If the 'rewrite_contact' option was enabled and a Contact header was received
which contained a '*' a crash would occur.
This change makes the res_pjsip_nat module ignore the Contact header if it
contains only a '*'.
(closes issue ASTERISK-23101)
Reported by: Matt Jordan
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@405019 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* The core external MWI resource provides for MWI message counts
persistence using sorcery. With sorcery, the user is able to configure
which sorcery wizzard backend to use if the default astdb is not desired.
* The core external MWI resoruce provides some debugging CLI commands
enabled by defining MWI_DEBUG_CLI.
The debugging CLI commands are:
"mwi delete all",
"mwi delete like <regex>",
"mwi delete mailbox <mailbox>",
"mwi list all",
"mwi list like <regex>",
"mwi show mailbox <mailbox>", and
"mwi update mailbox <mailbox> [<new> [<old>]]".
(closes issue AFS-43)
Review: https://reviewboard.asterisk.org/r/3061/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@404952 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Depending on which threading was loading the outbound registration it was
possible for the registration client to be allocated outside of a pj thread.
This change moves the creation inside the synchronous task where it is
guaranteed it will occur in a pj thread.
Reported by: Rob Thomas
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@404923 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Upon reload the module unconditionally "unloaded" the module (freeing memory
and setting pointers to NULL) and then when attempting a "load" if the config
file had not changed then nothing would be reinitialized.
By moving the "unload" to occur conditionally (reload only) after an attempted
configuration load, but before module "loading" alleviates the issue. The module
now loads/unloads/reloads correctly.
(closes issue ASTERISK-22871)
Reported by: Matteo
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@404859 65c4cc65-6c06-0410-ace0-fbb531ad65f3
An md5 hash is 32 bytes long. The char buffer must be at least 33 bytes to
avoid clobbering of the stack. This patch also fixes a potential clobbering
in test_utils.c.
Thanks to Andrew Nagy for reporting and testing this out in #asterisk-dev
Reported by: Andrew Nagy
Tested by: Andrew Nagy
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@404843 65c4cc65-6c06-0410-ace0-fbb531ad65f3
dahdi show channels output slices the callerid (which is dnid copied over on
PRI channels). If the channel naming structures look like:
'DAHDI/i1/1408409XXXX-6'
then the output slices 1408409XXXX down to 1408409XXX. This patch just opens
it up to 15 chars so you can see the whole thing.
(closes issue ASTERISK-22918)
Reported by: outtolunc
Patches:
svn_chan_dahdi.c.format12_15.diff.txt uploaded by outtolunc (license 5198)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@404786 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change moves outbound registration URI validation into the task executed
within a pjlib thread.
Reported by: Andrew Nagy
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@404725 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When calling REPLACE() with an empty replace-char argument, strcpy
is used to overwrite the the matching <find-char>. However as the
src and dest arguments to strcpy must not overlap, it causes other
parts of the string to be overwritten with adjacent characters and
the result is mangled. Patch replaces call to strcpy with memmove
and adds a test suite case for REPLACE.
(closes issue ASTERISK-22910)
Reported by: Gareth Palmer
Review: https://reviewboard.asterisk.org/r/3083/
Patches:
func_strings.patch uploaded by Gareth Palmer (license 5169)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@404676 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Channel creation in Asterisk is broken up into two steps: requesting and calling.
In some cases a channel may be requested but never called. This happens in the
ChanIsAvail dialplan application for determining if something is reachable or
not. The PJSIP channel driver did not take this situation into account and
attempted to end a session that was never called out on.
The code now checks the session state to determine if the session has been
called out on and if not terminates it instead of ending it.
(closes issue ASTERISK-23074)
Reported by: Kilburn
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@404652 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Hostnames specified in the 'match' field will be resolved and all addresses
returned. Each address will be added to the endpoint identifier for the
matching process.
