Added code to allow the StatsD dialplan application to
send data to the server specified in statsd.conf.
ASTERISK-25419
Change-Id: I400db2f37c6ddf61515ff5a019646e36dcd0f922
Added code to accept user input and validate it before
allowing it to be sent to the StatsD server.
ASTERISK-25419
Reported By: Ashley Sanders
Change-Id: I55c7ce44326a68ad6c5c1514b9575ac50f25bbc3
In app_queue added value Paused Reason on QueueMemberStatus when a member
on queue is paused and the reason was set.
ASTERISK-25480 #close
Reporte by: Rodrigo Ramírez Norambuena
Change-Id: Ia5db503482f50764c15e2020196c785f59d4a68e
Wrote the skeleton framework for the Asterisk StatsD dialplan
application. This includes a load function, unload function, a
callback for execution, and XML documentation.
ASTERISK-25419
Reported By: Ashley Sanders
Change-Id: I9597730e134c6e82c8a55ef4d5334b62dd473363
* When a call is answered and the outgoing channel name has changed then
force a connected line update because the channel is no longer the same.
The channel was masqueraded into by another channel. This is usually
because of a call pickup.
Note: Forwarded calls are handled in a controlled manner so the original
channel name is replaced with the forwarded channel.
ASTERISK-25423 #close
Reported by: John Hardin
Change-Id: Ie275ea9e99c092ad369db23e0feb08c44498c172
While the 'A' option is playing the announcement file allow the caller and
peer to exchange COLP update frames.
ASTERISK-25423
Reported by: John Hardin
Change-Id: Iac6cf89b56d26452c6bb88e9363622bbf23895f9
* When a call is answered and the outgoing channel name has changed then
force a connected line update because the channel is no longer the same.
The channel was masqueraded into by another channel. This is usually
because of a call pickup.
Note: Forwarded calls are handled in a controlled manner so the original
channel name is replaced with the forwarded channel.
ASTERISK-25423
Reported by: John Hardin
Change-Id: I2e01f7a698fbbc8c26344a59c2be40c6cd98b00c
Page uses the async method of dialing with the dial API. When a call gets
forwarded there is no calling channel available. If the predial handler
was set then the calling channel could not be put into auto-service
for the forwarded call because it doesn't exist. A crash is the result.
* Moved the callee predial parameter string processing to before the
string is passed to the dial API rather than having the dial API do it.
There are a few benefits do doing this. The first is the predial
parameter string processing doesn't need to be done for each channel
called by the dial API. The second is in async mode and the forwarded
channel is to have the predial handler executed on it then the
non-existent calling channel does not need to be present to process the
predial parameter string.
* Don't start auto-service on a non-existent calling channel to execute
the predial handler when the dial API is in async mode and forwarding a
call.
ASTERISK-25384 #close
Reported by: Chet Stevens
Change-Id: If53892b286d29f6cf955e2545b03dcffa2610981
The RECORDED_FILE variable is empty unless a '%d' is specified in the filename.
This patch makes it so the variable is always set to the filename.
ASTERISK-25410 #close
Change-Id: I4ec826d8eb582ae2ad184e717be8668b74d37653
When a queued caller transfers an agent to another extension sometimes the
raised AgentComplete event has a reason of "caller" and sometimes "transfer".
Since a transfer has taken place this should always be transfer. This occurs
because sometimes the stasis hangup event arrives before the transfer event
thus writing a different reason out.
With this patch, when a hangup event is received during a transfer it will
check to see if the channel that is hanging up is part of a transfer. If so
it will return and let the subsequently received transfer event handler take
care of the cleanup.
ASTERISK-25399 #close
Change-Id: Ic63c49bd9a5ed463ea7a032fd2ea3d63bc81a50d
During some transfer scenarios involving queues Asterisk would sometimes
crash when trying to obtain a channel snapshot (could happen on caller or
member channels). This occurred because the underlying channel had already
disappeared when trying to obtain the latest snapshot.
