Adds missing documentation for some channel,
bridge, and queue events.
ASTERISK-24427
ASTERISK-29515
Change-Id: I92b06b88c8cadc0155f95ebe3e870b3e795a8c64
The current TCP client connect code, blocks and does not handle EINTR
error case.
This patch makes the client socket non-blocking while connecting,
ensures a connect does not immediately fail due to EINTR "errors",
and adds a connect timeout option.
The original client start call sets the new timeout option to
"infinite", thus making sure old, orginal behavior is retained.
ASTERISK-29746 #close
Change-Id: I907571843a83e43c0742b95a64785f4411f02671
Adds tech-agnostic support for SF signaling
by adding SF sender and receiver applications
as well as Dial integration.
ASTERISK-29802 #close
Change-Id: I7ec50752e9a661af639425e5d1e339f17411bcad
A previous patch for ASTERISK_29578 caused a 'leak' of
extension state information across queues, causing the
state of the first member of unrelated queues to be
updated in addition to the correct member. Which queues
and members depended on the order of queues in the
iterator.
Additionally, it is possible to use the same 'hint:' on
multiple queue members, so the update cannot break out
of the update loop early when a match is found.
ASTERISK-29806 #close
Change-Id: If2c1d1cc2a752afd9286d79710fc818596e7a7ad
SayAlpha, SayAlphaCase, SayDigits, SayMoney, SayNumber, SayOrdinal,
and SayPhonetic all claim to allow DTMF interruption if the
SAY_DTMF_INTERRUPT channel variable is set to a truthy value, but we
are failing to break out of a given 'say' application if DTMF actually
occurs.
ASTERISK-29816 #close
Change-Id: I6a96e0130560831d2cb45164919862b9bcb6287e
The ast_rtp_codecs_payloads functions do not preserve the order in which
the payloads were specified on an incoming SDP media line. This leads to
a problem with the codec negotiation functionality, as the format
capabilities of the stream are extracted from the ast_rtp_codecs. This
commit moves the ast_rtp_codec to ast_format conversion to the place
where the order is still known.
ASTERISK-28863
ASTERISK-29320
Change-Id: I3aabcfed3f379c36654f59c1872c313d0cb57e25
It's not safe to keep the channel locked while locking
the peer Local channel, as it can result in a deadlock.
This change unlocks it during this time but keeps the
bridge locked to ensure nothing changes about the bridge.
ASTERISK-29821
Change-Id: Ib68eb7037e5a479bcc2aceee77337cdde1fbdde6
When test_timezone_watch runs very near a DST transition,
two time zones that would otherwise be expected to report the same
time can differ because of the DST transition.
Instead of having the test fail when this happens, report the
times, time zones, and dst flags.
ASTERISK-29722
Change-Id: Id59bdac8b277e14343ccdf0c99b89e92f79f316a
Adding upstream patch for pull request...
https://github.com/pjsip/pjproject/pull/2920
---------------------------------------------------------------
sip_inv: Additional multipart support (#2919)
sip_inv.c:inv_check_sdp_in_incoming_msg() deals with multipart
message bodies in rdata correctly. In the case where early media is
involved though, the existing sdp has to be retrieved from the last
tdata sent in this transaction. This, however, always assumes that
the sdp sent is in a non-multipart body. While there's a function
to retrieve the sdp from multipart and non-multpart rdata bodies,
no similar function for tdata exists. So...
* The existing pjsip_rdata_get_sdp_info2 was refactored to
find the sdp in any body, multipart or non-multipart, and
from either an rdata or tdata. The new function is
pjsip_get_sdp_info. This new function detects whether the
pjsip_msg->body->data is the text representation of the sdp
from an rdata or an existing pjmedia_sdp_session object
from a tdata, or whether pjsip_msg->body is a multipart
body containing either of the two sdp formats.
* The exsting pjsip_rdata_get_sdp_info and pjsip_rdata_get_sdp_info2
functions are now wrappers that get the body and Content-Type
header from the rdata and call pjsip_get_sdp_info.
* Two new wrappers named pjsip_tdata_get_sdp_info and
pjsip_tdata_get_sdp_info2 have been created that get the body
from the tdata and call pjsip_get_sdp_info.
* inv_offer_answer_test.c was updated to test multipart scenarios.
ASTERISK-29804
Change-Id: I483c7c3d413280c9e247a96ad581278347f9c71b
When OUTPUTDIR is set to another directory and the
--delete-results-after is set, the resulting txt files are
not deleted.
ASTERISK-29794 #close
Change-Id: I1c0071f6809a1e3f5cfc455d6eb08378bc0d7286
The variable cp4 in a variable substitution function
can potentially be used without being initialized
currently. This causes Asterisk to no longer compile.
This initializes cp4 to NULL to make the compiler
happy.
