This patch introduces a new identifier for channels: tenantid. It's
a stringfield on the channel that can be used for general purposes. It
will be inherited by other channels the same way that linkedid is.
You can set tenantid in a few ways. The first is to set it in the
dialplan with the Set and CHANNEL functions:
exten => example,1,Set(CHANNEL(tenantid)=My tenant ID)
It can also be accessed via CHANNEL:
exten => example,2,NoOp(CHANNEL(tenantid))
Another method is to use the new tenantid option for pjsip endpoints in
pjsip.conf:
[my_endpoint]
type=endpoint
tenantid=My tenant ID
This is considered the best approach since you will be able to see the
tenant ID as early as the Newchannel event.
It can also be set using set_var in pjsip.conf on the endpoint like
setting other channel variable:
set_var=CHANNEL(tenantid)=My tenant ID
Note that set_var will not show tenant ID on the Newchannel event,
however.
Tenant ID has also been added to CDR. It's read-only and can be accessed
via CDR(tenantid). You can also get the tenant ID of the last channel
communicated with via CDR(peertenantid).
Tenant ID will also show up in CEL records if it has been set, and the
version number has been bumped accordingly.
Fixes: #740
UserNote: tenantid has been added to channels. It can be read in
dialplan via CHANNEL(tenantid), and it can be set using
Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is recommended to
use the new tenantid option for pjsip endpoints (e.g., tenantid=My
tenant ID) so that it will show up in Newchannel events. You can set it
like any other channel variable using set_var in pjsip.conf as well, but
note that this will NOT show up in Newchannel events. Tenant ID is also
available in CDR and can be accessed with CDR(tenantid). The peer tenant
ID can also be accessed with CDR(peertenantid). CEL includes tenant ID
as well if it has been set.
UpgradeNote: A new versioned struct (ast_channel_initializers) has been
added that gets passed to __ast_channel_alloc_ap. The new function
ast_channel_alloc_with_initializers should be used when creating
channels that require the use of this struct. Currently the only value
in the struct is for tenantid, but now more fields can be added to the
struct as necessary rather than the __ast_channel_alloc_ap function. A
new option (tenantid) has been added to endpoints in pjsip.conf as well.
CEL has had its version bumped to include tenant ID.
(cherry picked from commit 9ee00e0d60)
func_periodic_hook does not hangup after playback, relying on hangup
which keeps the channel alive longer than necessary.
Resolves: #325
(cherry picked from commit cf623753a9)
func_periodic_hook was truncating long channel names which
causes issues when you need to run other dialplan functions/apps
on the channel.
Resolves: #319
(cherry picked from commit 2e59f9fd6a)
Using the Set dialplan application does not actually
delete channel or global variables. Instead the
variables are set to an empty value.
This change adds two dialplan functions,
GLOBAL_DELETE and DELETE which can be used to
delete global and channel variables instead
of just setting them to empty.
There is also no ability within the dialplan to
determine if a global or channel variable has
actually been set or not.
This change also adds two dialplan functions,
GLOBAL_EXISTS and VARIABLE_EXISTS which can be
used to determine if a global or channel variable
has been set or not.
Resolves: #289
UserNote: Four new dialplan functions have been added.
GLOBAL_DELETE and DELETE have been added which allows
the deletion of global and channel variables.
GLOBAL_EXISTS and VARIABLE_EXISTS have been added
which checks whether a global or channel variable has
been set.
(cherry picked from commit 405a08aaed)
Add periodic beep option to one-touch recording by setting
the touch variable TOUCH_MONITOR_BEEP or
TOUCH_MIXMONITOR_BEEP to the desired interval in seconds.
If the interval is less than 5 seconds, a minimum of 5
seconds will be imposed. If the interval is set to an
invalid value, it will default to 15 seconds.
A new test event PERIODIC_HOOK_ENABLED was added to the
func_periodic_hook hook_on function to indicate when
a hook is started. This is so we can test that the touch
variable starts the hook as expected.
