Richard Mudgett [Fri, 18 Apr 2014 16:44:48 +0000 (16:44 +0000)]
Originated calls: Fix several originate call problems.
* Restore the reason value set by pbx_outgoing_attempt() to use
AST_CONTROL_xxx values as all the consumers were expecting rather than
cause codes.
* Fixed the dial routines to set cause codes for more than just
ast_request() so pbx_outgoing_attempt() reason codes will function.
* Fix inconsistent locked_channel return status in pbx_outgoing_attempt().
The chanel may not have been locked or the channel may have been a stale
pointer.
* Fixed the OutgoingSpoolFailed channel to run dialplan whenever the
dialing fails for an originate exten and 1 < synchronous.
* Fix incorrect ast_cond_wait() usage in pbx_outgoing_attempt().
Indroduced by issue ASTERISK-22212 patch.
* Made struct pbx_outgoing use the ao2 lock instead of its own lock for
the cond wait mutex. No sense in having two locks associated with the
same struct when only one is needed.
Richard Mudgett [Fri, 18 Apr 2014 16:27:31 +0000 (16:27 +0000)]
app_dial and app_queue: Make lock the forwarding channel while taking the channel snapshot.
* Fixed ast_channel_publish_dial_forward() not locking the forwarded
channel when taking the channel snapshot.
* Fixed app_dial.c:do_forward() using the wrong channel to get the
original call forwarding string.
* Removed unnecessary locking when calling ast_channel_publish_dial() and
ast_channel_publish_dial_forward() in app_dial and app_queue. Holding
channel locks when calling ast_channel_publish_dial_forward() with a
forwarded channel could result in pausing the system while the stasis bus
completes processsing a forwarded channel subscription.
Kinsey Moore [Fri, 18 Apr 2014 14:25:47 +0000 (14:25 +0000)]
ARI: Add debug logging for events and responses
This adds DEBUG level logging for ARI websocket events and HTTP
responses similar to what is available for AMI. Logging for ARI HTTP
requests is already adequate for debugging purposes.
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Merged revisions 412565 from http://svn.asterisk.org/svn/asterisk/branches/12
res_pjsip: Handle reloading when permanent contacts exist and qualify is configured.
This change fixes a problem where permanent contacts being qualified were not
being updated. This was caused by the permanent contacts getting a uuid and not a
known identifier, causing an inability to look them up when updating in the
qualify code. A bug also existed where the new configuration may not be available
immediately when updating qualifies.
(closes issue ASTERISK-23514)
Reported by: Richard Mudgett
Jonathan Rose [Thu, 17 Apr 2014 21:57:36 +0000 (21:57 +0000)]
ARI: Add tones playback resource
Adds a tones URI type to the playback resource. The tone can be specified by
name (from indications.conf) or by a tone pattern. In addition, tonezone can
be specified in the URI (by appending ;tonezone=<zone>). Tones must be
stopped manually in order for a stasis control to move on from playback of
the tone. Tones may be paused, resumed, restarted, and stopped. They may
not be rewound or fast forwarded (tones can't be controlled in a way that
lets you skip around from note to note and pausing and resuming will also
restart the tone from the beginning). Tests are currently in development
for this feature (https://reviewboard.asterisk.org/r/3428/).
(closes issue ASTERISK-23433)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3427/
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Merged revisions 412535 from http://svn.asterisk.org/svn/asterisk/branches/12
Matthew Jordan [Thu, 17 Apr 2014 20:25:16 +0000 (20:25 +0000)]
main/Makefile: Fix build failure on SmartOS/Illumos/SunOS
This patch fixes two issues when building on SmartOS:
- channels/chan_oss.c: it makes sure soundcard.h is found
- main/Makefile: only use "-Wl,--version-script" when GNU LD is used as the Sun
Linker doesn't support that. Similar checks are already used elswhere in the
Makefile
Review: https://reviewboard.asterisk.org/r/3426
ASTERISK-23576 #close
Reported by: Sebastian Wiedenroth
patches:
fix-sunos.diff uploaded by Sebastian Wiedenroth (License 6597)
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Merged revisions 412468 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 412483 from http://svn.asterisk.org/svn/asterisk/branches/12
Matthew Jordan [Thu, 17 Apr 2014 19:50:05 +0000 (19:50 +0000)]
chan_sip: Add SIPURIPHONECONTEXT channel variable for Request TEL URIs
This patch is a continuation of https://reviewboard.asterisk.org/r/3349/,
committed in r412303.
It resolves a finding oej had that the phone-context be available in a
channel variable separate from SIPDOMAIN. This patch adds that variable as
SIPURIPHONECONTEXT. It also allows a local number (or global number specified
in the TEL URI) to be used to look up as a peer.
