provided by the
entity that raised the event.
Thanks to George Joseph (gtjoseph) for catching this and reporting it in
#asterisk-dev
Matthew Jordan [Fri, 7 Feb 2014 19:36:15 +0000 (19:36 +0000)]
funcs/func_cdr: Handle empty time values when extracting parsed values
When extracting timestamps that are parsed, time stamp values that are not set
(time values of 0.000000) should not actually result in a parsed string. The
value should be skipped, and the result of the CDR function should be an
empty string.
Prior to this patch, the result was fed to the time formatting, which would
result in an output of a date/time in 1969.
Richard Mudgett [Fri, 7 Feb 2014 18:18:26 +0000 (18:18 +0000)]
chan_iax2: Block unnecessary control frames to/from the wire.
Establishing an IAX2 call between Asterisk v1.4 and v1.8 (or later)
results in an unexpected call disconnect. The problem happens because
newer values in the enum ast_control_frame_type are not consistent between
the branch versions of Asterisk.
For example:
1) v1.4 calls v1.8 (or later) using IAX2
2) v1.8 answers and sends a connected line update control frame. (on v1.8
AST_CONTROL_CONNECTED_LINE = 22)
3) v1.4 receives the control frame as an end-of-q (on v1.4
AST_CONTROL_END_OF_Q = 22)
4) v1.4 disconnects the call once the receive queue becomes empty.
Several things are done by this patch to fix the problem and attempt to
prevent it from happening again in the future:
* Added a warning at the definition of enum ast_control_frame_type about
how to add new control frame values.
* Made block sending and receiving control frames that have no reason to
go over the wire.
* Extended the connectedline iax.conf parameter to also include the
redirecting information updates.
* Updated the connectedline iax.conf parameter documentation to include a
notice that the parameter must be "no" when the peer is an Asterisk v1.4
instance.
Matthew Jordan [Fri, 7 Feb 2014 16:46:14 +0000 (16:46 +0000)]
security_events: Fix error caused by DTD validation error
The appdocsxml.dtd specifies that a "required" attribute in a parameter may
have a value of yes, no, true, or false. On some systems, specifying "False"
instead of "false" would cause a validation error. This patch fixes the casing
to explicitly match the DTD.
Tzafrir Cohen [Fri, 7 Feb 2014 13:13:39 +0000 (13:13 +0000)]
indications.conf: add stutter tone; end properly
* If the "stutter" (voicemail indication) tone is indeed a stutter tone,
and it ends with a constant tone, make sure that it is the dial tone.
This was done for India (in), Mexico (mx) and the Philippines (ph).
* If no "stutter" tone exists for a country, provide one. This was done for
Spain (es), Malaysia (my) and Venezuela (ve).
Kevin Harwell [Thu, 6 Feb 2014 17:54:42 +0000 (17:54 +0000)]
pjsip realtime: already created enum failure for postgresql
If an enum had been previously created the alembic script would attempt to
re-create it and an error would be generated while running migrations for a
postgresql server. The work around for this is to use the ENUM object type
for postgres as opposed to the generic enum type used by sqlalchemy. Using
this type in the script seems to work properly for both postgres and mysql.
Richard Mudgett [Thu, 6 Feb 2014 17:06:47 +0000 (17:06 +0000)]
res_pjsip: Updates and adds more PJSIP CLI commands.
* Adds identify, transport, and registration support to the PJSIP CLI.
* Creates three additional callbacks, one for an iterator, one for a
comparator, and one for a container. This eliminates the link dependency
from higher level modules to lower level ones.
* Eliminates duplicate sorting in PJSIP CLI commands.
* Cleans up PJSIP CLI output formatting.
* Pushes CLI command registration down to the implementing source file.
* Adds several ast_sip_destroy_sorcery functions to complement existing
ast_sip_sorcery_initialize functions. The destroy functions unregister
PJSIP CLI commands and PJSIP CLI formatters.
Mark Michelson [Thu, 6 Feb 2014 16:53:24 +0000 (16:53 +0000)]
Fix alembic script to work properly in offline mode.
When run in offline mode, this would attempt to check the database for
the presence of a type it was going to try to create. I now check the
context to see if we're running in offline mode and change a parameter
accordingly.
Rusty Newton [Wed, 5 Feb 2014 23:03:01 +0000 (23:03 +0000)]
formats/format_wav: enhancing log message "Not a wav file" to be clear on what is supported
Modifying the log message to be more specific as to what is supported. Specifically it seems format_wav supports only PCM encoded versions with a lower-case '.wav' extension.
