Mark Michelson [Thu, 3 Nov 2016 21:42:40 +0000 (16:42 -0500)]
AGI: Only defer frames when in an interception routine.
AGI recently was modified to defer important frames. This was because
when AGI was used in a connected line interception routine, the
resulting connected line frame would end up getting discarded by the
AGI.
However, this caused bad behavior in other cases. Specifically, during a
transfer, if someone attempted to manually set the Caller ID on a
channel in an AGI, the deferred connected line frame would end up
overwriting what had been manually set in the AGI.
Since the initial issue was specific to interception routines, this
change removes the manual frame deferral from AGI and instead uses the
new frame deferral API in interception routines.
Michael Kuron [Sun, 6 Nov 2016 09:46:30 +0000 (10:46 +0100)]
automon: restore mixing of the both channels after recording stops
This is a regression over Asterisk 11, introduced by 2dc8a060064f359a17f5ebcd515d85fe5203c019. Previously, recordings started via
the automon DTMF code would automatically be mixed together using sox because
app_monitor would be called with the m option. This commit restores this
behavior.
Matt Jordan [Fri, 4 Nov 2016 20:42:09 +0000 (15:42 -0500)]
res_http_websocket: Increase the buffer size for non-LOW_MEMORY systems
Not surprisingly, using Respoke (and possibly other systems) it is
possible to blow past the 16k limit for a WebSocket packet size. This
patch bumps it up to 32k, which, at least for Respoke, is sufficient.
For now.
Because 32k is laughable on a LOW_MEMORY system (as is 16k, for that
matter), this patch adds a LOW_MEMORY directive that sets the buffer to
8k for systems who have asked for their reduced memory availability to
be considered.
Matt Jordan [Fri, 4 Nov 2016 20:40:58 +0000 (15:40 -0500)]
res_stasis: Set a video source mode on Stasis created bridges
When a bridge is created via ARI (through res_stasis), no video source
mode is set by default. As a result, any endpoint sending video media
won't ever see any video reflected back to it.
This patch defaults a bridge to a 'follow the talker' video mode.
Further work can be done to add routes that allow for the video mode to
be controlled through the /bridges resource.
Matt Jordan [Fri, 4 Nov 2016 20:37:57 +0000 (15:37 -0500)]
main/bridge_channel: Fix channel reference leak on video source
When a channel is made the video source, the bridge holds a reference to
it. Whenever the video source changes, that reference is released.
However, a ref leak does occur if the channel leaves the bridge (such as
being hung up) while it is the video source, as the bridge never
releases the ref in such a case.
This patch adds a line to the bridge_channel_internal_join routine such
that, when a channel finishes its time in the bridge, it notifies the
bridge via ast_bridge_remove_video_src that if it is a video source its
reference should be released.
Matt Jordan [Fri, 4 Nov 2016 20:36:42 +0000 (15:36 -0500)]
main/bridge: Add some verbose logging for video source changes
It's actually quite useful to see the source of a video stream change.
This doesn't happen terribly often, even with talk detection - but when
it does, it's nice to know which channel is now providing your video
stream.
As a verbose 5 level message, it shouldn't be terribly spammy or costly
to have, and is 'lower level' then most other verbose messages that the
bridge system emits.
Matt Jordan [Fri, 4 Nov 2016 20:33:35 +0000 (15:33 -0500)]
bridges/bridge_softmix: Remove SSRC changes on join/leave; update video source
WebRTC clients really, really want to know the SSRC of the media they're
getting. Changing the SSRC is generally not a good thing.
bridge_softmix, starting in Asterisk 12, started changing the SSRC of
parties as they joined or left the bridge. With most phones, this isn't
a problem: phones just play back the stream they're getting. With WebRTC
clients, however, the SSRC is tied to a media stream that may be
negotiated. When a new SSRC just shows up, the media can be dropped.
As it turns out, the SSRC change shouldn't even be necessary. From the
perspective of the client, it's still talking to Asterisk with the same
media stream: why indicate that the far party has suddenly changed to a
different source of media?
This patch opts to just remove the SSRC changes. With this patch, video
clients that join/leave a softmix bridge actually get the video stream
instead of freaking out.