Kevin Harwell [Thu, 17 Apr 2014 15:17:39 +0000 (15:17 +0000)]
res_pjsip_refer: Channel variable SIPREFERTOHDR not being set during blind transfer
The SIPREFERTOHDR channel variable is not being set on any channel when
performing a blind transfer using PJSIP. The 'refer->refer_to' was not
being set during a blind transfer. Updated so the 'refer_to' is set to
the target uri on a blind transfer.
(closes issue ASTERISK-23502)
Reported by: John Bigelow
Review: https://reviewboard.asterisk.org/r/3445/
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Merged revisions 412453 from http://svn.asterisk.org/svn/asterisk/branches/12
Russell Bryant [Tue, 15 Apr 2014 23:21:19 +0000 (23:21 +0000)]
(mix)monitor: Add options to enable a periodic beep
Add an option to enable a periodic beep to be played into a call if it
is being recorded. If enabled, it uses the PERIODIC_HOOK() function
internally to play the 'beep' prompt into the call at a specified
interval. This option is provided for both Monitor() and
MixMonitor().
Richard Mudgett [Tue, 15 Apr 2014 18:01:47 +0000 (18:01 +0000)]
Remove unused RAII_VAR() declarations.
* Remove unused RAII_VAR() declarations. The compiler cannot catch these
because the cleanup function "references" the unused variable. Some
actually allocated and released resources that were never used.
* Fixed some whitespace issues in stasis_bridges.c.
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Merged revisions 412399 from http://svn.asterisk.org/svn/asterisk/branches/12
The failing assertion ensures that the final snapshot gets generated so
CDR records can get finalized. The only place where a channel staging
snapshot flag could be left set is in chan_sip.c:handle_request_bye().
The function could return before clearing the flag because the channel
could dissappear while the function had to have the channel unlocked.
* Fixed handle_request_bye() channel snapshot staging coverage area to not
have a return in the middle of it and be unable to clear the staging flag.
* Pushed the channel snapshot staging coverage area into
ast_rtp_instance_set_stats_vars() to ensure that the staging is not
interrutped.
* Made callers of ast_rtp_instance_set_stats_vars() not call it with any
channels or channel driver private locks held to eliminate the deadlock
potential. The callers must hold references to the passed in channel and
rtp objects.
* Eliminated sip_hangup() trying to get the bridge peer. It is futile at
this point because the channel could never be in a bridge.
Richard Mudgett [Tue, 15 Apr 2014 16:38:35 +0000 (16:38 +0000)]
chan_sip.c: Moved some sip_pvt unrefs after their last use.
* Moved sip_pvt unref in ast_hangup() and handle_request_do() to the end
of the function. The unref needs to happen after the last use of the
pointer.
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Merged revisions 412348 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 412383 from http://svn.asterisk.org/svn/asterisk/branches/12
chan_sip: Send real CallerID information with P-Assserted-Identity (RFC-3325)
Prior to this patch, the P-Asserted-Identity header would include anonymous
caller id information which seems to go against the point of the
P-Asserted-Identity header. Now the real caller ID information will be
included in this header. Also, no privacy header would be included.
This patch adds 'Privacy: id' to outgoing SIP messages that include the
P-Asserted-Identity header.
(closes issue AST-1301)
---
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Merged revisions 412328 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 412330 from http://svn.asterisk.org/svn/asterisk/branches/12
autoservice acquires a local reference to the logger callid of each channel
in a loop. This local reference was not released, causing the callid of
every channel in autoservice to leak. This change moves the callid unref
inside the loop.
Matthew Jordan [Sat, 12 Apr 2014 02:27:43 +0000 (02:27 +0000)]
chan_sip: Support RFC-3966 TEL URIs in inbound INVITE requests
This patch adds support for handling TEL URIs in inbound INVITE requests.
This includes the Request URI and the From URI. The number specified in
the Request URI will be the destination of the inbound channel in the dialplan.
The phone-context specified in the Request URI will be stored in the
TELPHONECONTEXT channel variable.
Review: https://reviewboard.asterisk.org/r/3349
ASTERISK-17179 #close
Reported by: Geert Van Pamel
Tested by: Geert Van Pamel
patches:
asterisk-12.0.0-chan_sip-RFC3966_patch.txt uploaded by Geert Van Pamel (License 6140)
asterisk-12.0.0-reqresp_parser-RFC3966_patch.txt uploaded by Geert Van Pamel (License 6140)
Russell Bryant [Sat, 12 Apr 2014 01:35:34 +0000 (01:35 +0000)]
func_periodic_hook: move module ref
The previous code left one error path where the module would be unref'd twice
instead of once. It was done once in the error handling block, and again inside
of datastore destruction.