(closes issues ASTERISK-22310)
Reported by: Jim Credland
Review: https://reviewboard.asterisk.org/r/3188/
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Merged revisions 407512 from http://svn.asterisk.org/svn/asterisk/branches/11
Jonathan Rose [Wed, 5 Feb 2014 20:55:43 +0000 (20:55 +0000)]
CHANGES: Improved description of Name/Creator changes to bridge ARI, adds AMI
The changes log was written with language that was a little too internal
Asterisk specific, so it's been changed to be more in the frame of reference
of an ARI user. Also, previously the AMI event changes were omitted from the
change log as well as the ability to include a bridge name in the ARI post
bridges command.
Kinsey Moore [Wed, 5 Feb 2014 20:43:07 +0000 (20:43 +0000)]
Logger: Fix handling of absolute paths
This fixes path handling for log files so that an extra / is not
appended to the file path when the path is absolute (begins with /).
This would previously result in different but functionally equivalent
paths in the output of 'logger show channels'.
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Kevin Harwell [Wed, 5 Feb 2014 19:41:48 +0000 (19:41 +0000)]
res_pjsip: When no global type the debug option defaults to "yes"
If the global section was not specified in pjsip.conf then the configuration
object does not exist in sorcery so when retrieving "debug" option it would
return NULL. Then the NULL result was passed to ast_false utils function
which would return false because it wasn't set to some representation of
false, thus enabling sip debug logging. Made it so if the global config object
does not exist then it will return a default of "no" for sip debugging.
Matthew Jordan [Wed, 5 Feb 2014 00:08:50 +0000 (00:08 +0000)]
api.wiki.mustache: Update wiki template to support body parameters
This patch updates the api.wiki.mustache template and the swagger_model python
script to understand if an operation has a body parameter. If an operation
does have a body parameter, it will now be displayed in the corresponding
wiki entry.
Joshua Colp [Tue, 4 Feb 2014 02:21:31 +0000 (02:21 +0000)]
res_clialiases: Fix crash when reloading and re-aliasing an alias that is in use.
The code assumed that unregistering the alias would always succeed while in
practice this is not actually true. A common case is the "reload" command itself.
If the cli_aliases.conf configuration file was changed and reload executed the
command would fail to unregister and ultimately point to freed memory.
The reload process now checks whether unregistering succeeded or not and if not
the old CLI alias is retained.
(closes issue ASTERISK-19773)
Reported by: Joel Vandal
Damien Wedhorn [Tue, 4 Feb 2014 02:04:18 +0000 (02:04 +0000)]
Skinny - Fix deadlock when pickup of no call.
Locking issues in skinny when picking up a call that doesn't exist. Cleaned
up sub locking by fully removing and using the chan lock instead. Also
changed ast_call_pickup to check whether chan was masq'd.
Matthew Jordan [Mon, 3 Feb 2014 01:14:27 +0000 (01:14 +0000)]
cdrs: Check for applications to lock onto during dial begin handling
This patch brings CDR processing further in line with r407085. During some dial
operations, the application would not be locked to the Dial application and
would instead continue to show the previously known application. In particular,
this would occur when a Parked call would time out. This was due to a previous
snapshot already locking the application to Park - processing this in a Dial
Begin allows the Dial application to reassert its rightful place.
(CDRs. Ugh.)
But hooray for the Parked Call tests for catching this in the Asterisk Test
Suite.
Joshua Colp [Sat, 1 Feb 2014 16:23:56 +0000 (16:23 +0000)]
res_stasis: Enable transfers and provide events when they occur.
This change enables transfers within ARI created bridges and adds events
for when they occur. Unlike other events these will be received if *any*
subscribed object is involved in the transfer.
(closes issue ASTERISK-22984)
Reported by: David M. Lee
Corey Farrell [Sat, 1 Feb 2014 00:24:55 +0000 (00:24 +0000)]
app_stack: protect against missing parameters to STACK_PEEK and LOCAL_PEEK
STACK_PEEK requires 2 parameters and LOCAL_PEEK requires 1 parameter. This
protects against situations where those parameters are blank or missing by
logging an error and returning.
