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The menuselect conflict between app_voicemail and res_mwi_external
makes it hard to package 1 version of Asterisk. There no actual
build dependencies between the 2 so moving this check to runtime
seems like a better solution.
The ast_vm_register and ast_vm_greeter_register functions in app.c
were modified to return AST_MODULE_LOAD_DECLINE instead of -1 if there
is already a voicemail module registered. The modules' load_module
functions were then modified to return DECLINE instead of -1 to the
loader. Since -1 is interpreted by the loader as AST_MODULE_LOAD_FAILURE,
the modules were incorrectly causing Asterisk to stop so this needed
to be cleaned up anyway.
Now you can build both and use modules.conf to decide which voicemail
implementation to load.
The default menuselect options still build app_voicemail and not
res_mwi_external but if both ARE built, res_mwi_external will load
first and become the voicemail provider unless modules.conf rules
prevent it. This is noted in CHANGES.
Change-Id: I7d98d4e8a3b87b8df9e51c2608f0da6ddfb89247
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If a caller hangs up before dial is executed within an AGI then the AGI
has likely eaten all queued frames before executing the dial in DeadAGI
mode. With the caller hung up and no pending frames from the caller's
read queue, dial would not know that the call has hung up until a called
channel answers. It is rather annoying to whoever just answered the
non-existent call.
Dial should not continue execution in DeadAGI mode, hangup handlers, or
the h exten.
* Added a check early in dial to abort dialing if the caller has hungup.
ASTERISK-25307 #close
Reported by: David Cunningham
Change-Id: Icd1bc0764726ef8c809f76743ca008d0f102f418
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Update documentation to reflect that maximum_number_of_words
has functionality inconsistent with the variable name (and inconsistent
with prior documentation.)
Update documentation for silence_threshold, which previously implied
that it was measuring time, rather than noise averages in the sample.
Update the comments in amd.conf.sample.
ASTERISK-25639 #close
Change-Id: I4b1451e5dc9cb3cb06d59b6ab872f5275ba79093
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If a call enters on a queue and the members on that queue are updated in
realtime (ex: using mysql inserting a new agent) the queue members are
never refreshed and the call will stay in the queue until other event occurs.
This happens only if this is the first call of the queue and there is no
agents servicing.
This patch prevent this issue, ensuring realtime members are updated if
there is one call in the queue and no available agents
ASTERISK-25442 #close
Change-Id: If1e036d013a5c1d8b0bf60d71d48fe98694a8682
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The default value was never set for audio_buffers, causing bad
audio quality. This ensures the default is always set.
ASTERISK-25569 #close
Change-Id: I2d2ee3e644120b0f9f6ea6ab9286d7d590942a44
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This option adds the ability to specify a timeout, in seconds, for a
participant in a ConfBridge. When the user's timeout has been reached,
the user is ejected from the conference with the CONFBRIDGE_RESULT
channel variable set to "TIMEOUT".
The rationale for this change is that there have been times where we
have seen channels get "stuck" in ConfBridge because a network issue
results in a SIP BYE not being received by Asterisk. While these
channels can be hung up manually via CLI/AMI/ARI, adding some sort of
automatic cleanup of the channels is a nice feature to have.
ASTERISK-25549 #close
Reported by Mark Michelson
Change-Id: I2996b6c5e16a3dda27595f8352abad0bda9c2d98
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commit aae45acbd (Mark Michelson 2015-04-15 10:38:02 -0500 6525)
refer ASTERISK-24958
above commit removed ast_channel_lock(qe->chan);
but failed to remove corresponding ast_channel_unlock(qe->chan);
ASTERISK-25561 #close
Reported Alec Davis
Change-Id: Ie05f4e2d08912606178bf1fded57cc022c7a2e1a
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increased." into 13
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ASTERISK-25533 #close
Change-Id: Ie1a9d1a6511b3f1a56b93d04475fbf8a4e40010a
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Change-Id: Iba57efadf6c0b822e762c7a001bc89611d98afd7
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* 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
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Replace inlined code with update_connected_line_from_peer().
