It was possible for a queue member that is a member of at least 2 or more
queues to receive mulitiple calls at the same time. This happened because
of a race between when a member was being rung and when the device state
notified the other queue(s) member object of the state change.
This patch makes it so when a queue member is being rung it gets added to
a global pool of queue members. If that same member is tried again, e.g.
from another queue, and it is found to already exist in the pending member
container then it will not ring that member.
ASTERISK-16115 #close
Change-Id: I546dd474776d158c2b6be44205353dee5bac7e48
You cannot reference the passed in features struct after calling
ast_bridge_impart(). Even if the call fails.
Change-Id: I902b88ba0d5d39520e670fb635078a367268ea21
This module is used as part of testsuite tests to confirm
stuff works. I'm accordingly marking it as core as it is
required by those tests.
Change-Id: I558e7af7679b22b8ed641d7dd37ee4ca35b11e88
The test_voicemail_notify_endl test checks the end-of-line
characters of an email message to confirm that they are consistent.
The test wrongfully assumed that reading from the email message
into a buffer will always result in more than 1 character being
read. This is incorrect. If only 1 character was read the test
would go outside of the buffer and access other memory causing
a crash.
The test now checks to ensure that 2 or more characters are read
in ensuring the test stays within the buffer.
ASTERISK-25874 #close
Change-Id: Ic2c89cea6e90f2c0bc2d8138306ebbffd4f8b710
If try to move message to Cust1 (number 5)
the function 'save_to_folder' tries to create Greeting folder instead of Cust1.
This patch fixed it by setting GREETINGS_FOLDER = -1
ASTERISK-24927 #close
Change-Id: I03d1a761894bcc2d130ec9b003bbcddc28e25c51
Sometimes uw-imap function 'mail_fetchbody' returns huge len
which then pass to uw-imap function 'rfc822_base64'.
uw-imap tries to allocate huge memory and abort() on fail.
This patch check the len.
If the len more than max size (128 Mbytes) log error.
This patch also set variables len, newlen to avoid uninizialezed len.
This patch also check pointer returned by rfc822_base64.
ASTERISK-25899 #close
Change-Id: I4a0e7d655f11abef6a5224e2169df6d5c1f1caca
When using app_echo via WebRTC with VP8 video the video would appear
only after a few minutes, because there would be nothing to request
a full reference frame.
This fixes the problem in both ways:
- echos any VIDUPDATE frames received on the channel
- sends one such frame when first video frame is to be forwarded
This makes the echo work with Firefox and Chrome WebRTC implementation.
ASTERISK-25867 #close
Change-Id: I73bda87bf7532ee8bfb28d917045a21034908c1e
The configuration unsigned integer option handler sets flags for the
parser as if the option should be a signed integer (PARSE_INT32),
leading to errors on "out of range" values. Fix flags (PARSE_UINT32).
A fix to res_pjsip is also present which stops invalid flags from
being passed when registering sorcery object fields for qualify
status.
ASTERISK-25612 #close
Change-Id: I96b539336275e0e72a8e8033487d2c3344debd3e
This prevents pbx_core from hanging up the channel if the app isn't
registered.
ASTERISK-25846 #close
Change-Id: I63216a61f30706d5362bc0906b50b6f0544aebce
Channel masquerading had a conflict with autochannel locking.
When locking autochannel->channel, the channel is fetched from the
autochannel and then locked. During the fetch, the autochannel -- which
has no locks itself -- can be modified by someone who owns the channel
lock. That means that the value of autochan->channel cannot be trusted
until you hold the lock.
In practice, this caused problems with Local channels getting
masqueraded away while the ChanSpy attempted to get info from that
channel. The old channel which was about to get removed got locked, but
the new (replaced) channel got unlocked (no-op). Because the replaced
channel was now locked (and would never get unlocked), it couldn't get
removed from the channel list in a timely manner, and would now cause
deadlocks when iterating over the channel list.
