Local channel optimization could cause DTMF digits to be duplicated.
Pending DTMF end events would be posted to a bridge when the local channel
optimizes out and is replaced by the channel further down the chain. When
the real digit ends, the channel would get another DTMF end posted to the
bridge.
A -- LocalA;1/n -- LocalA;2/n -- LocalB;1 -- LocalB;2 -- B
1) LocalA has the /n flag to prevent optimization.
2) B is sending DTMF to A through the local channel chain.
3) When LocalB optimizes out it can move B to the position of LocalB;1
4) Without this patch, when B swaps with LocalB;1 then LocalB;1 would
settle an owed DTMF end to the bridge toward LocalA;2.
5) When B finally ends its DTMF it sends the DTMF end down the chain.
6) Without this patch, A would hear the DTMF digit end when LocalB
optimizes out and when B ends the original digit.
ASTERISK-25582
Change-Id: I1bbd28b8b399c0fb54985a5747f330a4cd2aa251
Frame hooks can conceivably return a control frame in exchange for an
audio frame inside ast_write(). Those returned control frames were not
handled quite the same as if they were sent to ast_indicate(). Now it
doesn't matter if you use ast_write() to send an AST_FRAME_CONTROL to a
channel or ast_indicate().
ASTERISK-25582
Change-Id: I5775f41421aca2b510128198e9b827bf9169629b
The ast_sorcery_create, update and delete function have been refactored
to better deal with caches and errors.
The action is now called on all non-caching wizards first. If ANY succeed,
the action is called on all caching wizards and the observers are notified.
This way we don't put something in the cache (or update or delete) before
knowing the action was performed in at least 1 backend and we only call the
observers once even if there were multiple writable backends.
ast_sorcery_create was never adding to caches in the first place which
was preventing contacts from getting added to a memory_cache when they
were created. In turn this was causing memory_cache to emit errors if
the contact was deleted before being retrieved (which would have
populated the cache).
ASTERISK-25811 #close
Reported-by: Ross Beer
Change-Id: Id5596ce691685a79886e57b0865888458d6e7b46
There are a few cases where we're emitting notices or warnings
for things that really need neither, like a client retrying to subscribe
to mwi when they're not conifgured for it. They get a 404 so there's no
need for non-debug messages.
Change-Id: I05e38a7ff6c2f2521146f4be6a79731b9864e61f
Use the correct comparison function since we only care if the address
without the port is the same.
Change-Id: Ibf6c485f843a1be6dee58a47b33d81a7a8cbe3b0
Introduced realloaction of ast_str buf in sqlite3_escape functions in case
the returned buffer from threadstorage was actually too small.
Change-Id: I3c5eb43aaade93ee457943daddc651781954c445
The 'reload' mechanism actually involves closing the underlying
socket and calling the appropriate udp, tcp or tls start functions
again. Only outbound_registration, pubsub and session needed work
to reset the transport before sending requests to insure that the
pjsip transport didn't get pulled out from under them.
In my testing, no calls were dropped when a transport was changed
for any of the 3 transport types even if ip addresses or ports were
changed. To be on the safe side however, a new transport option was
added (allow_reload) which defaults to 'no'. Unless it's explicitly
set to 'yes' for a transport, changes to that transport will be ignored
on a reload of res_pjsip. This should preserve the current behavior.
Change-Id: I5e759850e25958117d4c02f62ceb7244d7ec9edf
Warnings and errors in the pjproject libraries are generally handled by
Asterisk. In many cases, Asterisk wouldn't even consider them to be warnings
or errors so the messages emitted by pjproject directly are either superfluous
or misleading. A good exampe of this are the level-0 errors pjproject emits
when it can't open a TCP/TLS socket to a client to send an OPTIONS. We don't
consider a failure to qualify a UDP client an "ERROR", why should a TCP/TLS
client be treated any differently?
A config file for res_pjproject has bene added (pjproject.conf) and a new
log_mappings object allows mapping pjproject levels to Asterisk levels
(or nothing). The defaults if no pjproject.conf file is found are the same
as those that were hard-coded into res_pjproject initially: 0,1 = LOG_ERROR,
2 = LOG_WARNING, 3,4,5 = LOG_DEBUG<level>
Change-Id: Iba7bb349c70397586889b8f45b8c3d6c6c8c3898
When Asterisk receives a 412 (Conditional Request Failed) response
it has to recreate publish session.
There is bug in res_pjsip_outbound_publish.c
The function sip_outbound_publish_client_alloc is called with wrong object
while processing 412 (Conditional Request Failed) response.