Reported by: Rob Thomas
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@404613 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A deadlock can happen between a thread unloading or reloading the cel_pgsql
module and the core_event_dispatcher taskprocessor thread. Description of
what is happening:
Thread 1 (for example, a netconsole thread):
a "module reload cel_pgsql" is launched
the thread enter the "my_unload_module" function (cel_pgsql.c)
the thread acquire the write lock on psql_columns
the thread enter the "ast_event_unsubscribe" function (event.c)
the thread try to acquire the write lock on ast_event_subs[sub->type]
Thread 2 (core_event_dispatcher taskprocessor thread):
the taskprocessor pop a CEL event
the thread enter the "handle_event" function (event.c)
the thread acquire the read lock on ast_event_subs[sub->type]
the thread callback the "pgsql_log" function (cel_pgsql.c), since it's a subscriber of CEL events
the thread try to acquire a read lock on psql_columns
(closes issue ASTERISK-22854)
Reported by: Etienne Lessard
Patches:
cel_pgsql_fix_deadlock_event.patch uploaded by hexanol (license 6394)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@404605 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When applying configuration for outbound registrations the 'server_uri' and
'client_uri' fields were not validated. The code will now confirm that they
exist and that they contain parseable SIP URIs.
Reported by: Andrew Nagy
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@404592 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Asterisk does not support any of the transfer encodings specified in
HTTP/1.1, other than the default "identity" encoding.
According to RFC 2616:
A server which receives an entity-body with a transfer-coding it does
not understand SHOULD return 501 (Unimplemented), and close the
connection. A server MUST NOT send transfer-codings to an HTTP/1.0
client.
This patch adds the 501 Unimplemented response, instead of the hard work
of actually implementing other recordings.
This behavior is especially problematic for Node.js clients, which use
chunked encoding by default.
(closes issue ASTERISK-22486)
Review: https://reviewboard.asterisk.org/r/3092/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@404565 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When destroying a subscription we remove the serializer from its dialog
and decrease its reference count. Depending on which thread dropped the
subscription reference count to 0 it was possible for this to occur in
a thread where it is not possible.
(closes issue ASTERISK-22952)
Reported by: Matt Jordan
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@404553 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When wanting to pass *free as a function pointer, ast_free_ptr has to be used
instead of ast_free. This allows it to be compiled with MALLOC_DEBUG enabled.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@404531 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When we added support for specifying channel variables for an
origination, we didn't consider how that would interact with another
feature, namely specifying request parameters in a JSON request body.
The method of specifying channel variables (as a flat JSON object passed
in the JSON body) interferes with parsing parameters out of the request
body.
Unfortunately, fixing this would be a backward incompatible change. In
the interest of keeping the API sane and keeping our release schedule,
we're dropping the feature for specifying channel variables in the
origination request.
We will bring the feature back soon, as a backward compatible addition
to the API.
(closes issue ASTERISK-23051)
Review: https://reviewboard.asterisk.org/r/3088
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@404509 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Implements the following cli commands:
pjsip list aors
pjsip list auths
pjsip list channels
pjsip list contacts
pjsip list endpoints
pjsip show aor(s)
pjsip show auth(s)
pjsip show channels
pjsip show endpoint(s)
Also...
Minor modifications made to the AMI command implementations to facilitate
reuse.
New function ast_variable_list_sort added to config.c and config.h to implement
variable list sorting.
(issue ASTERISK-22610)
patches:
pjsip_cli_v2.patch uploaded by george.joseph (License 6322)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@404480 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When transferring to a dialplan extension that will not place any outbound
calls, the only control frames that the PJSIP REFER framehook will receive
are inconsequential (such as unhold or srcchange). As such, we shouldn't
allow for the reception of those types of frames prevent us from signaling
to the transferring party that the transfer has completed successfully once
voice frames are read.
Thanks to Jonathan Rose for pointing this out.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@404439 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The documentation for ARI already specifies that the device state resource when
used for subscribing for events is "deviceState", not "device_state". The code,
however, used "device_state"; although this was inconsistent as well in doxygen
comments in resource_applications.
Because the actual resource being subscribed to is /deviceStates/{device}/, it
makes sense for the resource type specifier to be deviceState.
Note that the key value in the events is still "device_state".
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@404437 65c4cc65-6c06-0410-ace0-fbb531ad65f3
AMI has received substantial updates over the past year. Not only has the
syntax been vastly improved and made consistent (which entails many event
changes), but the underlying things that those events convey have changed
substantially as well.
