Adds the Signal and WaitForSignal
applications, which can be used for inter-channel
signaling in the dialplan.
Signal supports sending a signal to other channels
listening for a signal of the same name, with an
optional data payload. The signal is received by
all channels waiting for that named signal.
ASTERISK-29810 #close
Change-Id: Ic34439de3d60f8609357666a465c354d81f5fef3
Adds option to app_directory to specify a filename from which to
read configuration instead of voicemail.conf ie;
same => n,Directory(,,c(directory.conf))
This configuration should contain a list of extensions using the
voicemail.conf format, ie;
2020=2020,Dog Dog,,,,attach=no|saycid=no|envelope=no|delete=no
ASTERISK-30404
Change-Id: Id58ccb1344ad1e563fa10db12f172fbd104a9d13
Adds support for arrays to JSON_DECODE by allowing the
user to print out entire arrays or index a particular
key or print the number of keys in a JSON array.
Additionally, adds support for recursively iterating a
JSON tree in a single function call, making it easier
to parse JSON results with multiple levels. A maximum
depth is imposed to prevent potentially blowing
the stack.
Also fixes a bug with the unit tests causing an empty
string to be printed instead of the actual test result.
ASTERISK-29913 #close
Change-Id: I603940b216a3911b498fc6583b18934011ef5d5b
Adds the overlap_context option, which can be used
to explicitly specify a context to use for overlap
dialing extension matches, rather than forcibly
using the context configured for the endpoint.
ASTERISK-30262 #close
Change-Id: Ibbcd4a8b11402428a187fb56b8d4e7408774a0db
Added NULL pointer check and channel lock to prevent resource release
while the chanspy is processing.
ASTERISK-29604
Change-Id: Ibdc675f98052da32333b19685b1708a3751b6d24
Variable references within global variable assignments are now
expanded rather than being included literally.
ASTERISK-30406 #close
Change-Id: I136e8d6395e90a4c92d9777a46a7bc3edb08d05d
In Asterisk 11, if a channel was redirected away during Playback(),
the PLAYBACKSTATUS variable would be set to SUCCESS. In Asterisk 12
(specifically commit 7d9871b394) that
behavior was inadvertently changed and the same operation would result
in the PLAYBACKSTATUS variable being set to FAILED. The Asterisk 11
behavior has been restored.
Partial fix for ASTERISK~25661.
Change-Id: I53f54e56b59b61c99403a481b6cb8d88b5a559ff
Rounding issues with double math were causing rtp timestamp
slips in outgoing packets. We're now back to integer math
and are getting no more slips.
ASTERISK-30391
Change-Id: I6ba992b49ffdf9ebea074581dfa784a188c661a4
For most modules that interacted with app_macro, this change is limited
to no longer looking for the current context from the macrocontext when
set. Additionally, the following modules are impacted:
app_dial - no longer supports M^ connected/redirecting macro
app_minivm - samples written using macro will no longer work.
The sample needs a re-write
app_queue - can no longer a macro on the called party's channel.
Use gosub which is currently supported
ccss - no callback macro, gosub only
app_voicemail - no macro support
channel - remove macrocontext and priority, no connected line or
redirection macro options
options - stdexten is deprecated to gosub as the default and only
pbx - removed macrolock
pbx_dundi - no longer look for macro
snmp - removed macro context, exten, and priority
ASTERISK-30304
Change-Id: I830daab293117179b8d61bd4df0d971a1b3d07f6
Each playback of WAV files results in logging
"Skipping unknown block 'LIST'".
To prevent unnecessary flooding of this DEBUG log this patch replaces
ast_log(LOG_DEBUG, ...) by ast_debug(1, ...).
Change-Id: Iaa09cf19c5348a05385518fdb8cb181b45fe05f0
Add ability to set HANGUPCAUSE when SIP causecode received in BYE (in addition to currently supported Q.850).
ASTERISK-30319 #close
Change-Id: I3f55622dc680ce713a2ffb5a458ef5dd39fcf645
-----------------
This commit reinstates MES with some casting fixes to the
functions in time.h that convert between doubles and timeval
structures. The casting issues were causing incorrect
timestamps to be calculated which caused transcoding from/to
G722 to produce bad or no audio.