Kevin Harwell [Fri, 28 Oct 2016 20:11:35 +0000 (15:11 -0500)]
stasis_recording/stored: remove calls to deprecated readdir_r function.
The readdir_r function has been deprecated and should no longer be used. This
patch removes the readdir_r dependency (replaced it with readdir) and also moves
the directory search code to a more centralized spot (file.c)
Also removed a strict dependency on the dirent structure's d_type field as it
is not portable. The code now checks to see if the value is available. If so,
it tries to use it, but defaults back to using the stats function if necessary.
Lastly, for most implementations of readdir it *should* be thread-safe to make
concurrent calls to it as long as different directory streams are specified.
glibc falls into this category. However, since it is possible that there exist
some implementations that are not safe, locking has been added for those other
than glibc.
Joshua Colp [Wed, 2 Nov 2016 15:52:13 +0000 (15:52 +0000)]
res_stasis: Don't unsubscribe from a NULL bridge.
A NULL bridge has special meaning in res_stasis for
unsubscribing. It means that a subscription to ALL
bridges should be removed. This should not be done
as part of the normal subscription management in
the res_stasis channel loop.
George Joseph [Thu, 3 Nov 2016 16:55:06 +0000 (10:55 -0600)]
pjproject_bundled: Fix issue with libasteriskpj needing libresample
libresample is only needed by pjproject if we're building pjsua, which
we only do if TEST_FRAMEWORK is selected. It's required by pjsua to
process audio which is needed by some testsuite tests. Unfortunately,
pjproject relies on a newer version of libresample than the version
that ships by most distros so we need to compile the version that's
bundled with pjproject. Since we only need it for pjsua, we DON'T want
it's symbols exposed when we actually build asterisk.
There was a problem however... TEST_FRAMEWORK is only known AFTER we've
already run ./configure on both asterisk and pjproject but pjproject's
./configure needs to test it to know whether to set up to build
libresample or not. The previous way of figuring this out was to
always tell ./configure "yes" but not actually build the library. This
caused an issue where building libasteriskpj was being told to include
libresample but it wasn't actually there.
The solution is to still do a default pjproject configure during an
asterisk ./configure but if makeopts or menuselect.makeopts changes
subsequently, we now reconfigure pjproject, taking into account the
current state of TEST_FRAMEWORK. Previously, if makeopts or
menuselect.makeopts changed, only a recompile of pjproject was done.
Alexander Traud [Tue, 13 Sep 2016 09:08:34 +0000 (11:08 +0200)]
rtp_engine: Allow more than 32 dynamic payload types.
The dynamic range (96-127) allows 32 RTP Payload Types. RFC 3551 section 3
allows to reassign other ranges. Consequently, when the dynamic range is
exhausted, you can go for "rtp_pt_dynamic = 35" (or 0) in asterisk.conf. This
enables the range 35-63 (or 0-63) giving room for another 29 (or 64) payload
types.
Joshua Colp [Wed, 2 Nov 2016 14:15:14 +0000 (14:15 +0000)]
app_dial: Fix incorrect device state when channel is picked up.
Given the scenario where multiple channels are dialed using Dial()
but the caller is picked up using PickupChan() all outgoing channels
except the channel specified to PickupChan() would be marked
as ringing until the call had been hung up.
When using the PickupChan application the channel executing the
application is swapped into place of another channel. As part
of this process the channel is answered. The Dial application
has explicit logic which checks if the channel is answered,
cancels all other outgoing channels, and bridges. This logic is
different than the normal logic that is executed when an outgoing
channel is answered. This different logic failed to publish dial
events stating that the other outgoing channels had been canceled.
As a result references to the outgoing channels were held onto by
the dial masquerade process until the call had been ended and
the channels had gone away. This would result in the channels
appearing in the "core show channels" list despite not being present
anymore and would also result in incorrect device state.
This change makes it so that this logic also publishes
dial events stating that the other outgoing channels have been
canceled.
Joshua Colp [Tue, 1 Nov 2016 11:56:24 +0000 (11:56 +0000)]
res_pjsip_sdp_rtp: Limit number of formats to defined maximum.
