For times when a reference in ARI might be ambiguous, the reference is
built as an URI (such as channel:1376341790.3).
An endpoint's channel list is not ambiguous, and in fact the field is
named 'channel_ids', but it had channel URI's instead of channel id's.
This patch changes the list to be the raw id instead of the URI.
(closes issue ASTERISK-22291)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397297 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In the shuffling around of res_stasis, control_continue was renamed to
stasis_app_control_continue, but the call in res_stasis wasn't updated.
In looking into it, it turns out it wasn't really the right thing to do
in res_stasis anyways.
This patch changes the handling of received a AST_CONTROL_HANGUP frame
to be the same as receiving a NULL frame, and removed the declaration of
control_continue(), since it doesn't exist any more.
(closes issue ASTERISK-22292)
Reported by: Denis Smirnov
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397295 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Added an option flags parameter to interval hooks. Interval hooks now
can specify if the callback will affect the media path or not.
* Added an option flags parameter to the bridge action custom callback.
The action callback now can specify if the callback will affect the media
path or not.
* Made the holding bridge technology reexamine the participant idle mode
option whenever the entertainment is restarted.
* Fixed app_agent_pool waiting agents needlessly starting and stopping MOH
every second by specifying the heartbeat interval hook as not affecting
the media path.
* Fixed app_agent_pool agent alert from restarting the MOH after the alert
beep. The agent entertainment is now changed from MOH to silence after
the alert beep.
* Fixed holding bridge technology to defer starting the entertainment. It
was previously a mixture of immediate and deferred.
* Fixed holding bridge technology to immediately stop the entertainment.
It was previously a mixture of immediate and deferred. If the channel
left the bridging system, any deferred stopping was discarded before
taking effect.
* Miscellaneous holding bridge technology rework coding improvements.
Review: https://reviewboard.asterisk.org/r/2761/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397294 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If a From header on an outbound out-of-call SIP MESSAGE were
malformed, the result could crash Asterisk.
In addition, if a From header on an incoming out-of-call SIP
MESSAGE request were malformed, the message was happily accepted
rather than being rejected up front. The incoming message path
would not result in a crash, but the behavior was bad nonetheless.
(closes issue ASTERISK-22185)
reported by Zhang Lei
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Merged revisions 397254 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397255 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This detects hangups that occur while bridged to allow channels to exit
app_stasis even if the hangup frame was absorbed by the bridge the
channel was in.
Reported by: David Lee
(closes issue ASTERISK-22297)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397244 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The SIP_CODEC family of variables let you set the preferred codec to be
offered on an outbound INVITE request. However, for video calls, you need to
be able to set both the audio and video codecs to be offered. This patch lets
the SIP_CODEC variables accept a comma delineated list of codecs. The first
codec in the list is set as the preferred codec; additional codecs are still
offered however.
This lets a dialplan writer set both audio and video codecs, e.g.,
Set(SIP_CODEC=ulaw,h264)
Note that this feature was written by both Dennis Guse and Frank Haase
Review: https://reviewboard.asterisk.org/r/2728
(closes issue ASTERISK-21976)
Reported by: Denis Guse
Tested by: mjordan, sysreq
patches:
patch-channels-chan__sip.c-393919 uploaded by dennis.guse (license 6513)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397243 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In 1.8, r384779 introduced a regression by retrieving an old dialog and keeping
the old recv address since recv was already set. This has caused a problem when
a proxy is involved since responses to incoming requests from the proxy server,
after an outbound call is established, are never sent to the correct recv
address.
In 11, r382322 introduced this regression.
The fix is to revert that change and always store the recv address on incoming
requests.
Thank you Walter Doekes for helping to point out this error and Mark Michelson
for your input/review of the fix.
(closes issue ASTERISK-22071)
Reported by: Alex Zarubin
Tested by: Alex Zarubin, Karsten Wemheuer
Patches:
asterisk-22071-store-recvd-address.diff by Michael L. Young (license 5026)
........
Merged revisions 397204 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 397205 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397206 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This is more-or-less a reversion of previous ACL behavior so that
it is more self-contained. ACL sections are now only parsed if res_pjsip_acl.so
is loaded. Moreover, the configuration section is now "type=acl" instead of
"type=security".