This change checks the autochannel after locking the channel for changes
to the autochannel. If the channel had been changed, the lock is
reobtained on the new channel.
In theory it seems possible that after this fix, the lock attempt on the
old (wrong) channel can be on an already destroyed lock, maybe causing
a crash. But that hasn't been observed in the wild and is harder induce
than the current deadlock.
Thanks go to Filip Frank for suggesting a fix similar to this and
especially to IRC user hexanol for pointing out why this deadlock was
possible and testing this fix. And to Richard for catching my rookie
while loop mistake ;)
ASTERISK-25321 #close
Change-Id: I293ae0014e531cd0e675c3f02d1d118a98683def
Fix calculate of average time for talktime is wrong when is completed the
first call beacuse the time for talked would be that call.
ASTERISK-25800 #close
Change-Id: I94f79028935913cd9174b090b52bb300b91b9492
A user cannot set new bridge options after the conference is created by
the first user. Attempting to do so is documented as undefined behavior.
This patch ensures that the bridge profile options used are from the
conference and not what a subsequent user may have tried to set.
Change-Id: I1b6383eba654679e5739d5a8de98199cf074a266
* changes:
app_confbridge: Add ability to get the muted conference state.
app_confbridge.c: Update CONFBRIDGE and CONFBRIDGE_INFO documentation.
app_confbridge: Make non-admin users join a muted conference muted.
* Added CONFBRIDGE_INFO(muted,) for querying the muted conference state.
* Added Muted header to AMI ConfbridgeListRooms action response list
events to indicate the muted conference state.
* Added Muted column to CLI "confbridge list" output to indicate the muted
conference state and made the locked column a yes/no value instead of a
locked/unlocked value.
ASTERISK-20987
Reported by: hristo
Change-Id: I4076bd8ea1c23a3afd4f5833e9291b49a0c448b1
Member lastcall time is updated later than member status. There was chance to
check wrapuptime for available member with wrong (old) lastcall time.
New boolean flag "in_call" is set to true right before connecting call, and
reset to false after update of lastcall time. Members with "in_call" set to true
are treat as unavailable.
ASTERISK-19820 #close
Change-Id: I1923230cf9859ee51563a8ed420a0628b4d2e500
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
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
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
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
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
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
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
* 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
* 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
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
Last time I checked, it's "Sangoma", not "Samgoma". Thanks to Brian
(GameGamer43) for pointing that out.
Change-Id: I43d7b196f6d7a2b2517b84915e3a8dfbc2894106
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
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
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
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
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434286 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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433839 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
Valgrind found some memory leaks associated with ast_sockaddr_resolve().
Most of the leaks had already been fixed by earlier memory leak hunt
patches. This patch performs an audit of ast_sockaddr_resolve() and found
one more.
* Fix ast_sockaddr_resolve() memory leak in
apps/app_externalivr.c:app_exec().
* Made main/netsock2.c:ast_sockaddr_resolve() always set the addrs
parameter for safety so the pointer will never be uninitialized on return.
The same goes for res/res_pjsip_acl.c:extract_contact_addr().
* Made functions that call ast_sockaddr_resolve() with RAII_VAR()
controlling the addrs variable use ast_free instead of ast_free_ptr to
provide better MALLOC_DEBUG information.
Review: https://reviewboard.asterisk.org/r/4509/
........
Merged revisions 433056 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433057 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch corrects the documentation for the AMD application. Specifically:
* It documents the maximum_word_length option, which limits the maximum allowed
length of a single utterance.
* It clarifies the AMDCAUSE values MAXWORDS and MAXWORDLENGTH. MAXWORDLENGTH
was documented as MAXWORDS, while MAXWORDS was undocumented.
Thanks to the issue reporter, Frank DiGennaro, for pointing out the issues.
ASTERISK-19470 #close
Reported by: Frank DiGennaro
........