Now the module ref is only released in the datastore destructor and only acquired
when the datastore has been successfully allocated.
Russell Bryant [Sat, 12 Apr 2014 01:26:28 +0000 (01:26 +0000)]
func_periodic_hook: add module ref counting
This module lacked necessary module ref count incrementing and decrementing
when used. This patch adds it. There's already a datastore used, so doing the
ref counting along with the lifetime of the datastore provides a convenient
place to do it.
Kinsey Moore [Fri, 11 Apr 2014 12:43:34 +0000 (12:43 +0000)]
bridging: Ensure locking during snapshot creation
While the vast majority of bridge snapshot creation is locked properly,
there are currently some instances that are not. This adds the missing
locking to ensure bridge state is not malleable during snapshot
creation.
(closes issue ASTERISK-22904)
Review: https://reviewboard.asterisk.org/r/3415/
Reported by: Matt Jordan
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Merged revisions 412193 from http://svn.asterisk.org/svn/asterisk/branches/12
Matthew Jordan [Fri, 11 Apr 2014 02:59:19 +0000 (02:59 +0000)]
main/astobj2: Make REF_DEBUG a menuselect item; improve REF_DEBUG output
This patch does the following:
(1) It makes REF_DEBUG a meneselect item. Enabling REF_DEBUG now enables
REF_DEBUG globally throughout Asterisk.
(2) The ref debug log file is now created in the AST_LOG_DIR directory.
Every run will now blow away the previous run (as large ref files
sometimes caused issues). We now also no longer open/close the file
on each write, instead relying on fflush to make sure data gets written
to the file (in case the ao2 call being performed is about to cause a
crash)
(3) It goes with a comma delineated format for the ref debug file. This
makes parsing much easier. This also now includes the thread ID of the
thread that caused ref change.
(4) A new python script instead for refcounting has been added in the
contrib/scripts folder.
(5) The old refcounter implementation in utils/ has been removed.
Russell Bryant [Fri, 11 Apr 2014 01:12:54 +0000 (01:12 +0000)]
monitor: use app options parsing helper code
This app is pretty ancient, so it was never converted to use the
option parsing helper code. I'd like to add an option to this app
that takes an argument, and that's a pain to do when not using this
helper, so start by doing this conversion.
Matthew Jordan [Thu, 10 Apr 2014 21:28:08 +0000 (21:28 +0000)]
res_hep_pjsip: Use the channel name instead of the call ID when it is available
During discussions with Alexandr Dubovikov at Kamailio World, it became
apparent that while the SIP call ID is a useful identifier prior to an Asterisk
channel being created, it is far more preferable to use the channel name (or
some channel based identifier) when the channel is available. Homer is smart
enough to tie the various messages together. This patch opts to use the channel
name when it is available, falling back to the call ID otherwise.
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Merged revisions 412088 from http://svn.asterisk.org/svn/asterisk/branches/12
Kevin Harwell [Thu, 10 Apr 2014 21:10:46 +0000 (21:10 +0000)]
res_pjsip_pubsub: Set the body generation result to 0 for a valid path
The result of the "ast_sip_pubsub_generate_body_content" was not
set/initialized. Consequently, the nominal path potentially returned
an invalid value, thus not sending mwi notifications.
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Merged revisions 412074 from http://svn.asterisk.org/svn/asterisk/branches/12
Mark Michelson [Wed, 9 Apr 2014 21:43:23 +0000 (21:43 +0000)]
Add a Command header to the AMI Mixmonitor action.
This fixes a parsing error that occurred during the processing of
the AMI action. The error did not result in MixMonitor itself
misbehaving, but it could result in the AMI response not giving
correct information back.
The new header allows for one to specify a post-process command
to run when recording finishes. Previously, in order to do this,
the post-process command would have to be placed at the end of
the Options: header.
Patches: mixmonitor_command_2.patch by jhardin (License #6512)
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Merged revisions 412048 from http://svn.asterisk.org/svn/asterisk/branches/12
Richard Mudgett [Tue, 8 Apr 2014 21:25:15 +0000 (21:25 +0000)]
Internal timing: Add notice that the -I and internal_timing option are no longer needed.
Add notice messages during execution that the -I command line option and
the astersik.conf internal_timing option are no longer needed. The
internal timing functionality is now always enabled if there is a timing
module loaded.
NOTE: Since the command line options and the asterisk.conf config file are
processed before the logging system is initialized, the messages are
output to stderr.