ASTERISK-30391
-----------------
This module has been updated to provide additional
quality statistics in the form of an Asterisk
Media Experience Score. The score is avilable using
the same mechanisms you'd use to retrieve jitter, loss,
and rtt statistics. For more information about the
score and how to retrieve it, see
https://wiki.asterisk.org/wiki/display/AST/Media+Experience+Score
* Updated chan_pjsip to set quality channel variables when a
call ends.
* Updated channels/pjsip/dialplan_functions.c to add the ability
to retrieve the MES along with the existing rtcp stats when
using the CHANNEL dialplan function.
* Added the ast_debug_rtp_is_allowed and ast_debug_rtcp_is_allowed
checks for debugging purposes.
* Added several function to time.h for manipulating time-in-samples
and times represented as double seconds.
* Updated rtp_engine.c to pass through the MES when stats are
requested. Also debug output that dumps the stats when an
rtp instance is destroyed.
* Updated res_rtp_asterisk.c to implement the calculation of the
MES. In the process, also had to update the calculation of
jitter. Many debugging statements were also changed to be
more informative.
* Added a unit test for internal testing. The test should not be
run during normal operation and is disabled by default.
Change-Id: I4fce265965e68c3fdfeca55e614371ee69c65038
If native HTTP is disabled but HTTPS is enabled and status page enabled
too, Core/HTTP crashes while loading. 'global_http_server' references
to NULL, but the status page tries to dereference it.
The patch adds a check for HTTP is enabled.
ASTERISK-30379 #close
Change-Id: I11b02fc920b72aaed9c809fc43210523ccfdc249
Currently, if a module declines to load, dlopen is called
to register the module but dlclose never gets called.
Furthermore, loader.c currently doesn't allow dlclose
to ever get called on the module, since it declined to
load and the unload function bails early in this case.
This can be problematic if a module is updated, since the
new module cannot be loaded into memory since we haven't
closed all references to it. To fix this, we now allow
modules to be unloaded, even if they never "loaded" in
Asterisk itself, so that dlclose is called and the module
can be properly cleaned up, allowing the updated module
to be loaded from scratch next time.
ASTERISK-30345 #close
Change-Id: Ifc743aadfa85ebe3284e02a63e124dafa64988d5
Adds a new application, Broadcast, which can be used for
one-to-many transmission and many-to-one reception of
channel audio in Asterisk. This is similar to ChanSpy,
except it is designed for multiple channel targets instead
of a single one. This can make certain kinds of audio
manipulation more efficient and streamlined. New kinds
of audio injection impossible with ChanSpy are also made
possible.
ASTERISK-30180 #close
Change-Id: I7ba72f765dbab9b58deeae028baca3f4f8377726
Since text frames contain a text body, make FRAME_TRACE
more useful for text frames by actually printing the text.
ASTERISK-30353 #close
Change-Id: Ia6ce3d15cecd7a673a528d34faac86854a2bab50
This removes the deprecated NoCDR application, which
was deprecated in Asterisk 12, having long been fully
superseded by the CDR_PROP function.
The deprecated e option to ResetCDR is also removed
for the same reason.
ASTERISK-30371 #close
Change-Id: Id9ed094d8e4baf98bcbc610035c2295bfafe9ec0
Do not crash when a URL has no path component as in this case the
ast_uri_path function will return NULL. Make the code cope with not
having a path.
The below would crash
> media cache create http://google.com /tmp/foo.wav
Thread 1 "asterisk" received signal SIGSEGV, Segmentation fault.