(closes issue ASTERISK-23220)
Reported by: James Sharp
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Matthew Jordan [Fri, 31 Jan 2014 23:40:24 +0000 (23:40 +0000)]
CDRs: fix a variety of dial status problems, h/hangup handler creating CDRs
This patch fixes a number of small-ish problems that were noticed when
witnessing the records that the FreePBX dialplan produces:
(1) Mid-call events (as well as privacy options) have the ability to change the
overall state of the Dial operation after the called party answers. This
means that publishing the DialEnd event when the called party is premature;
we have to wait for the execution of these subroutines to complete before
we can signal the overall status of the DialEnd. This patch moves that
publication and adds handlers for the mid-call events.
(2) The AST_FLAG_OUTGOING channel flag is cleared if an after bridge goto
datastore is detected. This flag was preventing CDRs from being recorded
for all outbound channels that had a 'continue' option enabled on them by
the Dial application.
(3) The CDR engine now locks the 'Dial' application as being the CDR
application if it detects that the current CDR has entered that app. This
is similar to the logic that is done for Parking. In general, if we entered
into Dial, then we want that CDR to record the application as such - this
prevents pre-dial handlers, mid-call handlers, and other shenaniganry
from changing the application value.
(4) The CDR engine now checks for the AST_SOFTHANGUP_HANGUP_EXEC in more places
to determine if the channel is in hangup logic or dead. In either case, we
don't want to record changes in the channel.
(5) The default option for "endbeforehexten" has been changed to "yes". In
general, you don't want to see CDRs in the 'h' exten or in hangup logic.
Since the semantics of that option changed in 12, it made sense to update
the default value as well.
(6) Finally, because we now have the ability to synchronize on the messages
published to the CDR topic, on shutdown the CDR engine will now synchronize
to the messages currently in flight. This helps to ensure that all
in-flight CDRs are written before shutting down.
(closes issue ASTERISK-23164)
Reported by: Matt Jordan
Matthew Jordan [Fri, 31 Jan 2014 23:34:00 +0000 (23:34 +0000)]
app_dial: Allow macro/gosub pre-bridge execution to occur on priorities
The parsing for the destination of the macro/gosub uses the '^' character to
separate out context, extension, and priority. However, the logic for the
macro/gosub execution was written such that it would only do the actual
macro/gosub jump if a '^' character existed. This doesn't apply when the
macro/gosub jump occurs in a priority/priority label. This patch changes
the logic so that the parsing still occurs, but the jump will occur even
for priorities/priority labels.
Kevin Harwell [Fri, 31 Jan 2014 23:14:19 +0000 (23:14 +0000)]
res_pjsip: Config option to enable PJSIP logger at load time.
Added a "debug" configuration option for res_pjsip that when set to "yes"
enables SIP messages to be logged. It is specified under the "system" type.
Also added an alembic script to add the option to realtime.
(closes issue ASTERISK-23038)
Reported by: Rusty Newton
Review: https://reviewboard.asterisk.org/r/3148/
Kevin Harwell [Fri, 31 Jan 2014 23:12:20 +0000 (23:12 +0000)]
res_pjsip_exten_state: Exporting global symbols caused load order issues
Removed the exportation of global symbols from the module as it is no longer
needed and it could potentially cause load problems as on some systems it
would try to load before res_pjsip_pubsub
Kevin Harwell [Fri, 31 Jan 2014 22:17:50 +0000 (22:17 +0000)]
alembic: script modifications due to errors
A couple of the scripts had errors that would not allow a full migration to
take place. The extensions table needed to make its 'id' column a primary
key in order to work with mysql. The other script ...add_endpoints... was
missing tables that it was trying to add columns to.
Added the primary key on id for extensions and added the tables in for the
missing pjsip configuration options. While it is not ideal to modify already
released scripts this was a case where it had to be done due to errors in
the script and lacking a better alternative.
Mark Michelson [Fri, 31 Jan 2014 22:11:18 +0000 (22:11 +0000)]
Decouple subscription handling from NOTIFY/PUBLISH body generation.
When the PJSIP pubsub framework was created, subscription handlers were required
to state what event they handled along with what body types they knew how to
generate. While this serves well when implementing a base RFC, it has problems
when trying to extend the body to support non-standard or proprietary body
elements. The code also was NOTIFY-specific, meaning that when the time comes
that we start writing code to send out PUBLISH requests with MWI or presence
bodies, we would likely find ourselves duplicating code that had previously been
written.
This changeset introduces the concept of body generators and body supplements. A
body generator is responsible for allocating a native structure for a given body
type, providing the primary body content, converting the native structure to a
string, and deallocating resources. A body supplement takes the primary body
content (the native structure, not a string) generated by the body generator and
adds nonstandard elements to the body. With these elements living in their own
module, it becomes easy to extend our support for body types and to re-use
resources when sending a PUBLISH request.