Change requested as a result of asterisk-dev list comments about the
commit for ASTERISK-22846 that removed the -I and internal_timing options.
res_pjsip: Ignore explicit transport configuration if a WebSocket transport is specified.
This change makes it so if a transport is configured on an endpoint that is a WebSocket
type the option will be ignored. In practice this is fine because the WebSocket
transport can not create outgoing connections, it can only reuse existing ones. By
ignoring the option the existing PJSIP logic for using the existing connection will
be invoked and stuff will proceed.
(closes issue ASTERISK-23584)
Reported by: Rusty Newton
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Merged revisions 411927 from http://svn.asterisk.org/svn/asterisk/branches/12
Russell Bryant [Tue, 8 Apr 2014 00:26:57 +0000 (00:26 +0000)]
func_periodic_hook: List more modules as dependencies
This module makes use of some existing Asterisk components. app_chanspy was
already listed as a dependency. There are a few function modules used, as
well, so list them.
Kinsey Moore [Mon, 7 Apr 2014 20:41:05 +0000 (20:41 +0000)]
PJSIP: Ensure test event has new state
The change that fixed the pubsub test event's use of a dangling pointer
also changed when it was processed relative to the pjsip subscription
state change processing. This change corrects the order of events while
holding a reference to the pointer that was previously dangling.
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Merged revisions 411883 from http://svn.asterisk.org/svn/asterisk/branches/12
Jonathan Rose [Mon, 7 Apr 2014 16:15:34 +0000 (16:15 +0000)]
AGI/Manager: Prevent multiple NewExten events during AGI application changes
AGI applications would trigger NewExten events every time the state of the AGI
application changed. This has historically not been the behavior and this
behavior was introduced with a CDR patch. This patch corrects that.
(closes issue ASTERISK-23390)
Reported by: Benjamin Keith Ford
Review: https://reviewboard.asterisk.org/r/3406/
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Merged revisions 411868 from http://svn.asterisk.org/svn/asterisk/branches/12
Kinsey Moore [Mon, 7 Apr 2014 14:29:37 +0000 (14:29 +0000)]
Stasis: Fix Stasis() bridge refcount issue
The Stasis() dialplan application monitors what bridge a channel is in
and so necessarily holds on to a bridge pointer. This change ensures
that it also holds on to a reference for that bridge to prevent the
bridge pointer from becoming a dangling pointer.
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Merged revisions 411804 from http://svn.asterisk.org/svn/asterisk/branches/12
Kinsey Moore [Mon, 7 Apr 2014 13:30:25 +0000 (13:30 +0000)]
PJSIP: Fix crash introduced in r411671
The test event introduced in revision 411671 uses a dangling pointer to
access information about pubsub state changes. This moves the event to
within the lifetime of the pointer.
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Merged revisions 411790 from http://svn.asterisk.org/svn/asterisk/branches/12
Russell Bryant [Sat, 5 Apr 2014 13:06:34 +0000 (13:06 +0000)]
func_periodic_hook: New function for periodic hooks.
This commit introduces a new dialplan function, PERIODIC_HOOK().
It allows you run to a dialplan hook on a channel periodically. The
original use case that inspired this was the ability to play a beep
periodically into a call being recorded. The implementation is much
more generic though and could be used for many other things.
The implementation makes heavy use of existing Asterisk components.
It uses a combination of Local channels and ChanSpy() to run some
custom dialplan and inject any audio it generates into an active call.
The other important bit of the implementation is how it figures out
when to trigger the beep playback. This implementation uses the
audiohook API, even though it's not actually touching the audio in any
way. It's a convenient way to get a callback and check if it's time
to kick off another beep. It would be nice if this was timer event
based instead of polling based, but unfortunately I don't see a way to
do it that won't interfere with other things.
Richard Mudgett [Fri, 4 Apr 2014 19:19:55 +0000 (19:19 +0000)]
internal_timing: Remove the option and always make it enabled if a timing module is loaded.
The masquerade supertest frequently fails because either the local channel
chain doesn't completely optimize out or the DTMF handshake doesn't
completely get accross. Local channel optimization requires frames
flowing to trigger when optimization can happen. When optimization
happens the media frame that triggered the optimization is dropped.
Sending DTMF requires frames to flow in the other direction for timing
purposes while sending nothing. If internal timing is not enabled when
MOH is playing, Asterisk switches to received timing when an audio frame
is received. With optimization dropping media frames and MOH not sending
frames unless it receives frames, occasionaly there are no more frames
being passed and the test fails.
* The asterisk command line -I option and the asterisk.conf
internal_timing option are removed. Asterisk now always uses internal
timing when needed if any timing module is loaded. The issue
ASTERISK-14861 did this quite awhile ago in v1.4 but effectively is broken
if other internal timing modules besides DAHDI are used. The
ast_read_generator_actions() now only does received timing if it has no
choice for frame generators like MOH, silence, and playback streaming.