ASTERISK-30446
Change-Id: I800e494a789ba7a930bbdcd717e89d86040d6661
Most issues were in stringfields and had to do with comparing
a pointer to an constant/interned string with NULL. Since the
string was a constant, a pointer to it could never be NULL so
the comparison was always "true". gcc now complains about that.
There were also a few issues where determining if there was
enough space for a memcpy or s(n)printf which were fixed
by defining some of the involved variables as "volatile".
There were also a few other miscellaneous fixes.
ASTERISK-30044
Change-Id: Ia081ca1bcfb329df6487c4660aaf1944309eb570
Some databases depending on their configuration using backslashes
for escaping. When combined with the use of ' this can result in
a broken func_odbc query.
This change adds a SQL_ESC_BACKSLASHES dialplan function which can
be used to escape the backslashes.
This is done as a dialplan function instead of being always done
as some databases do not require this, and always doing it would
result in incorrect data being put into the database.
ASTERISK-29838
Change-Id: I152bf34899b96ddb09cca3e767254d8d78f0c83d
* Updated .gitreview default branch to certified/18.9
* Updated .version to certified/18.9
* Set all extended support modules to be disabled by default
* fixed MODULEINFO order introduced by default script on wiki
Change-Id: Id12804325c31b1112b6c559fa97c8a6401c051b3
Most examples in the XML documentation use the
example tag to demonstrate examples, which gets
parsed specially in the Wiki to make it easier
to follow for users.
This fixes a few modules to use the example
tag instead of vanilla para tags to bring them
in line with the standard syntax.
ASTERISK-29777 #close
Change-Id: I9acb6cc5faf1d220e73c6dd28592371d768d279b
There are 3 separate changes here:
1. The documentation erroneously stated that the dsp_talking_threshold
argument was a number of milliseconds when it is actually an energy
level used by the DSP code to classify talking vs. silence.
2. Fixes a copy paste error in the argument handling code.
3. Don't erroneously switch to the talking state if we aren't actively
handling a frame we've classified as talking.
Patch inspired by one provided by Moritz Fain (License #6961).
ASTERISK-27816 #close
Change-Id: I5953fd570b98b49c41cee55bfe3b941753fb2511
Adds a function to check for the existence of a channel by
name or by UNIQUEID.
ASTERISK-29656 #close
Change-Id: Ib464e9eb6e13dc683a846286798fecff4fd943cb
Adds the STRBETWEEN function, which can be used to insert a
substring between each character in a string. For instance,
this can be used to insert pauses between DTMF tones in a
string of digits.
ASTERISK-29627
Change-Id: Ice23009d4a8e9bb9718d2b2301d405567087d258
Adds the DIRNAME and BASENAME functions, which are
wrappers around the corresponding C library functions.
These can be used to safely and conveniently work with
file paths and names in the dialplan.
ASTERISK-29628 #close
Change-Id: Id3aeb907f65c0ff96b6e57751ff0cb49d61db7f3
Up until now, all of the logic used to translate
arguments to the Say applications has been
directly coupled to playback, preventing other
modules from using this logic.
This refactors code in say.c and adds a SAYFILES
function that can be used to retrieve the file
names that would be played. These can then be
used in other applications or for other purposes.
Additionally, a SayMoney application and a SayOrdinal
application are added. Both SayOrdinal and SayNumber
are also expanded to support integers greater than
one billion.
ASTERISK-29531
Change-Id: If9718c89353b8e153d84add3cc4637b79585db19
The MIN, MAX, and ABS functions all support float
arguments, but currently return floats even if the
arguments are all integers and the response is
a whole number, in which case the user is likely
expecting an integer. This casts the float to an integer
before printing into the response buffer if possible.
ASTERISK-29495
Change-Id: I902d29eacf3ecd0f8a6a5e433c97f0421d205488
Adds a function to scramble audio on a channel using
whole spectrum frequency inversion. This can be used
as a privacy enhancement with applications like
ChanSpy or other potentially sensitive audio.
ASTERISK-29542
Change-Id: I01020769d91060a1f56a708eb405f87648d1a67e
Adds function to selectively drop specified frames
in the TX or RX direction on a channel, including
control frames.