0x0000ffff836616cc in strrchr () from /lib/aarch64-linux-gnu/libc.so.6
(gdb) bt
#0 0x0000ffff836616cc in strrchr () from /lib/aarch64-linux-gnu/libc.so.6
#1 0x0000ffff43d43a78 in file_extension_from_string (str=<optimized out>, buffer=buffer@entry=0xffffca9973c0 "",
capacity=capacity@entry=64) at res_http_media_cache.c:288
#2 0x0000ffff43d43bac in file_extension_from_url_path (bucket_file=bucket_file@entry=0x3bf96568,
buffer=buffer@entry=0xffffca9973c0 "", capacity=capacity@entry=64) at res_http_media_cache.c:378
#3 0x0000ffff43d43c74 in bucket_file_set_extension (bucket_file=bucket_file@entry=0x3bf96568) at res_http_media_cache.c:392
#4 0x0000ffff43d43d10 in bucket_file_run_curl (bucket_file=0x3bf96568) at res_http_media_cache.c:555
#5 0x0000ffff43d43f74 in bucket_http_wizard_create (sorcery=<optimized out>, data=<optimized out>, object=<optimized out>)
at res_http_media_cache.c:613
#6 0x0000000000487638 in bucket_file_wizard_create (sorcery=<optimized out>, data=<optimized out>, object=<optimized out>)
at bucket.c:191
#7 0x0000000000554408 in sorcery_wizard_create (object_wizard=object_wizard@entry=0x3b9f0718,
details=details@entry=0xffffca9974a8) at sorcery.c:2027
#8 0x0000000000559698 in ast_sorcery_create (sorcery=<optimized out>, object=object@entry=0x3bf96568) at sorcery.c:2077
#9 0x00000000004893a4 in ast_bucket_file_create (file=file@entry=0x3bf96568) at bucket.c:727
#10 0x00000000004f877c in ast_media_cache_create_or_update (uri=0x3bfa1103 "https://google.com",
file_path=0x3bfa1116 "/tmp/foo.wav", metadata=metadata@entry=0x0) at media_cache.c:335
#11 0x00000000004f88ec in media_cache_handle_create_item (e=<optimized out>, cmd=<optimized out>, a=0xffffca9976b8)
at media_cache.c:640
ASTERISK-30375 #close
Change-Id: I6a9433688cb5d3d4be8758b7642d923bdde6c273
The if statement here is always false after the for
loop finishes, so variables are never appended.
This removes that to properly append to the end
of the variable list.
ASTERISK-30351 #close
Reported by: Sebastian Gutierrez
Change-Id: I1b7f8b85a8918f6a814cb933a479d4278cf16199
json.h contains macros to get a string and an integer
from a JSON object. However, the macro to do this for
JSON reals is missing. This adds that.
ASTERISK-30361 #close
Change-Id: I8d0e28d763febf27b05801cdc83b73282aa6ee7a
When Asterisk receives a new websocket conenction, it creates a new
pjsip transport for it and copies connection data into it. The
transport manager then uses the remote IP address and port on the
transport to create a monitor for each connection. However, the
remote port wasn't being copied, only the IP address which meant
that the transport manager was creating only 1 monitoring entry for
all websocket connections from the same IP address. Therefore, if
one of those connections failed, it deleted the transport taking
all the the connections from that same IP address with it.
* We now copy the remote port into the created transport and the
transport manager behaves correctly.
ASTERISK-30369
Change-Id: Ib506d40897ea6286455ac0be4dfbb0ed43b727e1
This module has been updated to provide additional
quality statistics in the form of an Asterisk
Media Experience Score. The score is avilable using
the same mechanisms you'd use to retrieve jitter, loss,
and rtt statistics. For more information about the
score and how to retrieve it, see
https://wiki.asterisk.org/wiki/display/AST/Media+Experience+Score
* Updated chan_pjsip to set quality channel variables when a
call ends.
* Updated channels/pjsip/dialplan_functions.c to add the ability
to retrieve the MES along with the existing rtcp stats when
using the CHANNEL dialplan function.
* Added the ast_debug_rtp_is_allowed and ast_debug_rtcp_is_allowed
checks for debugging purposes.
* Added several function to time.h for manipulating time-in-samples
and times represented as double seconds.
* Updated rtp_engine.c to pass through the MES when stats are
requested. Also debug output that dumps the stats when an
rtp instance is destroyed.
* Updated res_rtp_asterisk.c to implement the calculation of the
MES. In the process, also had to update the calculation of
jitter. Many debugging statements were also changed to be
more informative.
* Added a unit test for internal testing. The test should not be
run during normal operation and is disabled by default.
ASTERISK-30280
Change-Id: I458cb9a311e8e5dc1db769b8babbcf2e093f107a
pbx_exec makes a channel snapshot before executing applications.
This doesn't cause an issue during normal dialplan execution
where pbx_exec is called over and over again in succession.
However, if pbx_exec is called "one off", e.g. using
ast_pbx_exec_application, then a channel snapshot never ends
up getting made after the executed application returns, and
inaccurate snapshot information will linger for a while, causing
"core show channels", etc. to show erroneous info.
This is fixed by manually making a channel snapshot at the end
of ast_pbx_exec_application, since we anticipate that pbx_exec
might not get called again immediately.