ASTERISK-29803 #close
Change-Id: I392579cbb76db2795d5820c9427cf55fbcee9e72
added that we set DIALEDPEERNUMBER on the outgoing channels
so it is avalible in b(content^extension^line)
this add the same behaviour as Dial
ASTERISK-29795
Change-Id: Icbc589ea2066f0c401a892bf478f6b2fd44e62f6
Previously, it was only possible to have one HTTP server in Asterisk.
With this patch it is now possible to have multiple HTTP servers
listening on different addresses.
Note, this behavior has only been made available through an API call
from within the TEST_FRAMEWORK. Specifically, this feature has been
added in order to allow unit test to create/start and stop servers,
if one has not been enabled through configuration.
Change-Id: Ic5fb5f11e62c019a1c51310f4667b32a4dae52f5
Currently, Asterisk doesn't throw warnings if options
are passed into applications that don't accept them.
This can confuse users if they're unaware that they
are doing something wrong.
This adds an additional check to parse_options so that
a warning is thrown anytime an option is parsed that
doesn't exist in the parsing application, so that users
are notified of the invalid usage.
ASTERISK-29801 #close
Change-Id: Id029274a57135caca193c913307a63fd75e24679
added support for playing the correct plural sound file
dependen on where you have 1 or multipe messages
based on the existing SE/NO code
ASTERISK-29797
Change-Id: I88aa814d02f3772bb80b474204b1ffb26fe438c2
Adds a ReceiveText application that can be used in
conjunction with SendText. Currently, there is no
way in Asterisk to receive text in the dialplan
(or anywhere else, really). This allows for Asterisk
to be the recipient of text instead of just the sender.
ASTERISK-29759 #close
Change-Id: Ica2c354a42bff69f323a0493d3a7cd0fb129d52d
The enum values for ast_strsep_flags includes
AST_STRSEP_STRIP. However, some comments reference
AST_SEP_STRIP, which doesn't exist. This fixes
these comments to use the correct value.
ASTERISK-29800 #close
Change-Id: If7bbd0c0e6226a211d25ddf9d1629347e2674943
Currently MSet can only parse a maximum of 24 variables.
If more variables are provided to MSet, the 24th variable
will simply contain the remainder of the string and the
remaining variables thereafter will never get set.
This increases the number of variables that can be parsed
in one go from 24 to 99. Additionally, documentation is added
since this limitation is currently undocumented and is
confusing to users who encounter this limitation.
ASTERISK-29766 #close
Change-Id: I3fe35b462dedec0a452fd9ea7f92c920a3939f16
Attempting to access ${CHANNEL(ruri)} in a pre-dial handler before
initiating an outgoing call will cause Asterisk to crash. This is
because a null field is accessed, resulting in an offset from null and
subsequent memory access violation.
Since RURI is not guaranteed to exist, we now check if the base
pointer is non-null before calculating an offset.
ASTERISK-29772
Change-Id: Icd3b02f07256bbe6615854af5717074087b95a83
Adds the JSON_DECODE function for parsing JSON in the
dialplan. JSON parsing already exists in the Asterisk
core and is used for many different things. This
function exposes the basic parsing capability to
the user in the dialplan, for instance, in conjunction
with CURL for using API responses.
ASTERISK-29706 #close
Change-Id: Iea60c49a7358dfdc2db60803cdc9a742f808ba2c
Includes some minor updates to extensions.conf
and iax.conf. In particular, the demonstration
of macros in extensions.conf is removed, as
Macro is deprecated and will be removed soon.
These examples have been replaced with examples
demonstrating the usage of Gosub instead.
The older exten => ...,n syntax is also mostly
replaced with the same keyword to demonstrate the
newer, more concise way of defining extensions.
IAXTEL no longer exists, so this example is replaced
with something more generic.
Some documentation is also added to extensions.conf
and iax.conf to clarify some of the new expanded
encryption capabilities with IAX2.
ASTERISK-29758 #close
Change-Id: I04fba9671aa1ee9ba1bd5027061f80bbe38e7b46
Currently, variable substitution involving dialplan
extensions is quite clunky since it entails obtaining
the current dialplan location, backing it up, storing
the desired variables for substitution on the channel,
performing substitution, then restoring the original
location.
In addition to being clunky, things could also go wrong
if an async goto were to occur and change the dialplan
location during a substitution.
Fundamentally, there's no reason it needs to be done this
way, so new API is added to allow for directly passing in
the dialplan location for the purposes of variable
substitution so we don't need to mess with the channel
information anymore. Existing API is not changed.
ASTERISK-29745 #close
Change-Id: I23273bf27fa0efb64a606eebf9aa8e2f41a065e4
Adds tech-agnostic support for MF signaling by adding
MF sender and receiver applications as well as Dial
integration.