Body generators and body supplements register themselves with the pubsub core,
similar to how subscription and publish handlers had done. Now, subscription
handlers do not need to know what type of body content they generate, but they
still need to inform the pubsub core about what the default body type for a
given event package is. The pubsub core keeps track of what body generators and
body supplements have been registered. When a SUBSCRIBE arrives, the pubsub core
will check that there is a subscription handler for the event in the SUBSCRIBE,
then it will check that there is a body generator that can provide the content
specified in the Accept header(s).
Because of the nature of body generators and supplements, it means
res_pjsip_exten_state and res_pjsip_mwi have been completely gutted. They no
longer worry about body types, instead calling
ast_sip_pubsub_generate_body_content() when they need to generate a NOTIFY body.
Kevin Harwell [Fri, 31 Jan 2014 22:05:32 +0000 (22:05 +0000)]
res_pjsip_mwi: Subscribe fails when missing aor name
When subscribing to MWI (res_pjsip_mwi) and the sip uri did not contain a name
(ex: sip:<ip address>) then the subscription would fail since it would be unable
to locate an associated aor. This patch makes it so that when a subscribe comes
with no aor name then it will subscribe to all aors on the located endpoint.
(closes issue ASTERISK-23072)
Reported by: Bob M
Review: https://reviewboard.asterisk.org/r/3164/
Kinsey Moore [Fri, 31 Jan 2014 15:01:08 +0000 (15:01 +0000)]
PJSIP: Fix address for ACK in NAT situations
In NAT scenarios where a call is placed to a Grandstream phone,
res_pjsip will sometimes send the ACK to a 200 OK to the private
address of the device behind the NAT instead of the address of the NAT
device. This corrects that behavior by rewriting the address in the
Contact header in the incoming 200 OK and the dialog's target address
if necessary (since it has already been rewritten to the incorrect
private address).
(closes issue ASTERISK-23106)
Review: https://reviewboard.asterisk.org/r/3168/
Reported by: Matt Jordan
Corey Farrell [Thu, 30 Jan 2014 20:34:43 +0000 (20:34 +0000)]
res_rtp_asterisk & udptl: fix port selection to work with SELinux restrictions
ast_bind to a port reserved for another program by SELinux causes
errno == EACCES. This caused random failures when binding rtp or
udptl sockets. Treat EACCES as a non-fatal error, try next port.
Kevin Harwell [Tue, 28 Jan 2014 23:35:17 +0000 (23:35 +0000)]
res_pjsip_pubsub: potential crash on timeout
What seems to be happening is if a subscription has been terminated and the
subscription timeout/expires is less than the time it takes for all pending
transactions (currently on the subscription) to end then the subscription
timer will not have been canceled yet and sub will be null. Since the
subscription has already been canceled nothing needs to be done so a null
check in the asterisk code is sufficient in working around this problem.
(closes issue ASTERISK-23129)
Reported by: Dan Jenkins
Asterisk's RADIUS module currently build against libradiusclient-ng, but this
project has been superseeded by libfreeradius-client. The API is 99% compatible
except that the header name has changed, the library name has changed, and
the configuration file location has changed.
Kevin Harwell [Tue, 28 Jan 2014 20:42:41 +0000 (20:42 +0000)]
res_pjsip,compat: INFINITY and NAN undefined
On some systems the values for INFINITY and NAN are not defined thus causing
a build error on those systems. Added definitions for those if they had
not previously been defined.
Kinsey Moore [Tue, 28 Jan 2014 19:13:30 +0000 (19:13 +0000)]
ARI: Make double subscribe respond with success
Currently, attempting to subscribe an application to a device state
that it has already subscribed to will generate a 500 error response.
This will now be treated as a subscription refresh even though ARI
subscriptions don't currently support lifetimes and will respond with
the normal response for a successful subscription (200 OK).
(closes issue ASTERISK-23143)
Reported by: Matt Jordan
rtp_engine: improved handling of get_rtp_info failure
In ast_rtp_instance_make_compatible(), after a failure of
channel tech call get_rtp_info() to return peer_instance,
the null pointer would be passed to ao2_ref, producing an
error that looked like a refernce counting problem but is
not. This patch corrects that and adds helpful LOG_ERROR
messages to indicate which failure path occurred.
Richard Mudgett [Tue, 28 Jan 2014 00:11:16 +0000 (00:11 +0000)]
test_cdr.c, test_cel.c: Correctly destroy created bridges.