This patch adds a reference to both the member and caller channels that
extends to the lifetime of the queue'd call, thus making sure the channels
will always exist when retrieving the latest snapshots.
ASTERISK-25185 #close
Reported by: Etienne Lessard
Change-Id: Ic397fa68fb4ff35fbc378e745da9246a7b552128
Setting the 'paused' and 'ringinuse' options on a queue member using the
dialplan function QUEUE_MEMBER did not behave the same way as the
equivalent dialplan applications or AMI actions.
* Made queue_function_mem_write() call the set_member_paused() and
set_member_value() for the 'paused' and 'ringinuse' options respectively.
A beneficial side effect is that the queue name is now optional and sets
the value in all queues the interface is a member.
* Update QUEUE_MEMBER XML documentation.
* Fix error checking in QUEUE_MEMBER() write.
ASTERISK-25215 #close
Reported by: Lorne Gaetz
Change-Id: I3a016be8dc94d63a9cc155295ff9c9afa5f707cb
* Extract set_queue_member_pause() from set_member_paused() for simpler
and more consistent code.
* Extract set_queue_member_ringinuse() from
set_member_ringinuse_help_members() for simpler code.
Change-Id: Iecc1f4119c63347341d7ea6b65f5fc4963706306
Last time I checked, it's "Sangoma", not "Samgoma". Thanks to Brian
(GameGamer43) for pointing that out.
Change-Id: I43d7b196f6d7a2b2517b84915e3a8dfbc2894106
Currently when requesting a channel the native formats of the
calling channel are provided to the core for usage when dialing
the outbound channel. This occurs without holding the channel lock
or keeping a reference to the formats. This is problematic as
the channel driver may end up changing the formats during this time.
In the case of chan_sip this happens when an SDP negotiation
completes.
This change makes it so app_dial keeps a reference to the native
formats of the calling channel which guarantees that they will
remain valid for the period of time needed.
ASTERISK-25172 #close
Change-Id: I2f0a67bd0d5d14c3bdbaae552b4b1613a283f0db
The voicemail.conf mailbox key/value pair is defined as:
<mailbox>=[<password>[,<full-name>[,<email>[,<pager>[,<options>]]]]]
Where all fields in the value including the field values are optional.
Since the parsing code for the mailbox key/value pair is sloppy, this
patch tightens the parsing for the directory information.
* Renamed the 'pos' and 'bufptr' variables to 'name' and 'options'
respectively in search_directory_sub(). Those names make more sense.
* Made sure that search_directory_sub() is dealing with the voicemail.conf
mailbox options field if it even exists when looking for the 'hidefromdir'
and 'alias' options.
* Fix crash if a voicemail.conf mailbox is just
<mailbox>=<password>,<name> when the 'a' option is used. If there were no
fields after the name then the 'options' pointer was not checked for NULL.
* Fix users.conf alias processing if the 'a' option is used. The wrong
variable was used.
ASTERISK-25087 #close
Reported by: Chet Stevens
Change-Id: I86052ea77307beddddba5279824d39dc0d593374
Although ast_context_find, ast_context_find_or_create and
ast_context_destroy perform locking of the contexts table,
any context pointer can become invalid at any time that the
contexts table is unlocked. This change adds locking around
all complete operations involving these functions.
Places where ast_context_find was followed by ast_context_destroy
have been replaced with calls ast_context_destroy_by_name.
ASTERISK-25094 #close
Reported by: Corey Farrell
Change-Id: I1866b6787730c9c4f3f836b6133ffe9c820734fa
If a channel hangs up while an audio file is playing, there's
no need to clutter up the logs with a warning so suppress it
if ast_check_hangup returns true.
Also, change warning to debug/2 in file.c if writing a frame
fails. Same reasoning.