* Cleaned up some code dealing with frame generators in
ast_deactivate_generator(), generator_write_format_change(),
ast_activate_generator(), and ast_channel_stop_silence_generator().
* Removed ast_internal_timing_enabled(), AST_OPT_FLAG_INTERNAL_TIMING, and
ast_opt_internal_timing.
Richard Mudgett [Fri, 4 Apr 2014 17:57:46 +0000 (17:57 +0000)]
Add some asserts that were handy when looking for a stasis cache problem.
* Assert if a channel is destroyed but has the snapshot staging flag set.
In this case the final channel destruction snapshot would never get taken.
* Assert if what we just got out of the stasis cache is not what we were
looking for. This assert would have saved several days searching for a
bug and a lot of my hair.
* Assert if the music on hold message posts could not find the associated
channel. A crash will happen later when manager tries to send the MOH AMI
message. This assert catches the problem when the stasis message is
posted instead of by the thread processing the defective message.
* Always generate a backtrace when an ast_assert() fails.
Matthew Jordan [Fri, 4 Apr 2014 15:13:55 +0000 (15:13 +0000)]
http: Fix spurious ERROR message in responses with no content
When a response has a content length of 0, fwrite would be called to write a
buffer with no data in it. This resulted in the following classic error
message:
Matthew Jordan [Thu, 3 Apr 2014 11:47:03 +0000 (11:47 +0000)]
res_hep: Fix crash when hep.conf not available
Parts of res_hep properly checked for a valid configuration object before
attempting to access the configuration. A check, however, was missed when
a packet is sent. This patch fixes the crash caused by not checking if the
configuration object is valid.
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Merged revisions 411668 from http://svn.asterisk.org/svn/asterisk/branches/12
Mark Michelson [Wed, 2 Apr 2014 18:57:29 +0000 (18:57 +0000)]
Prevent duplicate sorcery wizards from being applied to sorcery object types.
This commit contains several changes to sorcery:
1) Application of sorcery configuration based on module name is automatically performed
when sorcery is opened for a module.
2) Sorcery will not attempt to apply the same wizard to an object type more than once.
3) Sorcery gives more exact results when attempting to apply a wizard, whether as the
default or based on configuration.
Sorcery unit tests still pass for me after making these changes.
app_queue: Fix a bug where realtime members would be deleted during reload causing waiting callers to get ejected.
This patch causes realtime queue members to remain in queues during the reload process. Previously these
members would be removed causing any waiting callers to be ejected from the queue with a reason of "EXITEMPTY".
Matthew Jordan [Fri, 28 Mar 2014 18:32:50 +0000 (18:32 +0000)]
res_hep/res_hep_pjsip: Add a HEPv3 capture agent module and a logger for PJSIP
This patch adds the following:
(1) A new module, res_hep, which implements a generic packet capture agent for
the Homer Encapsulation Protocol (HEP) version 3. Note that this code is based
on a patch provided by Alexandr Dubovikov; I basically just wrapped it up,
added configuration via the configuration framework, and threw in a
taskprocessor.
(2) A new module, res_hep_pjsip, which forwards all SIP message traffic that
passes through the res_pjsip stack over to res_hep for encapsulation and
transmission to a HEPv3 capture server.
Much thanks to Alexandr for his Asterisk patch for this code and for a *lot*
of patience waiting for me to port it to 12/trunk. Due to some dithering on
my part, this has taken the better part of a year to port forward (I still
blame CDRs for the delay).
Alexandr Anikin [Fri, 28 Mar 2014 18:00:18 +0000 (18:00 +0000)]
process stack command even if gatekeeper client isn't register
don't destroy gatekeeper client if it is not started
don't destroy gatekeeper client in some sort of gatekeeper errors
signal rtp create condition when call cleared before rtp structure created
Matthew Jordan [Fri, 28 Mar 2014 17:41:23 +0000 (17:41 +0000)]
Update API versions and UPGRADE/CHANGES for 12.2.0
This patch does the following:
* It updates the AMI version to 2.2.0 to indicate backwards compatible
changes have been made since the last release
* It updates the ARI version to 1.2.0 to indicate backwards compatible
changes have been made since the last release
* It updates the UPGRADE/CHANGES files with changes that were not
mentioned
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Merged revisions 411529 from http://svn.asterisk.org/svn/asterisk/branches/12
Matthew Jordan [Fri, 28 Mar 2014 17:09:14 +0000 (17:09 +0000)]
res_config_odbc: Fix for nullable integer columns and keyfield existence check in update_odbc.