ASTERISK-30367 #close
Change-Id: I2a5131053aa9d11badbc0ef2ef40b1f83d0af086
Currently, there is no Caller ID available to us when
checking for an extension match when handling INVITEs.
As a result, extension patterns that depend on the Caller ID
are not matched and calls may be incorrectly rejected.
The Caller ID is not available because the supplement that
adds Caller ID to the session does not execute until after
this check. Supplement callbacks cannot yet be executed
at this point since the session is not yet in the appropriate
state.
To fix this without impacting existing behavior, the Caller ID
number is now retrieved before attempting to pattern match.
This ensures pattern matching works correctly and there is
no behavior change to the way supplements are called.
ASTERISK-28767 #close
Change-Id: Iec7f5a3b90e51b65ccf74342f96bf80314b7cfc7
This removes the ImportVar and SetAMAFlags applications
which have been deprecated since Asterisk 12, but were
never removed previously.
Additionally, it removes remnants of defunct options
that themselves were removed years ago.
ASTERISK-30335 #close
Change-Id: I749520c7b08d4c9d5eebbf640d4fbc81950eda8d
When a call is put on hold and it has moh_passthrough and rtp_timeout
set on the endpoint, the wrong timeout will be used. rtp_timeout_hold is
expected to be used, but rtp_timeout is used instead. This change adds a
couple of checks for locally_held to determine if rtp_timeout_hold needs
to be used instead of rtp_timeout.
ASTERISK-30350
Change-Id: I7b106fc244332014216d12bba851cefe884cc25f
Fixes a negative offset warning by initializing
the buffer to empty.
Additionally, although it doesn't currently complain
about it, the size of a buffer is increased to
accomodate the maximum size contents it could have.
ASTERISK-30240 #close
Change-Id: I8eecedf14d3f2a75864797f802277cac89a32877
When ast_stream_and_wait returns an error (for example, when attempting
to stream to a channel after hangup) the stream is not closed, and
callers typically do not check the return code. This results in leaking
file descriptors, leading to resource exhaustion.
This change ensures that the stream is closed in case of error.
ASTERISK-30198 #close
Reported-by: Julien Alie
Change-Id: Ie46b67314590ad75154595a3d34d461060b2e803
Currently, if a user attempts to set a Caller ID related
function to an invalid value, a warning is emitted,
except for when setting the redirecting reason.
We now emit a warning if we were unable to successfully
parse the user-provided reason.
ASTERISK-30332 #close
Change-Id: Ic341f5d5f7303b6f1115549be64db58a85944f5a
Removes see-also references to applications that don't
exist anymore (removed in Asterisk 19),
so these dead links don't show up on the wiki.
ASTERISK-30347 #close
Change-Id: I9539bc30f57cd65aa4e2d5ce8185eafa09567909
Fix aor lookup on sip path addition. Issue happens in case of dialing
with @ and overriding user part of RURI.
ASTERISK-30100 #close
Reported-by: Yury Kirsanov
Change-Id: I3f2c42a583578c94397b113e32ca3ebf2d600e13
The `ast_geoloc_datastore_add_eprofile` function does not return 0 on
success, it returns the size of the underlying datastore. This means
that the datastore will be freed and its pointer set to NULL when no
error occured at all.
ASTERISK-30346
Change-Id: Iea9b209bd1244cc57b903b9496cb680c356e4bb9
When adding AOC to an outgoing response the code
assumed that a body would exist for comparing the
Content-Type. This isn't always true.
The code now checks to make sure the response has
a body before checking the Content-Type.
ASTERISK-21502
Change-Id: Iaead371434fc3bc693dad487228106a7d7a5ac76
chan_sip supported sending AOC-D and AOC-E information in SIP INFO
messages in an "AOC" header in a format that was originally defined by
Snom. In the meantime, ETSI TS 124 647 introduced an XML-based AOC
format that is supported by devices from multiple vendors, including
Snom phones with firmware >= 8.4.2 (released in 2010).
This commit adds a new res_pjsip_aoc module that inserts AOC information
into outgoing messages or sends SIP INFO messages as described below.
It also fixes a small issue in res_pjsip_session which didn't always
call session supplements on outgoing_response.