The res_pjsip_sdp_rtp module did not restrict the number of
formats added to a media stream in the SDP to the defined
limit. If allow=all was used with additional loaded codecs this
could result in the next media stream being overwritten some.
This change restricts the module to limit it to the defined
maximum and also increases the maximum in our bundled pjproject.
Tzafrir Cohen [Tue, 1 Nov 2016 09:00:21 +0000 (11:00 +0200)]
define PATH_MAX for HURD
PATH_MAX is not guaranteed to be defined. In parctice, all but the HURD
define it to a constant. It is indeed not safe to assume there won't be
longer paths and Asterisk generally does err safely on such cases.
So even for HURD we'll just pretend PATH_MAX is 4096.
Kevin Harwell [Mon, 31 Oct 2016 22:35:47 +0000 (17:35 -0500)]
codecs.conf.sample: Add sample and option descriptions for codec_opus
codecs.conf.sample was missing codec opus's configuration options, descriptions,
and examples. This patch adds the configuration options and examples to
codecs.conf.sample that can be used with codec_opus.
If in sip.conf (general section) set option register_retry_403=no,
the command "sip show settings" return value:
Outbound reg. retry 403:0
If in sip.conf (general section) set option register_retry_403=yes,
the command "sip show settings" return value:
Outbound reg. retry 403:-1
* In static char "sip show settings" for "Outbound.reg. retry 403"
option use AST_CLI_YESNO
Matt Jordan [Thu, 20 Oct 2016 12:27:21 +0000 (07:27 -0500)]
res/stasis: Add CLI commands for displaying/debugging ARI apps
This patch adds three new CLI commands:
- ari show apps: list the registered ARI applications
- ari show app: show detailed information about an ARI application
- ari set debug: dump events being sent to an ARI application
Note that while these CLI commands live in the res_stasis module, we use
the 'ari' family for these commands. This was done as most users of
Asterisk aren't aware of the semantic differences between ARI and
res_stasis, and some 'ari' CLI commands already exist.
George Joseph [Mon, 31 Oct 2016 21:12:57 +0000 (15:12 -0600)]
pjproject_bundled: Fix compile of pjsua so it handles audio
In order for pjsua and its python binding to actually negotiate
audio for the testsuite tests, it needs g711 and resample. The
pj* libraries themselves do not. Unfortunately, pjproject relies
on a brand new libresample that most distros don't ship so we need
to use the libresample already bundled with pjproject. Only the pjsua
executable and the _pjsua.so python library are linked with it so it
shouldn't interfere with asterisk itself.
Also it was pointed out that apply_patches couldn't handle multiple
patches that depended on each other during the dry-run, so the
dry-run was removed.
Corey Farrell [Sun, 30 Oct 2016 18:33:12 +0000 (14:33 -0400)]
vector: Prevent NULL argument to memcpy.
Headers declare that memcpy does not accept NULL argument for the first
two parameters. Add a conditional block to prevent memcpy and ast_free
from running on vectors with NULL element array.
Corey Farrell [Sat, 29 Oct 2016 15:31:15 +0000 (11:31 -0400)]
astobj2: Declare private variable data_size for AO2_DEBUG only.
Every ao2 object contains storage for a private variable data_size,
though the value is never read if AO2_DEBUG is disabled. This change
makes the variable conditional, reducing memory usage.
George Joseph [Fri, 28 Oct 2016 21:59:19 +0000 (15:59 -0600)]
pjproject_bundled: Fix issue where "/version.mak" wasn't found
main/Makefile includes third-party/pjproject/build.mak but
doesn't set PJDIR beforehand so "include $(PJDIR)/version.mak"
evaluates to "/version.mak". Fix is to set PJDIR in main/Makefile
before the include.
The patch below removes a cached DNS response from the hash table when
another thread is referencing the old entry. The table still contained
the entry when it was destroyed which can result in inexplicable crashes.
Corey Farrell [Fri, 28 Oct 2016 02:49:43 +0000 (22:49 -0400)]
Fix shutdown crash caused by modules being left open.