This patch fixes setting nullable integer columns to NULL instead of an empty
string, which fails for PostgreSQL, for example. The current code is supposed
to do so, but the check is broken. The patch also allows the first column in
the list to be a nullable integer.
Also, the check for existence of a mandatory column checked for the first
column in the list instead of the key field lookup column. This patch fixes
that issue as well.
Finally, the compatibility option allow_empty_string_in_nontext, which was
added to previous revisions to allow for some database backends with certain
schemas to function, has been removed.
Matthew Jordan [Fri, 28 Mar 2014 16:49:09 +0000 (16:49 +0000)]
Blocked revisions 411512
........
res_config_odbc/res_odbc: Fix handling of non-text columns updates with empty values.
This patch fixes setting nullable integer columns to NULL instead of an empty
string, which fails for PostgreSQL, for example. The current code is supposed
to do so, but the check is broken. The patch also allows the first column in
the list to be a nullable integer.
This patch also adds a compatibility setting in res_odbc.conf,
allow_empty_string_in_nontext. It is enabled by default. It should be disabled
for database backends (such as PostgreSQL) that require NULL instead of an
empty string for Integer columns.
http: response body often missing after specific request
This patch works around a problem with the HTTP body
being dropped from the response to a specific client
and under specific circumstances:
a) Client request comes from node.js user agent
"Shred" via use of swagger-client library.
b) Asterisk and Client are *not* on the same
host or TCP/IP stack
In testing this problem, it has been determined that
the write of the HTTP body is lost, even if the data
is written using low level write function. The only
solution found is to instruct the TCP stack with the
shutdown function to flush the last write and finish
the transmission. See review for more details.
Matthew Jordan [Fri, 28 Mar 2014 14:19:20 +0000 (14:19 +0000)]
contrib/realtime: Remove empty SQL script files
Since the relatime scripts are now managed by Alembic, the previous realtime
scripts were previously removed. However, the removal process messed up, as
the files were still in the repository. The contents were just empty.
This removes the files from the tree.
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Merged revisions 411442 from http://svn.asterisk.org/svn/asterisk/branches/12
Matthew Jordan [Fri, 28 Mar 2014 03:55:26 +0000 (03:55 +0000)]
chan_sip: Add MESSAGE request to allowed methods
The allowed methods advertised by chan_sip did not previously note the MESSAGE
request. Even in Asterisk 1.8, we do accept in-dialog MESSAGE requests; we
should advertise that we support MESSAGE requests.
ASTERISK-23504 #close
ASTERISK-23504 #comment Reported by: Martin Kontsek
ASTERISK-23504 #comment Patch sip.h_patch.diff uploaded by Martin Kontsek (license 6587)
Mark Michelson [Thu, 27 Mar 2014 14:21:15 +0000 (14:21 +0000)]
Give sorcery instances a reference to their wizards.
On graceful shutdown, sorcery wizards are all killed off, but it is
possible for sorcery instances to still have dangling pointers after
this, possibly causing a crash. Giving the sorcery instances a reference
to their wizards ensures that the wizard reference will remain valid for
the lifetime of the sorcery instance.
Joshua Colp [Wed, 26 Mar 2014 22:45:10 +0000 (22:45 +0000)]
say: Fix a bug where SayNumber in Polish tries to play incorrect sound.
This change fixes a bug where calling SayNumber with a number divisible by
100 using the Polish language would cause the code to attempt to play a
sound file with an empty name.
Jonathan Rose [Wed, 26 Mar 2014 16:15:12 +0000 (16:15 +0000)]
chan_sip: Send real CallerID information with P-Assserted-Identity (RFC-3325)
Prior too this patch, the P-Asserted-Identity header would include anonymous
caller id information which seems to go against the point of the
P-Asserted-Identity header. Now the real caller ID information will be
included in this header. Also, no privacy header would be included.
This patch adds 'Privacy: id' to outgoing SIP messages that include the
P-Asserted-Identity header.
(closes issue AST-1301)
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Merged revisions 411189 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 411190 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 411193 from http://svn.asterisk.org/svn/asterisk/branches/12
Sean Bright [Tue, 25 Mar 2014 18:44:57 +0000 (18:44 +0000)]
ARI: Don't complain about missing ARI users when we aren't enabled
Currently, if ARI is not enabled it will still complain that there are no
configured users. This patch checks to see if ARI is enabled before logging and
error or iterating the container to validate the users.
Richard Mudgett [Tue, 25 Mar 2014 16:57:41 +0000 (16:57 +0000)]
res_pjsip: Fix contact authenticate_qualify endpoint lookup when qualifing a contact.