Change-Id: I2e66191af3c5b6e951c98e8f1c3fe3cf2cf7ed89
Reported-by: George Joseph <george.joseph@fairview5.com>
Tested-by: George Joseph <george.joseph@fairview5.com>
When completing voicemail playback of a message in the 'INBOX', the
message gets moved to the 'Old' messages folder. Without this patch, if
the 'Old' folder is already at its set limit, then the 'INBOX' message will
simply be deleted. With this patch, the flag to delete the message will be
removed if the save_to_folder function indicates that the message could
not be moved due to a full folder.
ASTERISK-25082 #close
Reported by: Jonathan Rose
Review: https://gerrit.asterisk.org/#/c/448/
Change-Id: I2be440a09f42e2d06d50975c40d1ad7f836ecb3f
This patch fixes EXITWITHTIMEOUT queue_log entry to always come with 3
parameters: position, original position and waiting time.
ASTERISK-25038 #close
Reported by: Etienne Lessard
Change-Id: I0c62045922e26bee2125e93aee1dee17eee79618
* The REF_DEBUG compiler flag no longer has any effect on code that uses
Astobj2. It is used to determine if reference debugging is enabled by
default. Reference debugging can be enabled or disabled in asterisk.conf.
* Caller information is provided in logger errors for ao2 bad magic numbers.
* Optimizes AO2 by merging internal functions with the public counterpart.
This was possible now that we no longer require a dual ABI.
ASTERISK-24974 #close
Reported by: Corey Farrell
Change-Id: Icf3552721fe999365ba8a8cf00a965aa6b897cc1
Confbridge dynamic profiles did not have a default profile unless you
explicitly used Set(CONFBRIDGE(bridge,template)=default_bridge). If a
template was not set prior to the bridge being created then some
options were left with no default values set. This patch makes it so
the default templates are set to the default bridge and user profiles.
ASTERISK-24749 #close
Reported by: philippebolduc
Change-Id: I1bd6e94b38701ac2112d842db68de63d46f60e0a
A potential problem that can arise is the following:
* Bob's phone is programmed to automatically forward to Carol.
* Carol's phone is programmed to automatically forward to Bob.
* Alice calls Bob.
If left unchecked, this results in an endless loops of call forwards
that would eventually result in some sort of fiery crash.
Asterisk's method of solving this issue was to track which interfaces
had been dialed. If a destination were dialed a second time, then
the attempt to call that destination would fail since a loop was
detected.
The problem with this method is that call forwarding has evolved. Some
SIP phones allow for a user to manually forward an incoming call to an
ad-hoc destination. This can mean that:
* There are legitimate use cases where a device may be dialed multiple
times, or
* There can be human error when forwarding calls.
This change removes the old method of detecting forwarding loops in
favor of keeping a count of the number of destinations a channel has
dialed on a particular branch of a call. If the number exceeds the
set number of max forwards, then the call fails. This approach has
the following advantages over the old:
* It is much simpler.
* It can detect loops involving local channels.
* It is user configurable.
The only disadvantage it has is that in the case where there is a
legitimate forwarding loop present, it takes longer to detect it.
However, the forwarding loop is still properly detected and the
call is cleaned up as it should be.
Address review feedback on gerrit.
* Correct "mfgium" to "Digium"
* Decrement max forwards by one in the case where allocation of the
max forwards datastore is required.
* Remove irrelevant code change from pjsip_global_headers.c
ASTERISK-24958 #close
Change-Id: Ia7e4b7cd3bccfbd34d9a859838356931bba56c23
This new macro allows a single line to add all additional
sources to a module. This helps prevent modules from
missing steps, and makes future changes easier since
they can be made in a single place.