It is only safe to run ast_register_cleanup callbacks when all modules
have been unloaded. Previously these callbacks were run during graceful
shutdown, making it possible to crash during shutdown.
George Joseph [Wed, 26 Oct 2016 23:48:24 +0000 (17:48 -0600)]
pjproject_bundled: Remove usage of tar's --strip-components option
Older versions of tar don't support the --strip-components option so
instead of doing 'tar --strip-components=1 -C source', we now just
untar to the tarball's root directory (pjproject-<version>) and
rename that directory to 'source'.
Also fixed an issue where the pjproject source directory is a hard
coded absolute pathname.
Joshua Colp [Thu, 27 Oct 2016 13:07:02 +0000 (13:07 +0000)]
res_pjsip_caller_id: Fix crash on session timers UPDATE on inbound calls.
The res_pjsip_caller_id module wrongly assumed that a
saved From header would always exist on sessions. This
is true until an inbound call is received and a session
timer causes an UPDATE to be sent. In this case there will
be no saved From header and a crash will occur. This change
makes it fall back to the From header of the outgoing request
if no saved From header is present.
Joshua Colp [Wed, 26 Oct 2016 12:51:50 +0000 (12:51 +0000)]
app_voicemail: Clear voice mailbox in MailboxExists and MAILBOX_EXISTS.
When executing the MailboxExists dialplan application and
MAILBOX_EXISTS dialplan function the passed in temporary voice
mailbox was not cleared, causing it to try to free garbage.
Joshua Colp [Sun, 23 Oct 2016 12:38:59 +0000 (12:38 +0000)]
pjsip: Fix a few media bugs with reinvites and asymmetric payloads.
When channel format changes occurred as a result of an RTP
re-negotiation the bridge was not informed this had happened.
As a result the bridge technology was not re-evaluated and the
channel may have been in a bridge technology that was incompatible
with its formats. The bridge is now unbridged and the technology
re-evaluated when this occurs.
The chan_pjsip module also allowed asymmetric codecs for sending
and receiving. This did not work with all devices and caused one
way audio problems. The default has been changed to NOT do this
but to match the sending codec to the receiving codec. For users
who want asymmetric codecs an option has been added, asymmetric_rtp_codec,
which will return chan_pjsip to the previous behavior.
The codecs returned by the chan_pjsip module when queried by
the bridge_native_rtp module were also not reflective of the
actual negotiated codecs. The nativeformats are now returned as
they reflect the actual negotiated codecs.
George Joseph [Tue, 25 Oct 2016 16:20:16 +0000 (10:20 -0600)]
test_astobj2_thrash: Fix multithreaded issues
The test uses 4 threads to grow, count, lookup and shrink 15K objects
in a container. If there's only 1 execution engine available, the test
will complete in <50ms. If each threads gets its own execution engine,
the test may timeout after 60 seconds because the count thread does a
locked ao2_callback on the whole container in a tight loop with only
a sched_yield to give up time. The lock contention makes the test
execution times wildly variable and mostly timeout. 2 execution
engines are OK, 3 results in about 33% failure rate and >=4 causes
a 80% failure rate.
To fix, the sched_yield was changed to a usleep(500).
Also, the number of buckets specified for the container was an even
number so that was changed to the next prime number greater than
(MAX_HASH_ENTRIES / 100). That's 151 currently.
George Joseph [Mon, 24 Oct 2016 15:55:23 +0000 (09:55 -0600)]
pjproject_bundled: Fixed various build issues
* CFLAGS is now properly set when using older gcc.
* All third-party pjproject targets have been removed. This fixes
an issue with older libsrtp in some distros.
* Manually removing the source directory now causes a rebuild.
* EXTERNALS_CACHE_DIR is now properly checked.
* Whitespace fixes.
This change adds support for dual stack automatically. No
configuration is required and the IP address and version
in the SIP messages and SDP will be automatically changed
based on the transport over which the message is being
sent. RTP usage has also been changed to listen on both
IPv4 and IPv6 simultaneously to allow media to flow, and
to allow ICE support on both simultaneously. This also
allows failover between IPv6 and IPv4 to work as expected.