* Fixed bad use of ao2_find() in on_endpoint().
* Replaced use of find_endpoints() with find_an_endpoint() since only the
first found endpoint is ever needed.
* Fixed qualify_contact_cb() to update the contact with the aor
authenticate_qualify setting. Otherwise, permanent contacts in the aor
type sections would have a config line order dependancy.
* Fixed off nominal path contact ref leak in qualify_contact(). The
comment saying the unref is not needed was wrong.
* Fixed off nominal path use of the endpoint parameter if it is NULL in
send_out_of_dialog_request().
* Added missing off nominal path unref of pjsip tdata in
send_out_of_dialog_request().
* Fixed off nominal path failing to call the callback in send_request_cb()
when the request is challenged for authentication.
* Eliminated silly RAII_VAR() use in qualify_contact_cb().
* Updated ast_sip_send_request() doxygen to better reflect reality.
Kinsey Moore [Tue, 25 Mar 2014 16:06:57 +0000 (16:06 +0000)]
chan_sip: Fix incorrect use of timers
If update_provisional_keepalive() is called while
send_provisional_keepalive_full() is waiting on the PVT lock, then
pvt->provisional_keepalive_sched_id will be changed to a new sched_id
value by update_provisional_keepalive(), but that new sched_id then may
be overwritten with -1 by send_provisional_keepalive_full(), killing
the pvt's reference to a schedule and "leaking" the reference.
(closes issue ASTERISK-22079)
Review: https://reviewboard.asterisk.org/r/3368/
Reported by: Jamuel Starkey, Matteo, Leif Madsen, Steve Davies
Patches:
provisional_keepalive_fix.diff uploaded by Steve Davies (license 5012)
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Jonathan Rose [Tue, 25 Mar 2014 15:56:05 +0000 (15:56 +0000)]
ARI: Resolve a subscription leak against implicit bridge subscriptions
When a channel in a stasis application is joined to a bridge, a subscription
for that bridge is created implicitly for the stasis application serving the
channel. Prior to this patch, subsequent removals of the channel from the
bridge would leave the subscription open.
Richard Mudgett [Thu, 20 Mar 2014 16:35:57 +0000 (16:35 +0000)]
assigned-uniqueids: Miscellaneous cleanup and fixes.
* Fix memory leak in ast_unreal_new_channels(). Made it generate the ;2
uniqueid on a stack variable instead of mallocing it.
* Made send error response to ARI and AMI requests instead of just logging
excessive uniqueid length and allowing truncation. action_originate() and
ari_channels_handle_originate_with_id().
* Fixed minor truncating uniqueid hole when generating the ;2 uniqueid
string length. Created public and internal lengths of uniqueid. The
internal length can handle a max public uniqueid plus an appended ;2.
* free() and ast_free() are NULL tolerant so they don't need a NULL test
before calling.
* Made use better struct initialization format instead of the position
dependent initialization format. Also anything not explicitly initialized
in the struct is initialized to zero by the compiler.
* Made ast_channel_internal_set_fake_ids() use the safer
ast_copy_string() instead of strncpy().
Mark Michelson [Wed, 19 Mar 2014 17:27:57 +0000 (17:27 +0000)]
PJSIP: Allow for identify sections to be specified in sorcery.conf.
"identify" is a special type of configuration object in PJSIP because
unlike the other objects, it is not provided by the base res_pjsip module.
Instead, it is provided by the res_pjsip_endpoint_identifier_ip module. If
using the default sorcery wizard (config,criteria=type=identify) then things
work because the module that applies the default wizard is the correct module.
However, if attempting to use sorcery.conf to apply an alternate wizard, it
was not possible. If you attempted to specify the identify object type in the
res_pjsip section, then the object could not be registered since the object
was undocumented for the res_pjsip module. There was no alternate configuration
section defined for it, so you were out of luck if you wanted to override the
default wizard.
With this change, the identify section will properly have a sorcery.conf-based
wizard applied when the identify definition is within the res_pjsip_endpoint_identifier_ip
section.
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Merged revisions 410933 from http://svn.asterisk.org/svn/asterisk/branches/12
Joshua Colp [Wed, 19 Mar 2014 12:54:25 +0000 (12:54 +0000)]
res_stasis: Extend bridge type to be a comma separated list of bridge attributes.
This change turns the bridge type field into a comma separated list of attributes.
These attributes include: mixing, holding, dtmf_events, and proxy_media. By setting
the various attributes a user can control the type of bridge created with the
behavior they need for their application.