ASTERISK-29478
Change-Id: I8147c9d55d74e2e48861edba6b22f930920541ec
The scenario where a channel still has an associated datastore we
cannot unload since there is a function pointer to the destroy and fixup
functions in play. Thus increase the module ref count whenever we
allocate a datastore, and decrease it during destroy.
In order to tighten the race that still exists in spite of this (below)
add some extra failure cases to prevent allocations in these cases.
Race:
If module ref is zero, an LOCK or TRYLOCK is invoked (near)
simultaneously on a channel that has NOT PREVIOUSLY taken a lock, and if
in such a case the datastore is created *prior* to unloading being set
to true (first step in module unload) then it's possible that the module
will unload with the destructor being called (and segfault) post the
module being unloaded. The module will however wait for such locks to
release prior to unloading.
If post that we can recheck the module ref before returning the we can
(in theory, I think) eliminate the last of the race. This race is
mostly theoretical in nature.
Change-Id: I21a514a0b56755c578a687f4867eacb8b59e23cf
Signed-off-by: Jaco Kroon <jaco@uls.co.za>
For example:
arthur*CLI> dialplan locks show
func_lock locks:
Name Requesters Owner
uls-autoref 0 (unlocked)
1 total locks listed.
Obviously other potentially useful stats could be added (eg, how many
times there was contention, how many times it failed etc ... but that
would require keeping the stats and I'm not convinced that's worth the
effort. This was useful to troubleshoot some other issues so submitting
it.
Change-Id: Ib875e56feb49d523300aec5f36c635ed74843a9f
Signed-off-by: Jaco Kroon <jaco@uls.co.za>
AST_TRAVERSE accessess current as current = current->(field).next ...
and since we free current (and ast_free poisons the memory) we either
end up on a ast_mutex_lock to a non-existing lock that can never be
obtained, or a segfault.
Incidentally add logging in the "we have to wait for a lock to release"
case, and remove an ineffective statement that sets memory that was just
cleared by ast_calloc to zero.
Change-Id: Id19ba3d9867b23d0e6783b97e6ecd8e62698b8c3
Signed-off-by: Jaco Kroon <jaco@uls.co.za>
In two places we bail out with failure after we've already incremented
the requesters counter, if this occured then it would effectively result
in unload to wait indefinitely, thus preventing clean shutdown.
Change-Id: I362a6c0dc424f736d4a9c733d818e72d19675283
Signed-off-by: Jaco Kroon <jaco@uls.co.za>
Introduces three new dialplan functions, MIN and MAX,
which can be used to calculate the minimum or
maximum of up to two numbers, and ABS, an absolute
value function.
ASTERISK-29431
Change-Id: I2bda9269d18f9d54833c85e48e41fce0e0ce4d8d
Up until now, the VOLUME function has been write
only, so that TX/RX values can be set but not
read afterwards. Now, previously set TX/RX values
can be read later.
ASTERISK-29439
Change-Id: Ia23e92fa2e755c36e9c8e69f2940d2703ccccb5f
This change embeds the MODULEINFO block of modules
into the core XML documentation. This provides a shared
mechanism for use by both menuselect and Asterisk for
information and a definitive source of truth.
ASTERISK-29335
Change-Id: Ifbfd5c700049cf320a3e45351ac65dd89bc99d90
minargs enables enforcing of minimum count of arguments to pass to
func_odbc, so if you're unconditionally using ARG1 through ARG4 then
this should be set to 4. func_odbc will generate an error in this case,
so for example
[FOO]
minargs = 4
and ODBC_FOO(a,b,c) in dialplan will now error out instead of using a
potentially leaked ARG4 from Gosub().
ARGC is needed if you're using optional argument, to verify whether or
not an argument has been passed, else it's possible to use a leaked ARGn
from Gosub (app_stack). So now you can safely do
${IF($[${ARGC}>3]?${ARGV}:default value)} kind of thing.
Change-Id: I6ca0b137d90b03f6aa9c496991f6cbf1518f6c24
Signed-off-by: Jaco Kroon <jaco@uls.co.za>
Under contention it becomes possible that multiple channels will be told
they successfully obtained the lock, which is a bug. Please refer
ASTERISK-29217
This introduces a couple of changes.