ASTERISK-25423
Reported by: John Hardin
Change-Id: I33bbd033596fcb0208d41d8970369b4e87b806f3
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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
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* 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
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Replace inlined code with update_connected_line_from_peer().
ASTERISK-25423
Reported by: John Hardin
Change-Id: Ia14f18def417645cd7fb453e1bdac682630a5091
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Change-Id: Ice1884a94315d3cb7e3bbd47a9fba76a27276c54
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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
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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
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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
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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
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* 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
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Change-Id: I7294e13d27875851c2f4ef6818adba507509d224
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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
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Last time I checked, it's "Sangoma", not "Samgoma". Thanks to Brian
(GameGamer43) for pointing that out.
Change-Id: I43d7b196f6d7a2b2517b84915e3a8dfbc2894106
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This patch updates a variety of Makefiles in Asterisk's build system to
remove .gcda and .gcno files when 'make clean' is executed. These files
are generated when '--enable-coverage' is passed to the Asterisk
configure script.
Change-Id: Ib70b41eea2ee2908885bff02e80faf9f40c84602
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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
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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
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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
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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>
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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
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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
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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
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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
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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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434449 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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)
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Merged revisions 434285 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434286 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Resolve compile errors caused by r433863 by fixing the
documentation xml to comply with the schema.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433888 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Resolve compile errors caused by r433839 by included the missing
header file, pbx.h.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433863 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433839 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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This patch fixes several warnings caught by clang - in this case, usage of the
abs function on non-integer values. This patch uses labs and fabs, as
appropriate, in the various affected files.
Review: https://reviewboard.asterisk.org/r/4525
ASTERISK-24917
Reported by: dkdegroot
patches:
rb4525.patch submitted by dkdegroot (License 6600)
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Merged revisions 433749 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433750 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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This patch fixes the -Wunused-value -Wunused-variable -Wunused-const-variable
errors caught by clang. Specifically:
* apps/app_queue.c: removed unused qpm_cmd_usage[], qum_cmd_usage[],
qsmp_cmd_usage[]
* cel/cel_sqlite3_custom.c: removed unused name[] = "cel_sqlite3_custom"
* channels/chan_pjsip.c: removed unused desc[] = "PJSIP Channel"
* codecs/gsm/src/gsm_create.c: removed unused ident[] = "$Header$"
* funcs/func_env.c:729: Fixed ast_str_append_substr.
* main/editline/np/strlcat.c: removed unused rcsid variable
* main/editline/np/strlcpy.c: removed unused rcsid variable
* main/security_events.c: removed unused TIMESTAMP_STR_LEN
* utils/conf2ael.c: removed unused cfextension_states
* utils/extconf.c: removed unused cfextension_states
Review: https://reviewboard.asterisk.org/r/4526
ASTERISK-24917
Reported by: dkdegroot
patches:
rb4526.patch submitted by dkdegroot (License 6600)
........
Merged revisions 433693 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433694 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Clang will treat ((a == b)) as a warning, as it reasonably expects that the
developer may have intended to write (a == b) or ((a = b)). This patch cleans
up all instances where equality, not assignment, was intended between two
parantheses.
Review: https://reviewboard.asterisk.org/r/4531/
ASTERISK-24917
Repoted by: dkdegroot
patches:
rb4531.patch submitted by dkdegroot (License 6600)
........
Merged revisions 433687 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433688 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Attempting to execute DTMF in a confbridge while file playback (prompt,
announcement, etc) is occurring is not allowed. You have to wait until
the sound file has completed before entering DTMF. This patch fixes it
so that app_confbridge now monitors for dtmf key presses during menu
driven file playback. If a key is pressed playback stops and it executes
the matched menu option.
ASTERISK-24864 #close
Reported by: Steve Pitts
Review: https://reviewboard.asterisk.org/r/4510/
........
Merged revisions 433445 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433446 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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In versiong 5.5, OpenBSD went to 64-bit time values. This requires a cast to
(long) when printing members of certain time structs.
Review: https://reviewboard.asterisk.org/r/4507
ASTERISK-24879 #close
Reported by: snuffy
Tested by: snuffy
patches:
openbsd-time64.diff uploaded by snuffy (License 5024)
........
Merged revisions 433268 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433269 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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