Adding format_name even to the end of ast_codec caused issued with
binary codec modules because the pointer would be garbage in asterisk
when they registered. So, the ast_codec structure was reverted and an
internal_ast_codec structure was created just for use in codec.c. A new
internal-only API was also added (__ast_codec_register_with_format) so
that codec_builtin could register codecs with the format_name in a
separate parameter rather than in the ast_codec structure.
ASTERISK-26144 #close
Reported-by: Alexei Gradinari
Change-Id: I6df1b08f6a6ae089db23adfe1ebc8636330265ba
gcc 6.1.1 caught a few more issues.
Made sure the unit tests still pass for the func_env and stdtime
issues.
ASTERISK-26157 #close
Change-Id: I6664d8f34a45bc1481d2a854481c7878b0c1cf8e
A non-existent constraint was being referenced in the upgrade script.
This patch corrects the problem by removing the reference.
In addition, the head of the alembic branch referred to a non-existent
revision. This has been fixed by referring to the proper revision.
This patch fixes another realtime problem as well. Our Alembic scripts
store booleans as yes or no values. However, Sorcery tries to insert
"true" or "false" instead. This patch introduces a new boolean type that
translates to "yes" or "no" instead.
ASTERISK-26128 #close
Change-Id: I51574736a881189de695a824883a18d66a52dcef
The internal HTTP/WebSocket server supports both TCP and TLS, which can be
activated separately via the file http.conf. The source code intends to re-use
the TCP parameter 'bindaddr' for TLS, even if 'tlsbindaddr' is not specified
explicitly. This did not work because of a typo. This change resolves this typo.
ASTERISK-26126 #close
Change-Id: I5efb0409ae12044dfb3495b6b97b6d40a8c9c51f
With CLI "core show settings", simply the parameter maxfiles of the file
asterisk.conf was shown. If that parameter was not set, nothing was displayed
although the environment might have set a default number itself. Or if maxfiles
were not granted (completely), still maxfiles was shown. Now, the maximum number
of possible file descriptors in the environment is shown.
ASTERISK-26097
Change-Id: I2df5c58863b5007b34b77adbe28b885dfcdf7e0b
With menuselect "DEBUG_FD_LEAKS" and CLI "core show fd", both the maximum max
and current max of possible file descriptors were shown. Both show the same
value always. Not to confuse users, just the current maximum is shown now.
ASTERISK-26097
Change-Id: I49cf7952d73aec9e3f6a88942842c39be18380fa
CEL wrongly assumed that a channel would only have a single dial
event on it. This is incorrect. Particularly in a queue each
call attempt to a member will result in a dial event, adding
a new dial status in CEL without removing the old one. This
would cause the container to grow with only one dial status
being removed when the channel went away. The other dial status
entries would remain leaking memory.
This change fixes the memory leak by ensuring that only one dial
status will only ever exist for each channel.
The behavior during the scenario where multiple events are received
has also been improved. For failure cases the first failure will
be the dial status. If an answer dial status is received, though,
it will take priority and the dial status for the channel will be
answer.
Memory usage has also been decreased by storing the minimal
amount of information and the code has been cleaned up slightly.
ASTERISK-25262 #close
Change-Id: I5944eb923db17b6a0faa7317ff6abc9307c009fe
Scenario: Caller blonde transfer
Bob calls Charlie who answers.
Bob puts Charlie on hold and calls Alice.
Before Alice answers, Bob transfers Charlie to Alice.
Charlie's channel triggers an assert because he gets an "ANSWERED"
event even though he never dialed anything. With recent changes to dial
events, this is now a valid scenario so the assert needed to be removed.
ASTERISK-26103 #close
Change-Id: I2679b517b696e7952ab7fb29403df9140e7d1de2
Stasis subscriptions and message routers create taskprocessors to process
the event messages. API calls are needed to be able to set the congestion
levels of these taskprocessors for selected subscriptions and message
routers.
* Updated CDR, CEL, and manager's stasis subscription congestion levels
based upon stress testing. Increased the congestion levels to reduce the
potential for bursty call setup/teardown activity from triggering the
taskprocessor overload alert. CDRs in particular need an extra high
congestion level because they can take awhile to process the stasis
messages.