The original reason for having ACLs configured in a "type=security" section
was to lump ACLs and other security-related items into the same section. The
problem is that ACLs really should be in their own sections and there are
no other security-related options implemented anyways.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397193 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This also removes documentation for the options that no longer exist.
(closes issue ASTERISK-22306)
reported by Rusty Newton
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397111 65c4cc65-6c06-0410-ace0-fbb531ad65f3
........
Fix xmldoc memory leak
This fixes a single-attribute memory leak that was occurring when the
"required" attribute was not true.
(closes issue ASTERISK-22249)
Reported by: Corey Farrell
Tested by: Corey Farrell
Patches:
xmldoc-free_attr_required.patch uploaded by Corey Farrell
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Merged revisions 397064 from http://svn.asterisk.org/svn/asterisk/branches/1.8
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397072 65c4cc65-6c06-0410-ace0-fbb531ad65f3
........
Protect CEL from an invalid config on reload
This patch fixes CEL to properly handle an invalid config on reload.
(closes issue ASTERISK-22259)
Reported by: Corey Farrell
Tested by: Corey Farrell
Patches:
cel-config.patch uploaded by Corey Farrell
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Merged revisions 397033 from http://svn.asterisk.org/svn/asterisk/branches/1.8
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397035 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Queue members who happen to be in multiple queues at the same time may not
have any wrap up time. This problem occurred due to a code change in Asterisk
11.3.0 that unified device state tracking of Queue members in multiple
Queues (which fixed some other problems, but unfortunately caused this one).
This patch fixes the behavior by having the is_member_available function
check the queue's wrap up time and the time of the member's last call, such
that for a particular queue, the member won't be considered available if their
last call is within the wrap up time.
(closes issue ASTERISK-22189)
Reported by: Tony Lewis
Tested by: Tony Lewis
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Merged revisions 396948 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396949 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When r382230 added an option to not denoise the MeetMe conference (if a user
had a channel whose format's sample rate changed frequently, for example),
the value added was the maximum allowed value for the constants that define
the options for MeetMe in 1.8. Not so in 11 - unfortunately, the option
CONFFLAG_DONT_DENOISE conflicts with CONFFLAG_INTROUESR_VMREC. This patch
fixes that, and also tweaks one of the way in which the constants was
declared for consistency.
Thanks to Tony Mountifield for pointing out the problem and solution.
(closes issue ASTERISK-22269)
Reported by: Tony Mountifield
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Merged revisions 396944 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396946 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Performing a blond transfer (attended transfer that is completed
before the transfer recipient picks up) externally through chan_sip
or chan_pjsip would result in lost references to the channels
involved with the transfer as well as their bridge.
(closes issue ASTERISK-22092)
Reported by: mmichelson
Review: https://reviewboard.asterisk.org/r/2766/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396923 65c4cc65-6c06-0410-ace0-fbb531ad65f3
It is not safe to iterate over a macro'd list of ao2 objects, deref them such
that the item's destructor is called, and leave them in the list. The list
macro to iterate over items requires the item to be a valid allocated object
in order to proceed to the next item; with MALLOC_DEBUG on the corruption of
the linked list is caught in the crash.
This patch fixes the invalid access to free'd memory by removing the ao2 item
from the list before de-refing it.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396915 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Added or modified text in the xml doc for the 'aor' config object to address a few issues:
* help for the 'mailboxes' option didn't make it clear how the "list" should be formatted.
* AoR object's involvement in inbound registration wasn't mentioned.
* help for the 'contact' option didn't describe how to specify multiple contacts.
* help for the 'max_contacts' option didn't tell whether it limited the amount of contacts defined through static configuration.
(issue ASTERISK-22118)
(closes issue ASTERISK-22118)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396902 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change protects accesses of res_parking such that it can unload
safely once transient uses of its registered functions are complete.
The parking API has been restructured such that its consumers do not
have access to the vtable exposed by the parking provider, but instead
route through stubs to prevent consumers from holding on to function
pointers.