After some conversation in #asterisk-dev, it was agreed that this is a good
time to jump to 2.
At the same time, since ARI will most likely use semantic versioning, we
might as well use that for AMI as well. That also affords us greater meaning
for the AMI version.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@404421 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Added another NAT example to pjsip.conf.sample. We had a few mentions of NAT configuration throughout the sample, but I added another for a little bit more clarity.
Additionally many pjsip options were affected by the change to snake case, so I fixed any instances of those options in pjsip.conf.
I regenerated the config option list (at the bottom of the file) from a new xml config doc dump, so all the snake case changes should be reflected there, as well as any other changes to those options.
(issue ASTERISK-23004)
(closes issue ASTERISK-23004)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3086/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@404405 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Under normal conditions it is unlikely we will ever receive a response for a transaction
or dialog we don't know about but if any are received ignore them.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@404371 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The process for resending an INVITE with authentication involves restarting the UAC
session. We were incorrectly passing in that a new offer is being sent, causing the
SDP negotiation to get into a (technically speaking) funky state.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@404369 65c4cc65-6c06-0410-ace0-fbb531ad65f3
For the explanation, here is a copy-paste of the review board explanation:
Initially, it was discovered that performing an attended transfer of a
multiparty bridge with a PJSIP channel would cause a deadlock. A PBX thread
started a masquerade and reached the point where it was calling the fixup()
callback on the "original" channel. For chan_pjsip, this involves pushing a
synchronous task to the session's serializer. The problem was that a task ahead
of the fixup task was also attempting to perform a channel masquerade. However,
since masquerades are designed in a way to only allow for one to occur at a
time, the task ahead of the fixup could not continue until the masquerade
already in progress had completed. And of course, the masquerade in progress
could not complete until the task ahead of the fixup task had completed.
Deadlock.
The initial fix was to change the fixup task to be asynchronous. While this
prevented the deadlock from occurring, it had the frightful side effect of
potentially allowing for tasks in the session's serializer to operate on a
zombie channel.
Taking a step back from this particular deadlock, it became clear that the
problem was not really this one particular issue but that masquerades
themselves needed to be addressed. A PJSIP attended transfer operation calls
ast_channel_move(), which attempts to both set up and execute a masquerade. The
problem was that after it had set up the masquerade, the PBX thread had swooped
in and tried to actually perform the masquerade. Looking at changes that had
been made to Asterisk 12, it became clear that there never is any time now that
anyone ever wants to set up a masquerade and allow for the channel thread to
actually perform the masquerade. Everyone always is calling ast_channel_move(),
performs the masquerade itself before returning.
In this patch, I have removed all blocks of code from channel.c that will
attempt to perform a masquerade if ast_channel_masq() returns true. Now, there
is no distinction between setting up a masquerade and performing the
masquerade. It is one operation. The only remaining checks for
ast_channel_masq() and ast_channel_masqr() are in ast_hangup() since we do not
want to interrupt a masquerade by hanging up the channel. Instead, now
ast_hangup() will wait for a masquerade to complete before moving forward with
its operation.
The ast_channel_move() function has been modified to basically in-line the
logic that used to be in ast_channel_masquerade(). ast_channel_masquerade() has
been killed off for real. ast_channel_move() now has a lock associated with it
that is used to prevent any simultaneous moves from occurring at once. This
means there is no need to make sure that ast_channel_masq() or
ast_channel_masqr() are already set on a channel when ast_channel_move() is
called. It also means the channel container lock is not pulling double duty by
both keeping the container locked and preventing multiple masquerades from
occurring simultaneously.
The ast_do_masquerade() function has been renamed to do_channel_masquerade()
and is now internal to channel.c. The function now takes explicit arguments of
which channels are involved in the masquerade instead of a single channel.
While it probably is possible to do some further refactoring of this method, I
feel that I would be treading dangerously. Instead, all I did was change some
comments that no longer are true after this changeset.
The other more minor change introduced in this patch is to res_pjsip.c to make
ast_sip_push_task_synchronous() run the task in-place if we are already a SIP
servant thread. This is related to this patch because even when we isolate the
channel masquerade to only running in the SIP servant thread, we would still
deadlock when the fixup() callback is reached since we would essentially be
waiting forever for ourselves to finish before actually running the fixup. This
makes it so the fixup is run without having to push a task into a serializer at
all.
