| Age | Commit message (Collapse) | Author |
|---|
|
|
|
|
|
into 13
|
|
together" into 13
|
|
and apps" into 13
|
|
|
|
into 13
|
|
|
|
* Tweak sorcery_object_type_alloc() formatting.
* Tweak ast_sorcery_init() formatting.
Change-Id: Ib02430023f15268cd7a2ea53f2c331213e4d3944
|
|
Do not check registrar of the first extension head. We should only check
the registrar when we match the priority.
Additionally fix a couple calls to strcmp which used the input callerid
instead of the clean version ex.cidmatch.
ASTERISK-26233
Change-Id: I17ea6881a18f40840ae9c1f5394aab1fbb3769f1
|
|
The Exchanging Device and Mailbox States could not working
if the Entity ID (EID) is not set manually and can't be obtained
from ethernet interface.
This patch replaces debug message to warning
and addes missing description about option 'entityid' to
asterisk.conf.sample.
With this patch the asterisk also:
(1) decline loading the modules which won't work without EID:
res_corosync and res_pjsip_publish_asterisk.
(2) warn if EID is empty on loading next modules:
pbx_dundi, res_xmpp
Starting with v197 systemd/udev will automatically assign "predictable"
names for all local Ethernet interfaces.
This patch also addes some new ethernet prefixes "eno" and "ens".
ASTERISK-26164 #close
Change-Id: I72d712f1ad5b6f64571bb179c5cb12461e7c58c6
|
|
Change-Id: Idbac539205aa732bf786c4f765577d8e9ff28ba4
|
|
Change-Id: I67e6b79fa3102e494b5fe6cc7510472249080e85
|
|
Change-Id: Iea89a36222712148c1775c05ed0ad1049d67a70e
|
|
Change-Id: I80f8a981f62f50e74609c69c49edcaca6c95efa4
|
|
This patch adds some see-also references between related AMI events. It
focuses primarily on those events that are guaranteed to come in pairs,
such as DTMFBegin/DTMFEnd, as well as those that occur during the life
cycle of an Asterisk channel, such as Newchannel/Hangup.
Change-Id: Iaab600477052018d0f8c03d0c624c0856e9ff1f3
|
|
|
|
|
|
Errors during startup result in an exit. These error branches should be
calling ast_run_atexit(0) to ensure mandatory cleanup is run.
ASTERISK-26267 #close
Change-Id: If226f2326ae2df7add20040696132214cf2bb680
|
|
* The high water check in ast_taskprocessor_alert_set_levels() would
trigger immediately if the new high water level is zero and the queue was
empty.
* The high water check in taskprocessor_push() was off by one.
Change-Id: I687729fb4efa6a0ba38ec9c1c133c4d407bc3d5d
|
|
The named aor lock was always being locked for writes so a rwlock adds no
benefit and may be slower because rwlocks are biased toward read locking.
Change-Id: I8c5c2c780eb30ce5441832257beeb3506fd12b28
|
|
If both channels which should be masqueraded
are in the same serializer:
1st channel will be locked waiting condition 'complete'
2nd channel will be locked waiting condition 'suspended'
On heavy load system a chance that both channels will be in
the same serializer 'pjsip/distibutor' is very high.
To reproduce compile res_pjsip/pjsip_distributor.c with
DISTRIBUTOR_POOL_SIZE=1
Steps to reproduce:
1. Party A calls Party B (bridged call 'AB')
2. Party B places Party A on hold
3. Party B calls Voicemail app (non-bridged call 'BV')
4. Party B attended transfers Party A to voicemail using REFER.
5. When asterisk masquerades calls 'AB' and 'BV',
a deadlock is happened.
This patch adds a suspension indicator to the taskprocessor.
When a session suspends/unsuspends the serializer
it sets the indicator to the appropriate state.
The session checks the suspension indicator before
suspend the serializer.
ASTERISK-26145 #close
Change-Id: Iaaebee60013a58c942ba47b1b4930a63e686663b
|
|
This ensures startup is canceled due to allocation failures from the
following initializations.
* channel.c: ast_channels_init
* config_options.c: aco_init
ASTERISK-26265 #close
Change-Id: I911ed08fa2a3be35de55903e0225957bcdbe9611
|
|
|
|
Upcoming features will require the generation and persistence
of a UUID.
Change-Id: I3ec0062427e133217db6ef496a4216f427c3b92d
|
|
SILK is a bit of a hog when it comes to using up our limited number of
dynamic payload types in the RTP engine. By freeing up four slots, it
allows for other codecs to potentially take the place.
Now, codec_silk.so will dynamically use the payload slots in the RTP
engine when it loads.
A better fix would be make RTP dynamic payload types actually
dynamic. However, at this stage of Asterisk 14 development, this is a
risky move that would be imprudent.