* Fixed the test_cel_attended_transfer_bridges_link unit test to also
account for the local channel link being destroyed now that the bridges
are actually destroyed.
* Made CDR unit test use its own version of do_sleep() from the CEL unit
tests.
Russell Bryant [Mon, 27 Jan 2014 20:36:37 +0000 (20:36 +0000)]
Allow nested #includes in extconfig.conf
extconfig.conf was hard-coded to not allow nested includes for some reason.
The code has been this way since a patch was merged for ASTERISK-3333 (revision
4889), which was a significant update to this code ("Merge config updates").
I can't figure out any good reason why this should be limited. This patch just
removes the limit and uses the default nesting depth limit.
Russell Bryant [Mon, 27 Jan 2014 01:19:18 +0000 (01:19 +0000)]
Protect ast_filestream object when on a channel
The ast_filestream object gets tacked on to a channel via
chan->timingdata. It's a reference counted object, but the reference
count isn't used when putting it on a channel. It's theoretically
possible for another thread to interfere with the channel while it's
unlocked and cause the filestream to get destroyed.
Use the astobj2 reference count to make sure that as long as this code
path is holding on the ast_filestream and passing it into the file.c
playback code, that it knows it's valid.
Joshua Colp [Sun, 26 Jan 2014 02:10:22 +0000 (02:10 +0000)]
res_pjsip_session: Be less strict with core requested outgoing capabilities.
The core may (depending on circumstances) request a single codec on outgoing
calls. Many channel drivers ignore or treat this as a suggestion while still
including configured codecs. The res_pjsip_session logic treated this as
an explicit request, leaving out other configured codecs.
This change makes res_pjsip_session behave like other channel driver and simply
adds the requested codec to the list.
Richard Mudgett [Fri, 24 Jan 2014 23:29:57 +0000 (23:29 +0000)]
CEL: Protect data structures during reload and shutdown.
The CEL data structures need to be protected during a configuration reload
and shutdown. Asterisk crashed during a shutdown because CEL events were
still in flight and the CEL data structures were already destroyed.
* Protected the cel_backends, cel_dialstatus_store, and cel_linkedids ao2
containers with a global ao2 object wrapper.
* Added NULL checks before use of the cel_backends, cel_dialstatus_store,
and cel_linkedids ao2 containers in case the CEL module is already
shutdown.
* Fixed overloading of the cel_linkedids held objects reference count.
During shutdown any held objects would be leaked.
* Fixed memory leak of cel_linkedids held objects if the LINKEDID_END is
not being tracked. The objects in the cel_linkedids container were not
removed if the LINKEDID_END event is not used.
* Added access protection to the cel_backends container during the CLI
"cel show status" command.
* Made cel_backends, cel_dialstatus_store, and cel_linkedids use the
standard ao2 callback templates for the hash and cmp functions.
* Eliminated unnecessary uses of RAII_VAR().
* Made ast_cel_engine_init() cleanup alocated resources on failure.
Richard Mudgett [Fri, 24 Jan 2014 18:04:54 +0000 (18:04 +0000)]
manager: Protect data structures during shutdown.
Occasionally, the manager module would get an "INTERNAL_OBJ: bad magic
number" error on a "core restart gracefully" command if an AMI connection
is established.
* Added ao2_global_obj protection to the sessions global container.
* Fixed the order of unreferencing a session object in session_destroy().
* Removed unnecessary container traversals of the white/black filters
during session_destructor().
Michael L. Young [Thu, 23 Jan 2014 22:54:26 +0000 (22:54 +0000)]
res_config_mysql: Fix Setting The Column Name Incorrectly
When support for a realtime sorcery module was added in revision 386731, the
wrong property was accidentally used for setting the column name to be updated
in the database table. This patch fixes the typo.
(closes issue ASTERISK-23177)
Reported by: Denis
Tested by: Denis
Patches:
asterisk-23177-use-field-name.diff by Michael L. Young (license 5026)
Mark Michelson [Thu, 23 Jan 2014 21:09:35 +0000 (21:09 +0000)]
Fix presence body errors found during testing:
* PIDF bodies were reporting an "open" state in many cases where
it should have been reporting "closed"
* XPIDF bodies had XML nodes placed incorrectly within the hierarchy.
* SIP URIs in XPIDF bodies did not go through XML sanitization
* XML sanitization had some errors:
* Right angle bracket was being replaced with "&rt;" instead of ">"
* Double quote, apostrophe, and ampersand were not being escaped.