ASTERISK-26088
Reported by: Richard Mudgett
Change-Id: Id0a716394b4eee746dd158acc63d703902450244
Sorcery creates taskprocessors for object types to process object observer
callbacks. An API call is needed to be able to set the congestion levels
of these taskprocessors for selected object types.
* Updated PJSIP's contact and contact_status sorcery object type observer
default congestion levels based upon stress testing. Increased the
congestion levels to reduce the potential for bursty register/unregister
and subscribe/unsubscribe activity from triggering the taskprocessor
overload alert.
ASTERISK-26088
Reported by: Richard Mudgett
Change-Id: I4542e83b556f0714009bfeff89505c801f1218c6
When taskprocessors get backed up, there is a good chance that we are
being overloaded and need to defer adding new work to the system.
* Implemented a high/low water alert mechanism for modules to check if the
system is being overloaded and take appropriate action. When a
taskprocessor is created it has default congestion levels set. A
taskprocessor can later have those congestion levels altered for specific
needs if stress testing shows that the taskprocessor is a symptom of
overloading or needs to handle bursty activity without triggering an
overload alert.
* Add CLI "core show taskprocessor" low/high water columns.
* Fixed __allocate_taskprocessor() to not use RAII_VAR(). RAII_VAR() was
never a good thing to use when creating a taskprocessor because of the
nature of how its references needed to be cleaned up on a partial
creation.
* Made res_pjsip's distributor check if the taskprocessor overload alert
is active before placing a message representing brand new work onto a
distributor serializer.
ASTERISK-26088
Reported by: Richard Mudgett
Change-Id: I182f1be603529cd665958661c4c05ff9901825fa
POSIX defines signal.h. sys/signal.h should not be used as it is
c-library internal header which may or may not exist. Notably with
musl it generates warning of being incorrect.
Change-Id: Ia56b0aa1d84b5c590114867b1b384a624f39a6fc
Icelandic has some weird grammar rules when dealing with dates and
numbers. There are different genders used depending on which number
you're dealing with, and only a handful of numbers do change depending
on the gender. There is also an implied gender in several cases.
This patch was originally written for asterisk 1.6, and has been in use
for several years without crashes. I cleaned it up a bit and rewrote
what was necessary for Asterisk 13.
The functions were copied from other similar languages and modified
where appropriate. If i recall correctly, the German and Danish
functions were used as a base.
ASTERISK-26087
Reported by: Örn Arnarson
Tested by: Örn Arnarson
Change-Id: Ib7d8bd7b0fede5767921ed821315b5b508c0e665
In several internal library projects, the files are archived with the help of
'ar cr'. Only the projects editline and the Objective Open H.323 stack
implementation in C (ooh323c) use 'ar cru' instead. Recently, some platforms
changed the default parameters of AR which creates "/usr/bin/ar: `u' modifier
ignored since `D' is the default (see `U')". For consistency and to avoid this
message all projects use 'ar cr' now.
ASTERISK-26091 #close
Change-Id: I710a9b1c01c1b5a1931a646098c044c8161ead40
Added a new channel variable FORWARDERNAME which indicates which
channel was responsible for a forwarding requests received on dial attempt.
Fixed a bug in the app_queue: FORWARD_CONTEXT is not used.
ASTERISK-26059 #close
Change-Id: I34e93e8c1b5e17776a77b319703c48c8ca48e7b2
If you create a local channel and don't specify an originator channel
to take capabilities from, we automatically add all audio formats to
the new channel's capabilities. When we try to make the channel
compatible with another, the "best format" functions pick the best
format available, which in this case will be slin192. While this is
great for preserving quality, it's the worst for performance and
overkill for the vast majority of applications.
In the absense of any other information, adding all formats is the
correct thing to do and it's not always possible to supply an
originator so a new parameter 'formats' has been added to the channel
create/originate functions. It's just a comma separated list of formats
to make availalble for the channel. Example: "ulaw,slin,slin16".
'formats' and 'originator' are mutually exclusive.
To facilitate determination of format names, the format name has been
added to "core show codecs".