This patch fixes it.
ASTERISK-25229 #close
Change-Id: I3b62f2debf6bb1e5817cde7b13ea39ef2bf14359
The threadpool_auto_increment test fails infrequently for a couple of
reasons
* The threadpool listener was notified of fewer tasks being pushed than
were actually pushed
* The "was_empty" flag was set to an unexpected value.
The problem is that the test pushes three tasks into the threadpool.
Test expects the threadpool to essentially gather those three tasks, and
then distribute those to the threadpool threads. It also expects that as
the tasks are pushed in, the threadpool listener is alerted immediately
that the tasks have been pushed. In reality, a task can be distributed
to the threadpool threads quicker than expected, meaning that the
threadpool has already emptied by the time each subsequent task is
pushed. In addition, the internal threadpool queue can be delayed so
that the threadpool listener is not alerted that a task has been pushed
even after the task has been executed.
From the test's point of view, there's no way to be able to predict
exactly the order that task execution/listener notifications will occur,
and there is no way to know which listener notifications will indicate
that the threadpool was previously empty.
For this reason, the test has been updated to only check the things it
can check. It ensures that all tasks get executed, that the threads go
idle after the tasks are executed, and that the listener is told the
proper number of tasks that were pushed.
Change-Id: I7673120d74adad64ae6894594a606e102d9a1f2c
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
The return type of ast_cel_track_event() is not large enough to return all
64 potential bits of the event enable mask. Fortunately, the defined CEL
events do not really need all 64 bits and the return value is only used to
determine if the requested CEL event is enabled.
* Made the ast_cel_track_event() return 0 or 1 only so the return value
can fit inside an int type instead of zero or a truncated 64 bit non-zero
value.
Change-Id: I783d932320db11a95c7bf7636a72b6fe2566904c
res_statsd.export.in was missing the _va variations of the log
functions causing Asterisk to crash in res_pjsip if OPTIONAL_API
wasn't enabled.
ASTERISK-25727 #close
Reported-by: Gergely Dömsödi
Change-Id: I395729f9f51bdd33c5ca757f5f96ebedad74077b
A new command (pjsip export config_wizard primitives) has been added that
will export all the pjsip objects it created to the console or a file
suitable for reuse in a pjsip.conf file.
ASTERISK-24919 #close
Reported-by: Ray Crumrine
Change-Id: Ica2a5f494244b4f8345b0437b16d06aa0484452b
If the PJSIP_HEADER dialplan function adds a PAI or RPID header and send_rpid
or send_pai is set, res_pjsip_caller_id attemps to retrieve, parse and modify
the header added by the dialplan function. Since the header added by the
dialplan function is generic string, there are no virtual functions to parse
the uri and we get a segfault when we try. Since the modify, was really only
an overwrite, we now just delete the old header if it was type PJSIP_H_OTHER
and recreate it.
This raises a question for another time though: What should happen with
duplicate headers? Right now res_pjsip_header_funcs doesn't check for dups
so if it's session supplement is loaded after res_pjsip_caller_id's (or any
other module that adds headers), there'll be dups in the message.
ASTERISK-25337 #close
Change-Id: I5e296b52d30f106b822c0eb27c4c2b0e0f71c7fa
It is possible when processing a SIP REGISTER request to have two
threads end up creating contact_status structures in sorcery.
contact_status is created using a "find or create" function. If two
threads call into this at the same time, each thread will fail to find
an existing contact_status, and so both will end up creating a new
contact status.
During testing, we would see sporadic failures because the
PJSIP_CONTACT() dialplan function would operate on a different
contact_status than what had been updated by res_pjsip/pjsip_options.
The fix here is two-fold:
1) The "find or create" function for contact_status now has a lock
around the entire operation. This way, if two threads attempt the
operation simultaneously, the first to get there will create the object,
and the second will find the object created by the first thread.
2) res_sorcery_memory has had its create callback updated so that it
will not allow for objects with duplicate IDs to be created.
Change-Id: I55b1460ff1eb0af0a3697b82d7c2bac9f6af5b97
A problem arose when testing the AMI subscription listing actions where it
was possible for a subscription that had not been fully initialized to be
listed. This was problematic as the underlying listing code would crash.
This change makes it so the subscription tree is fully set up before it is
added to the list of subscriptions. This ensures that when the listing actions
get the subscription it is valid.