This adds calls to all the parking unload functions and moves
application loading and unloading into functions in
parking_applications.c similar to the rest of the parts of res_parking.
Review: https://reviewboard.asterisk.org/r/2763/
(closes issue ASTERISK-22142)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396890 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This removes unused code, event types, IE pltypes, and event IE types
where possible and makes several functions private that were once
public. This includes a renumbering of the remaining event and IE types
which breaks binary compatibility with previous versions. The last
remaining consumers of the old event system (or parts thereof) are
main/security_events.c, res/res_security_log.c, tests/test_cel.c,
tests/test_event.c, main/cel.c, and the CEL backends.
Review: https://reviewboard.asterisk.org/r/2703/
(closes issue ASTERISK-22139)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396887 65c4cc65-6c06-0410-ace0-fbb531ad65f3
SIP/foo -- Local;1==Local;2 -- .... -- Local;1==Local;2 -- SIP/bar
Kick a ;1 channel and the chain toward SIP/foo goes away.
Kick a ;2 channel and the chain toward SIP/bar goes away.
This can leave a local channel chain between the kicked ;1 and ;2 channels
that are orphaned until you manually request one of those channels to
hangup or request the bridge to dissolve.
* Added ast_bridge_kick() as a companion to ast_bridge_remove(). The
functional difference is that ast_bridge_kick() may dissolve the bridge as
a result of the channel leaving the bridge.
* Made CLI "bridge kick <bridge> <channel>" use ast_bridge_kick() instead
of ast_bridge_remove() so the bridge can dissolve if needed.
* Renamed bridge_channel_handle_hangup() to ast_bridge_channel_kick() and
made it accessible to other files.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396877 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The horrid structure of the source in the utils directory strikes again.
Moved the _ast_mem_backtrace_buffer[] definition from the logical location
in utils.c to hashtab.c so the aelparse and conf2ael utilities can link.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396850 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Change r395954 reordered some stasis object destruction, which should
have been fine. Unfortunately, it caused some hard to reproduce issues
related to objects being accessed after they had been destroyed. The
patch in r396329 fixed the destruction order problem; this patch
addresses the underlying issue. A few other stasis-related fixes were
also added.
* Add ref-bumps around areas where objects may get transitively
destroyed. (For example, where we lock a topic, unref a subscription,
which unrefs the topic, which explodes the topic when we try to
unlock it.)
* Wrote an extensive doxygen page about Stasis implementation,
relationships between objects, lifecycles of objects, how the
refcounting works, etc. Many other comments were added, corrected, or
cleaned up.
* Added an assert to the topic dtor to catch extra ref decrements.
* Fixed type used after destruction errors for graceful shutdown in
stasis_channels.c.
* I added two unit tests in an attempt to catch destruction order
issues. Since the underlying cause is a race condition, though, the
tests rarely failed even when the code was wrong.
* Fixed a leak in stasis_cache_pattern.c.
(closes issue ASTERISK-22243)
Review: https://reviewboard.asterisk.org/r/2746/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396842 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This reworks the CLI commands used to access sounds information from
"sounds show[ soundid]" to "core show sounds" and
"core show sound <soundid>". This also reworks the "sounds reload" CLI
command to fall under normal module reloading ("module reload sounds").
Also, make trunk build when DEBUG_MALLOC is not enabled.
Review: https://reviewboard.asterisk.org/r/2745/
(closes issue ASTERISK-22141)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396829 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When asterisk has run out of memory (for whatever reason), the alloc
function logs a message. Logging requires memory. A recipe for
infinite recursion.
Stop the recursion by comparing the function call depth for sane values
before attempting another OOM log message.
Review: https://reviewboard.asterisk.org/r/2743/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396822 65c4cc65-6c06-0410-ace0-fbb531ad65f3
By their nature, the connected line and redirecting interception routines
are not supposed to affect the channel's media. Therefore, they should
not suspend and unsuspend the channel while running. The
suspend/unsuspend operations could be expensive depending upon the bridge
and channel technology involved.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396814 65c4cc65-6c06-0410-ace0-fbb531ad65f3