ASTERISK-24960 #close
Reported by: Corey Farrell
Change-Id: I38f12d8b72c5e7bb37a879b2fb51761a2855eb4b
Git does not support the ability to replace a token with a version
string during check-in. While it does have support for replacing a
token on clone, this is somewhat sub-optimal: the token is replaced
with the object hash, which is not particularly easy for human
consumption. What's more, in practice, the source file version was often
not terribly useful. Generally, when triaging bugs, the overall version
of Asterisk is far more useful than an individual SVN version of a file. As a
result, this patch removes Asterisk's support for showing source file
versions.
Specifically, it does the following:
* Rename ASTERISK_FILE_VERSION macro to ASTERISK_REGISTER_FILE, and
remove passing the version in with the macro. Other facilities
than 'core show file version' make use of the file names, such as
setting a debug level only on a specific file. As such, the act of
registering source files with the Asterisk core still has use. The
macro rename now reflects the new macro purpose.
* main/asterisk:
- Refactor the file_version structure to reflect that it no longer
tracks a version field.
- Remove the "core show file version" CLI command. Without the file
version, it is no longer useful.
- Remove the ast_file_version_find function. The file version is no
longer tracked.
- Rename ast_register_file_version/ast_unregister_file_version to
ast_register_file/ast_unregister_file, respectively.
* main/manager: Remove value from the Version key of the ModuleCheck
Action. The actual key itself has not been removed, as doing so would
absolutely constitute a backwards incompatible change. However, since
the file version is no longer tracked, there is no need to attempt to
include it in the Version key.
* UPGRADE: Add notes for:
- Modification to the ModuleCheck AMI Action
- Removal of the "core show file version" CLI command
Change-Id: I6cf0ff280e1668bf4957dc21f32a5ff43444a40e
Although it only occurred once, a crash occurred when a queue attempted to
evaluate a queue penalty rule that appeared to have already been destroyed.
In many locations in app_queue, a test is done to see if qe->pr is NULL;
however, when we dispose of a queue's penalty rules, we don't set the pointer
to NULL after free'ing it. This patch does that to prevent any dangling
pointers from lingering on the queue object.
Review: https://reviewboard.asterisk.org/r/4522
ASTERISK-23319 #close
Reported by: Vadim
patches:
rb4552.patch submitted by Stefan Engström (License 6691)
........
Merged revisions 434448 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 434449 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434450 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch fixes several warnings pointed out by the clang compiler.
* chan_pjsip: Removed check for data->text, as it will always be non-NULL.
* app_minivm: Fixed evaluation of etemplate->locale, which will always
evaluate to 'true'. This patch changes the evaluation to use
ast_strlen_zero.
* app_queue:
- Fixed evaluation of qe->parent->monfmt, which always evaluates to
true. Instead, we just check to see if the dereferenced pointer
evaluates to true.
- Fixed evaluation of mem->state_interface, wrapping it with a call to
ast_strlen_zero.
* res_smdi: Wrapped search_msg->mesg_desk_term with calls to ast_strlen_zero.
Review: https://reviewboard.asterisk.org/r/4541
ASTERISK-24917
Reported by: dkdegroot
patches:
rb4541.patch submitted by dkdegroot (License 6600)
........
Merged revisions 434285 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 434286 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434287 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When an error occurs while writing to a web socket, the web socket is
disconnected and the event is logged. A side-effect of this, however, is that
any application on the other side waiting for a response from Stasis is left
hanging indefinitely (as there is no mechanism presently available for
notifying interested parties about web socket error states in Stasis).
To remedy this scenario, this patch introduces a new channel variable:
STASISSTATUS.
The possible values for STASISSTATUS are:
SUCCESS - The channel has exited Stasis without any failures
FAILED - Something caused Stasis to croak. Some (not all) possible
reasons for this:
- The app registry is not instantiated;
- The app requested is not registered;
- The app requested is not active;
- Stasis couldn't send a start message
ASTERISK-24802
Reported By: Kevin Harwell
Review: https://reviewboard.asterisk.org/r/4519/
........
Merged revisions 433839 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433845 65c4cc65-6c06-0410-ace0-fbb531ad65f3