(closes issue ASTERISK-23437)
Reported by: Matt Jordan
ARI: allow json content type with zero length body
When a request was received with a Content-type of json,
the body was sent for json parsing - even if it was zero
length. This resulted in ARI requests failing that were
valid, such as a channel DELETE with no parameters. The
code has now been changed to skip json parsing with zero
content length.
(closes issue SWP-6748)
Reported by: Samuel Galarneau
Review: https://reviewboard.asterisk.org/r/3360/
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Matthew Jordan [Tue, 18 Mar 2014 15:28:45 +0000 (15:28 +0000)]
cdr: Add asserts for when we don't know about a CDR for a channel
In the CDR core, every channel should either be filtered out (due to being an
'internal' channel used as an implementation detail, such as playing media
back into a bridge) or it should get a CDR. Even if that CDR ends up being
discarded, we still give the channel a CDR in case we end up needing it. If we
hit a situation where a channel does not have a CDR, we should blow up in
-dev-mode. Asserts are appropriate for that.
This patch adds those asserts, as they would have quickly caught the error
fixed by r410814.
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g711_free() was introduced in spandsp 0.0.6pre4 and g711_release() became a
noop. I opted not to remove the call to g711_release() since it is harmless
and to call g711_free() if we have a sufficiently recent version of spandsp.
(issue ASTERISK-20149)
Reported by: Alexandr Gordeev
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Joshua Colp [Mon, 17 Mar 2014 22:54:32 +0000 (22:54 +0000)]
res_pjsip: Enable PJSIP DNS client support.
This change enables DNS client support within PJSIP. System
nameservers are automatically discovered using res_init or
res_ninit. If this fails then PJSIP will resort to using
gethostbyname for resolution.
By enabling this support we gain SRV support, failover, and
weight support.
(closes issue ASTERISK-23435)
Reported by: Matt Jordan
Joshua Colp [Mon, 17 Mar 2014 22:46:56 +0000 (22:46 +0000)]
res_pjsip_multihomed: Make address replacement less aggressive.
This change makes the res_pjsip_multihomed module less aggressive when
changing the address in messages. It will now only occur if the transport
in use is bound to the any address OR if the system determined source
address matches the bound address of the transport in use.
Russ Meyerriecks [Mon, 17 Mar 2014 22:24:03 +0000 (22:24 +0000)]
callerid: Logic error in checksum processing
Callerid checksum-ing was being handled incorrectly here. When the checksum is
calculated to be 0x00, it will perform 0x100-0x00 which results in 0x100. This
value will then fail the otherwise correct callerid message.
This patch changes the logic to simply add the calculated checksum to the
transmitted 2's compliment checksum.
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This is a merge of merged revisions 410750 410747 from http://svn.asterisk.org/svn/asterisk/branches/12
I didn't want a broken patch to be comitted to trunk so I pre-merge merged them.
Mark Michelson [Mon, 17 Mar 2014 19:35:17 +0000 (19:35 +0000)]
Revert changes to sorcery that accidentally got committed.
These changes were still up for review and have not been approved
yet. I must have had the changes in my working copy when making
a different change.
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Mark Michelson [Mon, 17 Mar 2014 17:22:12 +0000 (17:22 +0000)]
Fix stuck channel in ARI through the introduction of synchronous bridge actions.
Playing back a file to a channel in an ARI bridge would attempt to wait until
the playback concluded before returning. The method used involved signaling the
waiting thread in the ARI custom playback function.
The problem with this is that there were some corner cases that were not accounted for:
* If a bridge channel could not be found, then we never would attempt the playback but
would still attempt to wait for the playback to complete.
* If the bridge playfile action failed to queue, we would still attempt to wait for the
playback to complete.
* If the bridge playfile action were queued but some circumstance caused the playback
not to occur (the bridge dies, the channel is removed from the bridge), then we would
never be notified.
The solution to this is to move the waiting logic into the bridge code. A new bridge
API function is added to queue a synchronous action on a bridge. The waiting thread
is notified when the queued frame has been freed, either due to an error occurring
or due to successful playback. As a failsafe, the waiting thread has a 10 minute
timeout just in case there is a frame leak somewhere.
Matthew Jordan [Sun, 16 Mar 2014 20:27:28 +0000 (20:27 +0000)]
stasis/app.c: Add some extra debugging for subscription counts
Events are sent to a connected ARI application based on the things that ARI
application cares about. These subscriptions can be set up implicitly - such
as when that ARI application creates a new object - or explicitly, via the
application resource's subscription operations. Debugging *why* something was
being sent to an application - or why something was not being sent to an
application - was a bit tricky, as there was no debug information for the
subscriptions.
This patch adds some debug level 3 statements that show the subscription counts
for applications. (Level 3 was chosen as it matches the verbose level 3
statements elsewhere)
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