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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.
(closes issue ASTERISK-23509)
Reported by: zvision
Review: https://reviewboard.asterisk.org/r/3378/
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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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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.
Review: https://reviewboard.asterisk.org/r/3391/
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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.
(closes issue ASTERISK-23254)
Reported by: rmudgett
Review: https://reviewboard.asterisk.org/r/3381/
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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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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.
Review: https://reviewboard.asterisk.org/r/3380/
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Add some temporary sanity checks to hunt for locking problems with the
masquerade supertest.
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to restricted.
(closes issue ASTERISK-20841)
Reported by: Kelly Goedert
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* Fix variable shadowing of 'updated' by renaming it to 'contact_update'.
* Checked 'contact_update' for ast_sorcery_copy() failure.
* Removed silly use of RAII_VAR() for 'contact_update'.
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Switched a bunch of LOG_NOTICEs to ast_debug. This time without breaking the
build.
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There is enough indication from the CLI that we are loading a realtime engine
as it is.
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Switched a bunch of LOG_NOTICEs to ast_debug.
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(closes issue ASTERISK-23461)
Reported by: Chico Manobela
Review: https://reviewboard.asterisk.org/r/3373/
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* 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().
Review: https://reviewboard.asterisk.org/r/3371/
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"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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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
Review: https://reviewboard.asterisk.org/r/3359/
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for the res_ari general section
Added note and see-also reminding user to enable the HTTP server.
(closes issue ASTERISK-22499)
Reported by: Rusty Newton
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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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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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failure.
Thanks Walter Doekes!
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Per Johann Steinwendtner on the asterisk-dev mailing list:
http://lists.digium.com/pipermail/asterisk-dev/2014-March/066102.html
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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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
Review: https://reviewboard.asterisk.org/r/3343/
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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.
Review: https://reviewboard.asterisk.org/r/3369/
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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.
Review: https://reviewboard.asterisk.org/r/3356/
(closes issue ASTERISK-23488)
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I didn't want a broken patch to be comitted to trunk so I pre-merge merged them.
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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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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.
Review: https://reviewboard.asterisk.org/r/3338
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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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There were a number of instances in this header file where "function all" was
intended to be "function call". This patch fixes that up.
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The store realtime callback needs to return a positive value for
sorcery to treat the store as a success.
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(closes issue ASTERISK-23420)
Reported by: Etienne Lessard
Patches:
manager_eventfilter_leak uploaded by Etienne Lessard (license 6394)
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Realtime backends' update and store callbacks return the number of rows affected,
or -1 if there was a failure. There were a couple of issues:
* The config API was treating 0 as a successful return, and positive values as
a failure. Now the config API treats anything >= 0 as a success.
* res_sorcery_realtime was treating 0 as a successful return from the store
procedure, and any positive values as a failure. Now sorcery treats anything
> 0 as a success. It still considers 0 a "failure" since there is no change
to report to observers.
Review: https://reviewboard.asterisk.org/r/3341
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If an endpoint is receiving unsolicited MWI for a mailbox and then attempts
to subscribe to an AOR that provides MWI for the same mailbox, then the SUBSCRIBE
is rejected with a 500 response.
Review: https://reviewboard.asterisk.org/r/3345
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Note the new features provided by uniqueid in the
CHANGES file.
(issue ASTERISK-23120)
Review: https://reviewboard.asterisk.org/r/3316/
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(closes issue ASTERISK-23235)
Reported by: George Joseph
Review: https://reviewboard.asterisk.org/r/3324/
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The syncing thread sleeps for a second before waiting to be
told to attempt to sync again. If a signal were sent during this
sleeping period, we would end up having to wait until the next
sync signal occurred in order to sync up the astdb.
This code rearrangement also ensures that any pending transactions
will be synced prior to Asterisk shutting down.
Patches: db_sync.patch by John Hardin (License #6512)
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(closes issue ASTERISK-23444)
Reported by: Ben Merrills
Review: https://reviewboard.asterisk.org/r/3340/
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One of the things missing when external MWI support was added was the
ability to clear the stasis cache entry of deleted external MWI mailboxes.
Review: https://reviewboard.asterisk.org/r/3325/
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* Trivial common code hoisting in handle_bridge_leave_message().
* Some whitespace fixing.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410542 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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This fixes an issue where a Stasis application running over ARI and
subscribed to ari/events could miss the ChannelEnteredBridge event
because it did not subscribe to the new bridge fast enough.
To accomplish this, it subscribes the application controlling the
channel to the new bridge before adding it to that bridge which
required the stasis_app_control structure to maintain a reference to
the stasis_app.
(closes issue ASTERISK-23295)
Review: https://reviewboard.asterisk.org/r/3336/
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Merged revisions 410527 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410528 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r410509 | file | 2014-03-13 06:23:14 -0700 (Thu, 13 Mar 2014) | 2 lines
res_pjsip_multihomed: Fix a bug where the 200 OK for a REGISTER would contain the wrong contact.
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r410510 | file | 2014-03-13 06:24:17 -0700 (Thu, 13 Mar 2014) | 2 lines
res_pjsip_multihomed: Remove change for testing fix.
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Merged revisions 410509-410510 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410511 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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started/stopped.
* Made res_musiconhold.c always post the MusicOnHoldStart/MusicOnHoldStop
events when it actually starts/stops the music streams. This allows the
events to always happen when MOH starts/stops. The event posting code was
moved to the MOH alloc/release routines.
* Made channel_do_masquerade() stop any MOH on the original channel before
masquerading so the original channel will get a stop event with correct
information.
* Cleaned up a couple odd codings in moh_files_alloc() and moh_alloc()
dealing with the music state variable.
(issue ASTERISK-23311)
Reported by: Benjamin Keith Ford
Review: https://reviewboard.asterisk.org/r/3306/
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Merged revisions 410493 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410494 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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