Merged revisions 432918 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432920 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When an IMAP backend is in use and greetings are set to be used, but aren't
present for a user in their IMAP folder, Asterisk will crash. This occurs
due to the mailstream being set to the 'greetings' folder and being left
in that particular state, regardless of the success/failure of the attempt
to access the folder the mailstream points to. Later access of the mailstream
assumes that it points to the 'INBOX' (or some other folder), resulting in
either a crash (if the greetings folder didn't exist and the mailstream is
invalid) or an inability to read messages from the 'INBOX' folder.
This patch restores the mailstream to its correct state after accessing the
greetings. This fixes the crash, and sets the mailstream to the state that
VoiceMailMain expects.
Note that while ASTERISK-23390 also contained a patch for this issue, the
patch on ASTERISK-24786 is the one being merged here.
Review: https://reviewboard.asterisk.org/r/4459/
ASTERISK-23390 #close
Reported by: Ben Smithurst
ASTERISK-24786 #close
Reported by: Graham Barnett
Tested by: Graham Barnett
patches:
app_voicemail.c.patch.SIGSEGV3rev2 uploaded by Graham Barnett (License 6685)
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When using IMAP voicemail with FreePBX, you will often get ERROR messages
complaining about not being able to find a mailbox. This is due to how FreePBX
handles voicemail mailboxes. Unfortunately, app_voicemail has to consider this
a configuration error, as in any other system it would be indicative of
someone misconfiguring their system.
Regardless, a misconfiguration is a WARNING, and not an ERROR. This patch
demotes the message so that system administrators can hopefully reduce some
of the noise in their log files.
Note that in the original patch this was made into a NOTICE, but that's a
too forgiving.
ASTERISK-24790 #close
Reported by: Graham Barnett
patches:
app_voicemail.c.patch_noise uploaded by Graham Barnett (License 6685)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432099 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When interfacing with Microsoft Exchange, custom headers will be returned as
all lower case. Currently, the IMAP header code will fail to parse the returned
custom headers, as it will be performing a case sensitive comparison. This can
cause playback of messages to fail, as needed information - such as origtime -
will not be present.
This patch updates app_voicemail's header parsing code to perform a case
insensitive lookup for the requested custom headers. Since the headers are
specific to Asterisk, e.g., 'x-asterisk-vm-orig-time', and headers should be
unique in an IMAP message, this should cause no issues with other systems.
ASTERISK-24787 #close
Reported by: Graham Barnett
patches:
app_voicemail.c.patch_MSExchange uploaded by Graham Barnett (License 6685)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432013 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The Test Event for MIXMONITOR_END - which signals that a MixMonitor has
completed - technically fired before the filestream was closed. If a test
used this to trigger a condition to verify that the file was written, it
could result in a race condition where the file size would not be what the
test expected.
Luckily, no tests were using this (although they should have been). Since the
test event needed to be moved after the point where the MixMonitor autochan has
been destroyed, the test event no longer emits the channel name. Luckily,
nothing needs it.
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There are three CLI commands to stop and restart Asterisk each.
1) core stop/restart now - Hangup all calls and stop or restart Asterisk.
New channels are prevented while the shutdown request is pending.
2) core stop/restart gracefully - Stop or restart Asterisk when there are
no calls remaining in the system. New channels are prevented while the
shutdown request is pending.
3) core stop/restart when convenient - Stop or restart Asterisk when there
are no calls in the system. New calls are not prevented while the
shutdown request is pending.
ARI has made stopping/restarting Asterisk more problematic. While a
shutdown request is pending it is desirable to continue to process ARI
HTTP requests for current calls. To handle the current calls while a
shutdown request is pending, a new committed to shutdown phase is needed
so ARI applications can deal with the calls until the system is fully
committed to shutdown.
* Added a new shutdown committed phase so ARI applications can deal with
calls until the final committed to shutdown phase is reached.
* Made refuse new HTTP requests when the system has reached the final
system shutdown phase. Starting anything while the system is actively
releasing resources and unloading modules is not a good thing.