1. Replaces requesters ao2 container with simple counter (we don't
really care who is waiting for the lock, only how many). This is
updated undex ->mutex to prevent memory access races.
2. Correct semantics for ast_cond_timedwait() as described in
pthread_cond_broadcast(3P) is used (multiple threads can be released
on a single _signal()).
3. Module unload races are taken care of and memory properly cleaned
up.
Change-Id: I6f68b5ec82ff25b2909daf6e4d19ca864a463e29
Signed-off-by: Jaco Kroon <jaco@uls.co.za>
Currently any response from res_curl where we get an answer from the
web server, regardless of what the response is (404, 403 etc.) Asterisk
currently treats it as a success. This patch allows you to set which
codes should be considered as a failure by Asterisk. If say we set
failurecodes=404,403 then when using curl in realtime if a server gives
a 404 error Asterisk will try to failover to the next option set in
extconfig.conf
ASTERISK-28825
Reported by: Dovid Bender
Code by: Gobinda Paul
Change-Id: I94443e508343e0a3e535e51ea6e0562767639987
Because we use shared thread-local cURL instances, we need to ensure
that the state of the cURL instance is correct before each invocation.
In the case of custom headers, we were not resetting cURL's internal
HTTP header pointer which could result in a crash if subsequent
requests do not configure custom headers.
ASTERISK-29085 #close
Change-Id: I8b4ab34038156dfba613030a45f10e932d2e992d
The following fields return an error when read from dialplan:
- exten
- context
- userfield
- channame
ASTERISK-28796 #close
Change-Id: Ieacaac629490f8710fdacc9de80ed5916c5f6ee2
We allow for 'maxredirs' to be set, but this value is ignored when
followlocation is not enabled which, by default, it is not.
ASTERISK-17491 #close
Reported by: candrews
Change-Id: I96a4ab0142f2fb7d2e96ff976f6cf7b2982c761a
add missing argument "rtt" and "status" to the documentation
The change to the dtd file allow an enumlist to contain one or many
configOptionToEnum or enum.
This is different from the previous patch I submitted when you could have a
configOptionToEnum or (a configOptionToEnum followed by one or manu enums) or
(one or many enums)
ASTERISK-28626
Change-Id: Ia71743ee7ec813f40297b0ddefeee7909db63b6d
This reverts commit 7e3015d779.
Reason for revert: Regression in XML validation.
validity error : Content model of enumlist is not determinist:
(configOptionToEnum | (configOptionToEnum , enum+) | enum+)
As we are preparing to do releases and this is not critical
I am reverting this for now until resolved.
Change-Id: I30c2295f9d7f0a0475674ee77071a7ebabf5b83f
If the last line of a file does not have a terminating EOL sequence, we
potentially add garbage to the value returned from the FILE() function.
There is no overflow potential here as we are reading from a buffer of a
known size, we are just reading too much of it.
ASTERISK-26481 #close
Change-Id: I50dd4fcf416fb3c83150040a1a79a59d9eb1ae01
When user wants to send json data, the default Content-Type header
is incorect (application/x-www-form-urlencoded). This patch allows
to set any custom headers so the Content-Type header can be
overriden. User can set multiple headers by multiple calls of
curlopt(). This approach is not consistent with other parameters,
but is more readable in dialplan than one call with multiple
headers.
ASTERISK-28613
Change-Id: I4dd68c3f4e25362ef941d73a3861f58348dcfbf9
This change adds support to the JITTERBUFFER dialplan function
for audio and video synchronization. When enabled the RTCP SR
report is used to produce an NTP timestamp for both the audio and
video streams. Using this information the video frames are queued
until their NTP timestamp is equal to or behind the NTP timestamp
of the audio. The audio jitterbuffer acts as the leader deciding
when to shrink/grow the jitterbuffer when adaptive is in use. For
both adaptive and fixed the video buffer follows the size of the
audio jitterbuffer.
ASTERISK-28533
Change-Id: I3fd75160426465e6d46bb2e198c07b9d314a4492