Change-Id: I5774e09408f9a203db189529eabdc0d3f4c1e612
(cherry picked from commit d50895c7b04036aeaad58990089399e46db4c817)
|
|
|
|
|
|
|
|
|
|
|
|
This adds a two strings to ast_exten. name to go with exten and
cidmatch_display to go with cidmatch. The new fields contain input used
to add the extension in the first place. The existing fields now
contain stripped input that excludes insignificant spaces and dashes.
These stripped fields should always be used for comparisons. The
unstripped fields should normally be used for display, but displaying
stripped values will not cause runtime errors.
Note the actual string is only stored twice if it contains dashes. If
no dashes are found then both 'char *' fields point to the same memory.
So this change has a minimum effect on memory usage.
The existing functions ast_get_extension_name and
ast_get_extension_cidmatch return unstripped values as they did before
this change. Other similar bugs likely still exist where unstripped
extensions are saved outside pbx.c then passed back in.
ASTERISK-26233 #close
Change-Id: I6cd61ce57acc1570ca6cc14960c4c3b0a9eb837f
|
|
We can allow dangerous functions when adding a hint since altering
dialplan is itself a privileged activity. Otherwise, we could never
execute dangerous functions.
ASTERISK-25996 #close
Reported by: Andrew Nagy
Change-Id: I4929ff100ad1200a0198262d069a34f2296e77ba
|
|
Fixed an assert check that would trigger when the passed in value was negative.
The negative value was being cast to an unsigned value. This resulted in the
check failing.
Also fixed another problem when loading formats in the engine. When setting the
mime type the format's name was being passed in instead of the codec's name.
Change-Id: I1a201cd419ba4d8e9a40d337e36b6fbe1737192c
|
|
* Add fax amplitude and frequency sweep tests.
* Add DTMF amplitude and twist unit tests.
Change-Id: I8d77c9a1eec89e440d715f998c928687e870c3f7
|
|
* Added doxygen to describe some struct members and what is going on in
the code.
Change-Id: I2ec706a33b52aee42b16dcc356c2bd916a45190d
|
|
Change-Id: Ia131da3ec29acf385cb43a586a29ecc975eb3896
|
|
The Goertzel calculations get less accurate the lower the signal level
being worked with becomes because there is less resolution remaining.
If it is too low we can erroneously detect a tone where none really
exists. The searched for fax frequencies not only need to be so much
stronger than the background noise they must also be a minimum strength.
* Add needed minimum threshold test to tone_detect().
* Set TONE_THRESHOLD to allow low volume frequency spread detection.
ASTERISK-26237 #close
Reported by: Richard Mudgett
Change-Id: I84dbba7f7628fa13720add6a88eae3b129e066fc
|
|
|
|
This adds support for tagging functions with the noreturn attribute.
If DO_CRASH is enabled then ast_do_crash never returns. If AST_DEVMODE
and DO_CRASH are enabled then failed assertions never return. This can
resolve a large number of false positives with static analyzers.
ASTERISK-26220 #close
Change-Id: Icfb61e5fe54574eced4c3e88b317244f467ec753
|
|
The new endpoint option allows the PJSIP channel driver's fax_detect
endpoint option to timeout on a call after the specified number of
seconds into a call. The new feature is disabled if the timeout is set
to zero. The option is disabled by default.
ASTERISK-26214
Reported by: Richard Mudgett
Change-Id: Id5a87375fb2c4f9dc1d4b44c78ec8735ba65453d
|
|
adsi.h is no longer used by features.c since parking was moved to a
module.
Change-Id: I2248b8a455225a17cb6ddaafd6c20c511a1eaf59
|
|
|
|
This commit adds scaffolding in order to support the SILK audio format
on calls. Roughly, this is what is added:
* Cached silk formats. One for each possible sample rate.
* ast_codec structures for each possible sample rate.
* RTP payload mappings for "SILK".
In addition, this change overhauls the res_format_attr_silk file in the
following ways:
* The "samplerate" attribute is scrapped. That's native to the format.
* There are far more checks to ensure that attributes have been
allocated before attempting to reference them.
* We do not SDP fmtp lines for attributes set to 0.
These changes make way to be able to install a codec_silk module and
have it actually work. It also should allow for passthrough silk calls
in Asterisk.
Change-Id: Ieeb39c95a9fecc9246bcfd3c45a6c9b51c59380e
|
|
|
|
into 13
|
|
|
|
|
|
Create include_free to run ast_destroy_timing and ast_free, use that in
all places that freed an ast_include structure. This fixes a couple of
paths that previously did not run ast_destroy_timing.
ASTERISK-26196 #close
Change-Id: I1671bd111bef0dc113e8bf8f77f89fcfc395d838
|
|
into 13
|