* AOC-S in the 180/183/200 responses to an INVITE request
* AOC-S in SIP INFO (if a 200 response has already been sent or if the
INVITE was sent by Asterisk)
* AOC-D in SIP INFO
* AOC-D in the 200 response to a BYE request (if the client hangs up)
* AOC-D in a BYE request (if Asterisk hangs up)
* AOC-E in the 200 response to a BYE request (if the client hangs up)
* AOC-E in a BYE request (if Asterisk hangs up)
The specification defines one more, AOC-S in an INVITE request, which
is not implemented here because it is not currently possible in
Asterisk to have AOC data ready at this point in call setup. Once
specifying AOC-S via the dialplan or passing it through from another
SIP channel's INVITE is possible, that might be added.
The SIP INFO requests are sent out immediately when the AOC indication
is received. The others are inserted into an appropriate outgoing
message whenever that is ready to be sent. In the latter case, the XML
is stored in a channel variable at the time the AOC indication is
received. Depending on where the AOC indications are coming from (e.g.
PRI or AMI), it may not always be possible to guarantee that the AOC-E
is available in time for the BYE.
Successfully tested AOC-D and both variants of AOC-E with a Snom D735
running firmware 10.1.127.10. It does not appear to properly support
AOC-S however, so that could only be tested by inspecting SIP traces.
ASTERISK-21502 #close
Reported-by: Matt Jordan <mjordan@digium.com>
Change-Id: Iebb7ad0d5f88526bc6629d3a1f9f11665434d333
Adds support for the capture agent name field
of the Homer protocol to Asterisk by allowing
users to specify a name that will be sent to
the HEP server.
ASTERISK-30322 #close
Change-Id: I6136583017f9dd08daeb8be02f60fb8df4639a2b
This fixes a small typo in the from_domain documentation on the endpoint documentation
ASTERISK-30328 #close
Change-Id: Ia6f0897c3f5cab899ef2cde6b3ac07265b8beb21
Adds the If, ElseIf, Else, ExitIf, and EndIf
applications for conditional execution
of a block of dialplan, similar to the While,
EndWhile, and ExitWhile applications. The
appropriate branch is executed at most once
if available and may be broken out of while
inside.
ASTERISK-29497
Change-Id: I3aa3bd35a5add82465c6ee9bd86b64601f0e1f49
Some SIP devices use an empty extension for PLAR functionality.
Rather than rejecting these empty extensions, we now use the s
extension for such calls to mirror the existing PLAR functionality
in Asterisk (e.g. chan_dahdi).
ASTERISK-30265 #close
Change-Id: I0861a405cd49bbbf532b52f7b47f0e2810832590
Updates the documentation for the 'contact_user' field to point out the
default outbound contact if no contact_user is specified 's'
ASTERISK-30316 #close
Change-Id: I61f24fb9164e4d07e05908a2511805281874c876
Adds support for custom URI and header parameters
in the From header in PJSIP. Parameters can be
both set and read using this function.
ASTERISK-30150 #close
Change-Id: Ifb1bc3c512ad5f6faeaebd7817f004a2ecbd6428
msg_create_from_file currently does not dispatch emails,
which means that applications using this function, such
as MixMonitor, will not trigger notifications to users
(only AMI events are sent our currently). This is inconsistent
with other ways users can receive voicemail.
This is fixed by adding an option that attempts to send
an email and falling back to just the notifications as
done now if that fails. The existing behavior remains
the default.
ASTERISK-30283 #close
Change-Id: I597cbb9cf971a18d8776172b26ab187dc096a5c7
When passing a JSON body to the 'create' channel route
it would be converted into Asterisk variables, but never
freed resulting in a memory leak.
This change makes it so that the variables are freed in
all cases.
ASTERISK-30344
Change-Id: I924dbd866a01c6073e2d6fb846ccaa27ef72d49d
The commit that rearchitected media formats,
a2c912e997 (ASTERISK_23114)
introduced a regression by improperly translating code in res_adsi.c.
In particular, the pointer to the frame buffer was initialized
at the top of adsi_careful_send, rather than dynamically updating it
for each frame, as is required.
This resulted in the first frame being repeatedly sent,
rather than advancing through the frames.
This corrupted the transmission of the CAS to the CPE,
which meant that CPE would never respond with the DTMF acknowledgment,
effectively completely breaking ADSI functionality.
This issue is now fixed, and ADSI now works properly again.
ASTERISK-29793 #close
Change-Id: Icdeddf733eda2981c98712d1ac9cddc0db507dbe