ASTERISK-29496-mf #do-not-close
Change-Id: I61962b359b8ec4cfd05df877ddf9f5b8f71927a4
Otherwise, the value 'false' was not found in the enumerated set of
the XML DTD for the XML attribute 'required' in the XML element
'parameter'. Therefore, DTD validation of the runtime XML failed.
ASTERISK-29790
Change-Id: Id13f230ad65a70dd8c2e3ae9ac85d1e841aed03e
In developer mode, use internal documentation as well.
This should produce no warnings. Fix yours!
In noisy mode, output all possible warnings of Doxygen.
This creates zillion of warnings. Double-check your current module!
Any warnings are in the file './doxygen.log'. Beside that, this change
avoids deprecated parameters because the configuration file for Doxygen
contains only those parameters which differ from the default. This
avoids the need to update the file on each run. Furthermore, it adds
AST_VECTOR to be expanded. Finally, the default name for that file is
Doxyfile. Therefore, let us use that!
ASTERISK-26991
ASTERISK-20259
Change-Id: I4129092a199d5e24c319a09cd088614b121015af
We know that passing a NULL or empty argument to
ast_channel_get_by_name() will never result in a matching channel and
will always result in an error being emitted, so just short-circuit
out in that case.
ASTERISK-28219 #close
Change-Id: I88eadc748e9c6996fc17467b0a05881bbfd00bce
res/res_rtp_asterisk.c: Adding 1 to rtpstart if it is deteremined
that rtpstart was configured to be an odd value. Also adding a loop
counter to prevent a possible infinite loop when looking for a free
port.
ASTERISK-27406
Change-Id: I90f07deef0716da4a30206e9f849458b2dbe346b
changed that when we recive a CANCEL that we set HANGUPCAUSE to AST_CAUSE_NORMAL_CLEARING
ASTERISK-28053
Reported by: roadkill
Change-Id: Ib653aec2282f55b59d87484391cc07c8e6612b89
Newer versions of spandsp did refactoring of code to add new features
like color FAXing. This refactoring broke backwards compatibility.
Add support for the new version while retaining support for 0.0.6.
ASTERISK-29729 #close
Change-Id: I3bd74550604ebcf0304528d647fa39abc62fbaa1
Since Doxygen 1.8.16, a special comment block is required. Otherwise
(pure C comment), the group command is ignored. Additionally, several
unbalanced group commands were fixed.
ASTERISK-29732
Change-Id: I4687857b9d56e6f44fd440b73af156691660202e
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
A backend's implementation of the realtime 'require' function may call
va_arg() and then fail, leaving the va_list in an undefined
state. Pass a copy of the va_list instead.
ASTERISK-29771 #close
Change-Id: I555565a72af84e96d49f62fe8cb66ba5a78461f4
Refactors generic functions used for email generation
into utils.c so that they can be used by multiple
modules, including app_voicemail and app_minivm,
to avoid code duplication.
ASTERISK-29715 #close
Change-Id: I1de0ed3483623e9599711129edc817c45ad237ee
This avoids a few long-name overflows, at the cost of less instructive
names in the case of C++ (specifically overloaded functions and class
methods). This in turn is offset against the fact that we're logging
the filename and line numbers in any case.
Change-Id: I54101a0bb5f8cb9ef63ec12c5e0d4c8edafff9ed
Signed-off-by: Jaco Kroon <jaco@uls.co.za>
In the AO2_ALLOC_OPT_LOCK_NOLOCK case the referenced obj
structure is freed, but is then referenced later if ref_log is
enabled. The change is to store the obj->priv_data.options value
locally and reference it instead of the value from the freed obj
ASTERISK-29730
Change-Id: I60cc5dc1f5a4330e7ad56976fc38a42de0ab6072
Local channels are made up of two pairs - the 1 and 2
sides. When a frame goes in one side, it comes out the
other. Back and forth. When both halves are in a
bridge this creates an infinite loop of frames.
This change makes it so that bridging no longer
allows both of these sides to exist in the same
bridge.
ASTERISK-29748
Change-Id: I29928b6de87cd9be996a77daccefd7c360fef651
Makes basic call progress tone detection available
in a tech-agnostic manner with the addition of the
ToneScan application. This can determine if the channel
has encountered a busy signal, SIT tones, dial tone,
modem, fax machine, etc. A few basic async progress
tone detect options are also added to the TONE_DETECT
function.
ASTERISK-29720 #close
Change-Id: Ia02437e0450473031e294798b8cb421fb8f24e90
Naveen Albert
Kevin Harwell
Steve Davies
Sean Bright
Florentin Mayer
Joshua C. Colp
Josh Soref
George Joseph
Frederic Van Espen
Mark Petersen
Alexander Traud
Mike Bradeen
Dustin Marquess
Jaco Kroon
Boris P. Korzun