ASTERISK-25738 #close
Change-Id: Iace2b13641c31bbcc0d43a39f99aba1f340c0f48
listener() opens the same code block in two places (#if and #else). This
confuses some folding editors causing it to think that an extra code block
was opened. Folding in 'geany' causes all code after listener() to be
folded as if it were part of that procedure.
ASTERISK-24813 #close
Change-Id: I4b8c766e6c91e327dd445e8c18f8a6f268acd961
load_module was just too hairy with every step having to clean up all
previous steps on failure.
Some of the pjproject init calls have now been moved to a separate
load_pjsip function and the unload_pjsip function was enhanced to clean
up everything if an error happened at any stage of the load process.
In the process, a bunch of missing pj_shutdowns, serializer_pool_shutdowns
and ast_threadpool_shutdowns were also corrected.
Change-Id: I5eec711b437c35b56605ed99537ebbb30463b302
* heap-use-after-free happens when we free "cfg"
but then use "value" which refers to it
* A memory leak occurs because in some cases
it is not released "defaults"
ASTERISK-25721 #close
Reported by: Badalyan Vyacheslav
Tested by: Badalyan Vyacheslav
Change-Id: I3807d3f4726df6864430ec144cf6265d3f538469
Pjproject has deprecated pjsip_dlg_create_uas in 2.5 and replaced it with
pjsip_dlg_create_uas_and_inc_lock which, as the name implies, automatically
increments the lock on the returned dialog. To account for this, configure.ac
now detects the presence of pjsip_dlg_create_uas_and_inc_lock and res_pjsip.c
has an #ifdef HAVE_PJSIP_DLG_CREATE_UAS_AND_INC_LOCK to decide whether to use
the original call or the new one. If the new one was used, the ref count is
decremented before returning.
ASTERISK-25751 #close
Reported-by Josh Colp
Change-Id: I1be776b94761df03bd0693bc7795a75682615ca8
In older versions of the compiler was not sanitizes.
Compilers other than GCC can not support the Usan and TSAN
or have other options for *FLAGS.
ASTERISK-25767 #close
Reported by: Badalyan Vyacheslav
Tested by: Badalyan Vyacheslav
Change-Id: Iefce6608221fa87884b82ae3cb5649b7b1804916
USAN can be used together with other sanitizers.
Reported by: Badalyan Vyacheslav
Tested by: Badalyan Vyacheslav
Change-Id: I3bffa350d70965c3026651dba3a12414d0aaa45f
FD_SET contains a conditional statement to protect against buffer
overruns. The statement was overly complicated and prevented use
of the last array element of ast_fdset. We now just verify the fd
is less than ast_FDMAX.
Change-Id: I41895c0b497b052aef5bf49d75c817c48b326f40
When terminating the threads thrashing a sorcery memory cache each
would be told to stop and then we would wait on them. During at
least one thrashing test this was problematic due to the specific
usage pattern in use. It would take some time for termination of the
thread to occur.
This would occur due to contention between the threads retrieving
and the threads updating the cache. As the retrieving threads are
given priority it may be some time before the updating threads
are able to proceed.
This change makes it so all threads are told to stop and then each
are joined to ensure they stop. This way all the threads should
stop at around the same time instead of waiting for one to stop,
the next to stop, then the next, and so on. As a result of this
the execution time for each thrash test is much closer to their
expected value than previously seen as well.
Change-Id: I04a53470b0ea4170b8819180b0bd7475f3642827
Attempting to load a transport from realtime was forcing asterisk into an
infinite recursion loop. The first thing transport_apply did was to do a
sorcery retrieve by id for an existing transport of the same name. For files,
this just returns the previous object from res_sorcery_config's internal
container, if any. For realtime, the res_sourcery_realtime driver looks in the
database and finds the existing row but now it has to rehydrate it into a
sorcery object which means calling... transport_apply. And so it goes.
The main issue with loading from realtime (apart from the loop) was that
transport stores structures and pointers directly in the ast_sip_transport
structure instead of the separate ast_transport_state structure. This patch
separates those items into the ast_sip_transport_state structure. The pattern
is roughly the same as res_pjsip_outbound_registration.
Although all current usages of ast_sip_transport and ast_sip_transport_state
were modified to use the new ast_sip_get_transport_state API, the original
items are left in ast_sip_transport and kept updated to maintain ABI
compatability for third-party modules. They are marked as deprecated and
noted that they're now in ast_sip_transport_state.
ASTERISK-25606 #close
Reported-by: Martin Moučka
Change-Id: Ic7a836ea8e786e8def51fe3f8cce855ea54f5f19
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.