(closes issue ASTERISK-22936)
Reported by Jonathan Rose
Review: https://reviewboard.asterisk.org/r/3069
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@404356 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In fax_detect_framehook() a null pointer reference can occur where a
voice frame is processed but no dsp is attached to the fax detection
structure. The code block that rejects frames that detection cannot
be processed on is checking for dsp but falls through when it should
instead return, as this change implements.
(closes issue ASTERISK-22942)
Reported by: adomjan
Review: https://reviewboard.asterisk.org/r/3076/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@404352 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change is in preparation for external MWI support.
Removed code from the system for normal mailbox handling that appends
@default to the mailbox identifier if it does not have a context. The
only exception is the legacy hasvoicemail users.conf option. The legacy
option will only work for app_voicemail mailboxes. The system cannot make
any assumptions about the format of the mailbox identifer used by
app_voicemail.
chan_sip and chan_dahdi/sig_pri had the most changes because they both
tried to interpret the mailbox identifier. chan_sip just stored and
compared the two components. chan_dahdi actually used the box
information.
The ISDN MWI support configuration options had to be reworked because
chan_dahdi was parsing the box@context format to get the box number. As a
result the mwi_vm_boxes chan_dahdi.conf option was added and is documented
in the chan_dahdi.conf.sample file.
Review: https://reviewboard.asterisk.org/r/3072/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@404348 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When Asterisk is shut down, the astdb_atexit() function releases
(finalize) the previously initiated (prepared) SQL statements in
sqlite3. Another thread making a subsequent request can cause a
crash in sqlite3. This patch eliminates that issue by resetting
the statement pointer after it is released/cleared. The sqlite3
code detects the null pointer, and aborts the operation cleanly.
(closes issue AST-1265)
Reported by: Alexander Hömig
(closes issue ASTERISK-22350)
Reported by: Birger "WIMPy" Harzenetter
Review: https://reviewboard.asterisk.org/r/3078/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@404345 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Introduce new 'stopped' state for gk client and restart gk client
on failures
Remove ooh323 stack command lock as it is not need now.
(closes issue ASTERISK-21960)
Reported by: Dmitry Melekhov
Patches:
ASTERISK-21960.patch
ASTERISK-21960-stacklockup-2.patch
Tested by: Dmitry Melekhov
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@404320 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Moved channel locking into setsubstate so that a process can complete
working on a sub before another starts changing it. The existing code
was causing some Fracks with schedule deletion.
Removed multiple rtp cleanup. Now only cleansup up once, fixing ao2
object cleanup issues.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@404306 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When doing the rework of the CDR engine that pushed all of the logic into cdr.c
and made it respond to changes in channel state over Stasis, we knew that
accessing the CDR engine from the dialplan would be "slightly"
non-deterministic. Dialplan threads would be accessing CDRs while Stasis
threads would be updating the state of said CDRs - whereas in the past,
everything happened on the dialplan threads. Tests have shown that "slightly"
is in reality "very".
This patch synchronizes things by making the dialplan applications/functions
that manipulate CDRs do so over Stasis. ForkCDR, NoCDR, ResetCDR, CDR, and
CDR_PROP now all use Stasis to send their requests over to the CDR engine,
and synchronize on the channel Stasis topic via a subscription so that they
return their values/control to the dialplan at the appropriate time.
While going through this, the following changes were also made:
* DISA, which can reset the CDR when a user successfully authenticates, now
just uses the ResetCDR app to do this. This prevents having to duplicate
the same Stasis synchronization logic in that application.
* Answer no longer disables CDRs. It actually didn't work anyway - calling
DISABLE on the channel's CDR doesn't stop the CDR from getting the Answer
time - it just kills all CDRs on that channel, which isn't what the caller
would intend.
(closes issue ASTERISK-22884)
(closes issue ASTERISK-22886)
Review: https://reviewboard.asterisk.org/r/3057/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@404294 65c4cc65-6c06-0410-ace0-fbb531ad65f3