* Split the bridging framework shutdown to not cleanup the global bridging
containers when shutting down in a hurry. This is similar to how other
modules prevent crashes on rapid system shutdown.
* Moved ast_begin_shutdown(), ast_cancel_shutdown(), and
ast_shutting_down(). You should not have to include channel.h just to
access these system functions.
ASTERISK-24752 #close
Reported by: Matthew Jordan
Review: https://reviewboard.asterisk.org/r/4399/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431692 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When the app_agent_pool module initially loads there is a race condition
between the thread loading agents.conf and the device state internal
processing thread. If the device state internal processing thread handles
the agent creation state updates before the thread that loaded agents.conf
registers the device state provider callback then the cached agent state
is "Invalid". When a consumer module like app_queue asks for the agent state
it gets the cached "Invalid" state instead of the real state from the provider.
* Moved loading the agents.conf configuration to the last thing setup by
app_agent_pool in load_module(). Now the device state provider callback
is registered before the config is loaded so the agent creation state
updates are guaranteed to get the initial device state.
* Removed some now redundant config cleanup on error in load_config().
* Added lock protection when accessing the device state in
agent_pvt_devstate_get() and eliminated the RAII_VAR() usage.
ASTERISK-24737 #close
Reported by: Steve Pitts
Review: https://reviewboard.asterisk.org/r/4390/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431492 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Starting and stopping conference recording more than once causes the
recording channels to be leaked. For v13 the channels also show up in the
CLI "core show channels" output.
* Reworked and simplified the recording channel code to use
ast_bridge_impart() instead of managing the recording thread in the
ConfBridge code. The recording channel's ref handling easily falls into
place and other off nominal code paths get handled better as a result.
ASTERISK-24719 #close
Reported by: John Bigelow
Review: https://reviewboard.asterisk.org/r/4368/
Review: https://reviewboard.asterisk.org/r/4369/
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When issuing a 'confbridge list XXXX' CLI command, the resulting output no
longer displays the menu associated with a ConfBridge participant.
The issue was caused by ASTERISK-22760. When that patch was done, it removed
the copying of the menu name associated with the user from the actual user
profile.
This patch fixes the issue by copying the menu name over to the user profile
when the menu hooks are applied to the user. Since that function now does a
little bit more than just apply the hooks, the name of the function has been
changed to cover the copying of the menu name over as well.
In addition, there is a disparity between the menu name length as it is stored
on the conf_menu structure and the confbridge_user structure; this patch makes
the lengths match so that a strcpy can be used.
Review: https://reviewboard.asterisk.org/r/4372/
ASTERISK-24723 #close
Reported by: Steve Pitts
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431134 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Channel names should be different from other channels in the system while
the channel exists.
* Use a sequence number for CBRec channels instead of a random number
because the same random number could be picked again for the next CBRec
channel.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431052 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The MixMonitor m() option allows a recording to be pushed to a specific
voicemail mailbox. If the message is delivered to the mailbox's INBOX, however,
no MWI notification is currently raised.
This patch corrects the issue by properly calling notify_new_state from the
msg_create_from_file function. This will cause MWI to be triggered if the
message was placed in the mailbox's INBOX.
ASTERISK-24709 #close
Reported by: Gareth Palmer
patches:
app_voicemail-430919.patch uploaded by Gareth Palmer (License 5169)
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The Dial application has some interesting options with the mid-call Macro (M)
and GoSub (U) options. If the MACRO_RESULT/GOSUB_RESULT returns specific
values, the Dial application will take some action upon the channels involved
in the dial operation (such as hanging up a particular party, etc.) The Dial
application ensures that a Stasis message is published in the event that
MACRO_RESULT/GOSUB_RESULT returns a value that kills the dial operation, so
that there is a corresponding DialEnd event published in AMI/ARI for the
DialBegin event that preceeded it.