Mark Michelson [Mon, 17 Oct 2016 19:18:57 +0000 (14:18 -0500)]
ARI: Detect duplicate channel IDs
ARI and AMI allow for an explicit channel ID to be specified
when originating channels. Unfortunately, there is nothing in
place to prevent someone from using the same ID for multiple
channels. Further complicating things, adding ID validation to channel
allocation makes it impossible for ARI to discern why channel allocation
failed, resulting in a vague error code being returned.
The fix for this is to institute a new method for channel errors to be
discerned. The method mirrors errno, in that when an error occurs, the
caller can consult the channel errno value to determine what the error
was. This initial iteration of the feature only introduces "unknown" and
"channel ID exists" errors. However, it's possible to add more errors as
needed.
ARI uses this feature to determine why channel allocation failed and can
return a 409 error during origination to show that a channel with the
given ID already exists.
Michael Walton [Tue, 4 Oct 2016 23:24:54 +0000 (12:24 +1300)]
res_rtp_asterisk: Add ice_blacklist option
Introduces ice_blacklist configuration in rtp.conf. Subnets listed in the
form ice_blacklist = <subnet spec>, e.g. ice_blacklist =
192.168.1.0/255.255.255.0, are excluded from ICE host, srflx and relay
discovery. This is useful for optimizing the ICE process where a system
has multiple host address ranges and/or physical interfaces and certain
of them are not expected to be used for RTP. Multiple ice_blacklist
configuration lines may be used. If left unconfigured, all discovered
host addresses are used, as per previous behavior.
Mark Michelson [Tue, 18 Oct 2016 21:30:17 +0000 (16:30 -0500)]
CDR: Alter destruction pattern for CDR chains.
CDRs form chains. When the root of the chain is destroyed, it then
unreferences the next CDR in the chain. That CDR is destroyed, and it
then unreferences the next CDR in the chain. This repeats until the end
of the chain is reached. While this typically does not cause any sort of
problems, it is possible in strange scenarios for the CDR chain to grow
way longer than expected. In such a scenario, the destruction pattern
can result in a stack overflow.
This patch fixes the problem by switching from a recursive pattern to an
iterative pattern for destruction. When the root CDR is destroyed, it is
responsible for iterating over the rest of the CDRs and unreferencing
each one. Other CDRs in the chain, since they are not the root, will
simply destroy themselves and be done. This causes the stack depth not
to increase.
Alexander Traud [Tue, 18 Oct 2016 08:01:47 +0000 (10:01 +0200)]
cli: Auto-complete File not Module for core set debug.
Since Asterisk 1.8, the command "core set debug" on the command-line interface
asks not for a file (.c) but a module name. This change shows modules (.so) on
the auto-completion via a tabulator or the question mark. Now, when you
partially type a module name, TAB or ?, you get the correct candidiates.
Matt Jordan [Sun, 16 Oct 2016 01:05:05 +0000 (20:05 -0500)]
res/ari: Add the Asterisk EID field to outgoing events
This patch adds the Asterisk EID field to all outgoing ARI events.
Because this field should be added to all events as they are
transmitted, it is appended to the JSON message just prior to it being
handed off to the application message handler. This makes it somewhat
resilient to both new events being added to ARI, as well as other
potential event transport mechanisms.
George Joseph [Sun, 16 Oct 2016 22:25:35 +0000 (16:25 -0600)]
utils.c: Fix ast_set_default_eid for multiple platforms
ast_set_default_eid was searching for ethX, emX, enoX, ensX and even
pciD#U interface names. While this was a good attempt, it wasn't
inclusive enough to capture interfaces like enp6s0 or ens6d1, etc.
Rather than relying on interface names, we now simply find the first
interface returned by the OS that has a hardware address and that
address isn't all 0x00 or all 0xff. The code IS different for BSD,
Solaris and Linux based on what method is available for enumerating
interfaces.
Tested on:
FreeBSD9
CentOS6
Ubuntu14
Fedora24
I was unable to test on Solaris at this time but the code for Solaris
is used elsewhere at Digium.
projects / thirdparty / asterisk.git / log