ASTERISK-26070 #close
Change-Id: I091b23ecd41c1b4128d85028209772ee139f604b
Resolver state is not part of res_search API. This fixes
compilation error:
dns.c:261:8: error: too many arguments to function 'res_search'
ret = res_search(&dns_state,
Change-Id: Ia600a58557040df83f744da3dde23225293845a5
POSIX defines poll.h, sys/poll.h should not be used at is c-library
internal header which may or may not exist. Notable in musl it
generates warning of being incorrect. And add explict include of
sys/cdefs.h where needed.
Change-Id: I142930df53fe7585a06b854b6faddc5301e024be
The stringfields refactor to allow adding stringfields to the end of a
structure (f6f4cf459f) exposed some
incomplete cleanup code by some stringfield users.
The most noticeable leaker is the logging system where there is a leak for
every log message generated.
ASTERISK-26078 #close
Reported by: Etienne Lessard
Patches:
jira_asterisk_26078_v13.patch (license #5621) patch uploaded
by Richard Mudgett
Change-Id: If6a08b31336b492c3de6f9dfd07c447f8d5a8782
Dial events up to this point have come in two flavors
* A Dial event with no status to indicate that dialing has begun
* A Dial event with a status to indicate that dialing has ended
With this change, Dial events have been expanded to also give
intermediate events, such as "RINGING", "PROCEEDING", and "PROGRESS".
This is especially useful for ARI dialing, as it gives the application
writer the opportunity to place a channel into an early bridge when
early media is detected.
AMI handles these in-progress dial events by sending a new event called
"DialState" that simply indicates that dial state has changed but has
not ended. ARI never distinguished between DialBegin and DialEnd, so no
change was made to the event itself.
Another change here relates to dial forwards. A forward-related event
was previously only sent when a channel was successfully able to forward
a call to a new channel. With this set of changes, if forwarding is
blocked, we send a Dial event with a forwarding destination but no
forwarding channel, since we were prevented from creating one. This is
again useful for ARI since application writers can now handle call
forward attempts from within their own application.
ASTERISK-25925 #close
Reported by Mark Michelson
Change-Id: I42cbec7730d84640a434d143a0d172a740995543
As res_pjsip_nat rewrites contact's address, only the last Via header
can contain the source address of registered endpoint.
Also Call-Id header may contain the source address of registered
endpoint.
Added "via_addr", "via_port", "call_id" to contact.
Added new fields ViaAddress, CallID to AMI event ContactStatus.
ASTERISK-26011
Change-Id: I36bcc0bf422b3e0623680152d80486aeafe4c576
worker_start checked for ZOMBIE status without holding a lock. All
other read/write of worker status are performed with a lock, so this
check should do the same.
ASTERISK-25777 #close
Change-Id: I5e33685a5c26fdb300851989a3b82be8c4e03781
Invisible bridges function the same as normal bridges, but they have the
following restrictions:
* They never show up in CLI, AMI, or ARI queries.
* They do not have Stasis messages published about them.
Invisible bridges' main use is for when use of the bridging system is
desired, but the bridge should not be known to users of the Asterisk
system.
ASTERISK-25925
Change-Id: I804a209d3181d7c54e3d61a60eb462e7ce0e3670
Scenario:
Local fax -> Asterisk w/ firewall -> Provider -> Remote fax
* Local fax starts rtp call to remote fax
* Remote fax starts t38 call back to local fax.
* Local fax sends t38 no-signal to Asterisk before sending an OK.
* udptl processes the frame and increments the expected sequence number.
* chan_sip drops the frame because the call isn't up so nothing goes out
the external interface to open the port for incoming packets.
* Local fax sends OK and Asterisk sends OK to the remote fax.
* Remote fax sends t38 packets which are dropped by the firewall.
* Local fax re-sends t38 no-signal with the same sequence number.
* udptl drops the frame because it thinks it's a dup.
* Still no outgoing packets to open the firewall.
* t38 negotiation fails.
The patch drops frames t38 received before udptl sequence processing
when the call hasn't been answered yet. The second no-signal frame
is then seen as new and is relayed out the external interface which
opens the port and allows negotiation to continue.
ASTERISK-26034 #close
Change-Id: I11744b39748bd2ecbbe8ea84cdb4f3c5943c5af9
George Joseph
Joshua Colp
Mark Michelson
zuul
Alexei Gradinari
Alexander Traud
Richard Mudgett
Timo Teräs
Örn Arnarson
Jaco Kroon
Niklas Larsson
Corey Farrell