Mark Michelson [Thu, 23 Jan 2014 21:00:24 +0000 (21:00 +0000)]
Fix presence body errors found during testing:
* PIDF bodies were reporting an "open" state in many cases where
it should have been reporting "closed"
* XPIDF bodies had XML nodes placed incorrectly within the hierarchy.
* SIP URIs in XPIDF bodies did not go through XML sanitization
* XML sanitization had some errors:
* Right angle bracket was being replaced with "&rt;" instead of ">"
* Double quote, apostrophe, and ampersand were not being escaped.
pbx.c: Pre-initialize timezone to avoid crash on destroy
In ast_build_timing, initialize the timezone value to NULL
in order to avoid deferencing an uninitialized value later
when calling ast_destroy_timing. The timezone value could
be uninitialized if ast_build_timing were to fail due to a
zero length time string.
(closes issue ASTERISK-22861)
Reported by: Sebastian Murray-Roberts
Review: https://reviewboard.asterisk.org/r/3134/
Patches:
ast_build_timing-initialize-timezone.patch uploaded by coreyfarrell (license 5909)
........
Merged revisions 406241 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 406245 from http://svn.asterisk.org/svn/asterisk/branches/11
Kinsey Moore [Wed, 22 Jan 2014 19:34:15 +0000 (19:34 +0000)]
ConfBridge: Fix channel parameter documentation
Confbridge AMI and CLI commands for mute, unmute, and setting the
single video source can accept channel prefixes in lieu of a full
channel name, but documentation states only that it is required and is
a channel name. This corrects the documentation.
(closes issue PQ-1397)
Reported by: Steve Pitts
........
Merged revisions 406217 from http://svn.asterisk.org/svn/asterisk/branches/11
Kinsey Moore [Wed, 22 Jan 2014 18:32:21 +0000 (18:32 +0000)]
chan_sip: Decline image streams on unsupported transports
This change allows chan_sip to decline individual image streams over
unsupported transports in the SDP of the 200 response. Previously,
an image stream offer with RTP/AVP as the transport would cause
chan_sip to respond with a 488.
Kinsey Moore [Wed, 22 Jan 2014 14:00:01 +0000 (14:00 +0000)]
res_stasis_playback: Correct error argument order
Several of the playback error messages for invalid media input in
res_stasis_playback.c had the media name and channel name reversed.
They now correctly identify the channel name and media name.
Rusty Newton [Tue, 21 Jan 2014 21:47:13 +0000 (21:47 +0000)]
res_pjsip: Documentation improvement for Endpoint and AOR mailbox options.
Making the help text for both more explicit regarding the format of mailbox identifiers. i.e. clarifying the format for app_voicemail mailboxes vs mailboxes from external MWI sources through modules such as res_external_mwi.
Kinsey Moore [Tue, 21 Jan 2014 17:14:24 +0000 (17:14 +0000)]
PJSIP: Handle headers in a list appropriately
The PJSIP header parsing function (pjsip_parse_hdr) can generate more
than one header instance from a single header field. These header
instances exist as a list attached to the returned header and must be
handled appropriately when they are added to a message or else only the
first header instance will be used. This changes the linked list
functions used in outbound proxy code to merge the lists properly.
Kinsey Moore [Tue, 21 Jan 2014 14:15:21 +0000 (14:15 +0000)]
ARI: Support channel variables in originate
This adds back in support for specifying channel variables during an
originate without compromising the ability to specify query parameters
in the JSON body. This was accomplished by generating the body-parsing
code in a separate function instead of being integrated with the URI
query parameter parsing code such that it could be called by paths with
body parameters. This is transparent to the user of the API and
prevents manual duplication of code or data structures.
(closes issue ASTERISK-23051)
Review: https://reviewboard.asterisk.org/r/3122/
Reported by: Matt Jordan
Damien Wedhorn [Mon, 20 Jan 2014 21:53:02 +0000 (21:53 +0000)]
Skinny: fix up session logging.
Logging from the skinny session loop was providing some incorrect reasons
for exiting the loop. Cleaned up messages and handling so correct reason
displayed.
Damien Wedhorn [Sat, 18 Jan 2014 23:57:57 +0000 (23:57 +0000)]
Skinny: fix reversed device reset from CLI.
Existing code would do a full device restart when "skinny reset device"
was entered at the CLI and do a reset when "skinny reset device restart"
entered.
projects / thirdparty / asterisk.git / log