A bug exists where that same DialEnd event will be published on Stasis even if
the value returned in MACRO_RESULT/GOSUB_RESULT is not one that the Dial
application cares about. This causes two DialEnd events to be published - one
with the MACRO_RESULT/GOSUB_RESULT and another with "ANSWERED" - which is all
sorts of wrong.
This patch fixes the bug by ensuring that we only publish a DialEnd message to
Stasis if the Dial application's mid-call Macro/GoSub returns something that
Dial cares about.
Review: https://reviewboard.asterisk.org/r/4336
ASTERISK-24682 #close
Reported by: Matt Jordan
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430842 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When using ODBC or IMAP storage, temporary files created on the file system
must be disposed of using the DISPOSE macro. The DELETE macro will map to a
deletion function for the backend storage, but does not clean up any local
files created as a result of the operation.
When using voicemail with the operator and review options enabled, pressing
0 to enter the menu, followed by 1 to save the message, followed by any
other DTMF press to delete the message, will result in the temporary file
lingering on the file system.
This patch properly calls DISPOSE after the DELETE. This causes the local
file to be disposed of.
ASTERISK-24288 #close
Reported by: LEI FU
patches:
voicemail_odbc_review_fix.diff uploaded by LEI FU (License 6640)
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v11: If a channel redirect to a macro exten of a macro that is active
happens, the redirect location doesn't get executed. Instead the original
macro location is restored and gets reexecuted.
v13: An additional effect happens if a parked call times out to an
extension in the macro that parked the call then the macro is reexecuted
instead of the expected park return location.
* Made not restore the macro calling location on an
AST_SOFTHANGUP_ASYNCGOTO.
* Increased the locked channel range when setting up the macro execution
environment to cover things that should be done while the channel is
locked.
* Removed unnecessary NULL tests before calling ast_free() in
_macro_exec().
ASTERISK-23850 #close
Reported by: Andrew Nagy
Review: https://reviewboard.asterisk.org/r/4292/
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* Made the following AMI actions use list API calls for consistency:
Agents
BridgeInfo
BridgeList
BridgeTechnologyList
ConfbridgeLIst
ConfbridgeLIstRooms
CoreShowChannels
DAHDIShowChannels
DBGet
DeviceStateList
ExtensionStateList
FAXSessions
Hangup
IAXpeerlist
IAXpeers
IAXregistry
MeetmeList
MeetmeListRooms
MWIGet
ParkedCalls
Parkinglots
PJSIPShowEndpoint
PJSIPShowEndpoints
PJSIPShowRegistrationsInbound
PJSIPShowRegistrationsOutbound
PJSIPShowResourceLists
PJSIPShowSubscriptionsInbound
PJSIPShowSubscriptionsOutbound
PresenceStateList
PRIShowSpans
QueueStatus
QueueSummary
ShowDialPlan
SIPpeers
SIPpeerstatus
SIPshowregistry
SKINNYdevices
SKINNYlines
Status
VoicemailUsersList
* Incremented the AMI version to 2.7.0.
* Changed astman_send_listack() to not use the listflag parameter and
always set the value to "Start" so the start capitalization is consistent.
i.e., The FAXSessions used "Start" while the rest of the system used
"start". The corresponding complete event always used "Complete".
* Fixed ami_show_resource_lists() "PJSIPShowResourceLists" to output the
AMI ActionID for all of its list events.
* Fixed off-nominal AMI protocol error in manager_bridge_info(),
manager_parking_status_single_lot(), and
manager_parking_status_all_lots(). Use of astman_send_error() after
responding to the original AMI action request violates the action response
pattern by sending two responses.
* Fixed minor protocol error in action_getconfig() when no requested
categories are found. Each line needs to be formatted as "Header: text".
* Fixed off-nominal memory leak in manager_build_parked_call_string().
* Eliminated unnecessary use of RAII_VAR() in ami_subscription_detail().
ASTERISK-24049 #close
Reported by: Jonathan Rose
Review: https://reviewboard.asterisk.org/r/4315/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430434 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In r413586 (1.8) various casts were added to silence gcc 4.10 warnings.
Those fixes included things like:
-out += sprintf(out, "%%%02X", (unsigned char) *ptr);
+out += sprintf(out, "%%%02X", (unsigned) *ptr);
That works for low ascii characters, but for the high range that yields
e.g. FFFFFFC3 when C3 is expected.
This changeset:
- fixes those casts to use the 'hh' unsigned char modifier instead
- consistently uses %02x instead of %2.2x (or other non-standard usage)
- adds a few 'h' modifiers in various places
- fixes a 'replcaes' typo
- dev/urandon typo (in 13+ patch)
Review: https://reviewboard.asterisk.org/r/4263/
ASTERISK-24619 #close
Reported by: Stefan27 (on IRC)
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The UW IMAP library is instrinsically not thread-safe, and relies upon higher
level applications to guarantee thread safety. For the most part, this is
provided by the vms object, which provides locking for individual streams.
Unfortunately, this is not sufficient for calls to mail_open which create the
IMAP stream. mail_open can, on some systems, call into a UW IMAP specific
function for determining the address of a system based on a hostname,
ip_nametoaddr.
In the ip6_unix implementation of this function, static variables are used
to hold parsing buffers. This can cause a crash if multiple threads attempt
to convert a hostname to an address at the same time. Locking on a single
mail stream is not sufficient to prevent simultaneous access to these static
variables.
In the IMAP library, this function can be called from the mail_open and
imap_status functions. As the imap_status function is not used by
app_voicemail, locking on access to mail_open is sufficient to prevent
any mangling of the buffers.
Review: https://reviewboard.asterisk.org/r/4188/
ASTERISK-24516 #close
Reported by: David Duncan Ross Palmer
Tested by: David Duncan Ross Palmer
patches:
ASTERISK-24516.diff uploaded by David Duncan Ross Palmer (License 6660)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@428865 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Prior to this patch, all Stasis subscriptions would receive a dedicated
thread for servicing published messages. In contrast, prior to r400178
(see review https://reviewboard.asterisk.org/r/2881/), the subscriptions
shared a thread pool. It was discovered during some initial work on Stasis
that, for a low subscription count with high message throughput, the
threadpool was not as performant as simply having a dedicated thread per
subscriber.
For situations where a subscriber receives a substantial number of messages
and is always present, the model of having a dedicated thread per subscriber
makes sense. While we still have plenty of subscriptions that would follow
this model, e.g., AMI, CDRs, CEL, etc., there are plenty that also fall into
the following two categories:
* Large number of subscriptions, specifically those tied to endpoints/peers.
* Low number of messages. Some subscriptions exist specifically to coordinate
a single message - the subscription is created, a message is published, the
delivery is synchronized, and the subscription is destroyed.
In both of the latter two cases, creating a dedicated thread is wasteful (and
in the case of a large number of peers/endpoints, harmful). In those cases,
having shared delivery threads is far more performant.
This patch adds the ability of a subscriber to Stasis to choose whether or not
their messages are dispatched on a dedicated thread or on a threadpool. The
threadpool is configurable through stasis.conf.
Review: https://reviewboard.asterisk.org/r/4193
ASTERISK-24533 #close
Reported by: xrobau
Tested by: xrobau
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Kevin Harwell
DarkS
Richard Mudgett
Joshua Colp
ibercom
Alexei Gradinari
Jacek Konieczny
Gianluca Merlo
Andrew Nagy
Walter Doekes
Rodrigo Ramírez Norambuena
George Joseph
Martin Tomec
Dade Brandon
Carlos Oliva
Corey Farrell
Mark Michelson
Alec Davis
Alexander Traud
Matt Jordan
Jonathan Rose
Ivan Poddubny
Ashley Sanders