kkm [Wed, 25 Jan 2017 04:31:38 +0000 (20:31 -0800)]
app_queue: Fix queues randomly disappearing on reload
With 500+ queues and a reload every minute, a random queue disappears
upon reload. The cause is mususe of the 'dead' flag. Namely, all queues
were marked dead up front, and then "resurrected" by dropping this flag
for those found in the configuration. But a queue marked dead can be
removed also when control leaves the app entry point on a PBX thread.
With this change, the queue is marked only not found, and at the end of
reload only the queues that are still not found are actually marked as
dead, so the dead flag is never reset, and set only on positively dead
queues.
Joshua Colp [Tue, 24 Jan 2017 21:39:39 +0000 (21:39 +0000)]
res_pjsip_endpoint_identifier_ip: Ensure error defaults to 0.
When configuring a match using a netmask the error variable was
not defaulting to 0. For some people this would cause the code
to think an error occurred when adding the match when in reality
it added perfectly fine.
Richard Mudgett [Tue, 10 Jan 2017 19:11:20 +0000 (13:11 -0600)]
main/app.c: Memory corruption from early format destruction.
* make_silence() created a malloced silence slin frame without adding a
slin format ref. When the frame is destroyed it will unref the slin
format that never had a ref added. Memory corruption is expected to
follow.
* Simplified and fixed counting the number of samples in a frame list for
make_silence().
* Eliminated an unnecessary RAII_VAR associated with the make_silence()
frame.
Mark Michelson [Mon, 23 Jan 2017 22:18:18 +0000 (16:18 -0600)]
Free endpoint ACLs when destroying PJSIP endpoints.
If endpoint ACLs were specified, they were not being freed
when endpoints were destroyed. On systems with realtime endpoints, this
could add up quickly since each DB lookup would allocate the ACL without
freeing it.
George Joseph [Mon, 23 Jan 2017 15:10:50 +0000 (08:10 -0700)]
pjproject_bundled: Fix setting max log level
An earlier attempt to prevent pjsua from spitting out an extra 6795
lines of debug output every time the testsuite called it was also
turning off the ability for asterisk to output debug info when it
needed to. This patch reverts the earlier fix and instead adds
a pjproject patch that sets the startup log level to 1 for pjsua
pjsystest and the pjsua python binding. This is an asterisk-only
patch that does not affect pjproject functionality and will not be
submitted upstream.
Joshua Colp [Mon, 23 Jan 2017 16:08:37 +0000 (16:08 +0000)]
res_pjsip_endpoint_identifier_ip: Read settings before resolving.
An option has been added, srv_lookups, which controls whether
SRV lookups are performed on the provided match hosts or not.
It was possible for this option to be applied after resolution
had already happened.
This change makes it so hosts are stored away, settings are read
and applied, and then resolution is done. This ensures that no
matter the ordering the srv_lookups option is in effect.
Richard Mudgett [Sun, 22 Jan 2017 23:25:57 +0000 (17:25 -0600)]
LISTFILTER: Remove outdated ERROR message.
Feeding LISTFILTER an empty variable results in an invalid ERROR message.
Earlier changes made the message useless because we can no longer tell if
the variable is empty or does not exist. It is valid to try to remove a
value from an empty list just as it is valid to try to remove a value that
is not in a non-empty list.
* Removed the outdated ERROR message.
* Added more test cases to the LISTFILTER unit test.
Richard Mudgett [Thu, 5 Jan 2017 21:11:12 +0000 (15:11 -0600)]
res_pjsip_pubsub.c: Fix AMI event list counts.
Fix the AMI PJSIPShowSubscriptionsInbound, PJSIPShowSubscriptionsOutbound,
and PJSIPShowResourceLists actions event counts. The reported counts may
not necessarily be accurate depending on what happens.
The subscriptions count would be wrong if Asterisk ever has outbound
subscriptions.
The resource list count could be wrong if a list were added or removed
during the AMI action being processed.
George Joseph [Fri, 13 Jan 2017 17:03:15 +0000 (10:03 -0700)]
debug_utilities: Create ast_loggrabber
ast_loggrabber gathers log files from customizable search patterns,
optionally converts POSIX timestamps to a readable format and
tarballs the results.
It was possible for a frame to be re-inserted into a jitter buffer after it
had been removed from it. A case when this happened was if a frame was read
out of the jitterbuffer, passed to the translation core, and then multiple
frames were returned from said translation core. Upon multiple frames being
returned the first is passed on, but sebsequently "chained" frames are put
back into the read queue. Thus it was possible for a frame to go back into
the jitter buffer where this would cause problems.
This patch adds a flag to frames that are inserted into the channel's read
queue after translation. The abstract jitter buffer code then checks for this
flag and ignores any frames marked as such.
Richard Mudgett [Sat, 14 Jan 2017 03:23:49 +0000 (21:23 -0600)]
taskprocessor.c: Change when high water warning logged.
The task processor queue reached X scheduled tasks message was originally
intended to get logged only once per task processor to prevent spamming
the log. This is no longer necessary since high and low water thresholds
can better control when the message is logged.
It is beneficial to generate the warning each time a task processor
reaches the high water level because PJSIP stops processing new requests
while any high water alert is active. Without this change you would have
to enable at least debug level 3 logging to know about a repeated alert
trigger.
* Made generate the warning message whenever a task is pushed into the
task processor that triggers the high water alert.
* Appended 'again' to the warning for a repeated high water alert trigger.
Aaron An [Tue, 10 Jan 2017 11:54:16 +0000 (19:54 +0800)]
res_rtp_asterisk: Fix bug in function CHANNEL(rtcp, all_rtt)
Function CHANNEL(rtcp,all_rtt) CHANNEL(rtcp,all_loss) CHANNEL(rtcp,all_jitter)
always return 0.0 due to wrong define of macro "AST_RTP_SATA_SET" and
"AST_RTP_STAT_STRCPY".
It should compare "combined" with "stat" not "current_stat".
ASTERISK-26710 #close Reported-by: Aaron An Tested-by: AaronAn
Change-Id: Id4140fafbf92e2db689dac5b17d9caa009028a15
George Joseph [Wed, 11 Jan 2017 00:10:39 +0000 (17:10 -0700)]
debug_utilities: Create the ast_coredumper utility
This utility allows easy manipulation of asterisk coredumps.
* Configurable search paths and patterns for existing coredumps
* Can generate a consistent coredump from the running instance
* Can dump the lock_infos table from a coredump
* Dumps backtraces to separate files...
- thread apply 1 bt full -> <coredump>.thread1.txt
- thread apply all bt -> <coredump>.brief.txt
- thread apply all bt full -> <coredump>.full.txt
- lock_infos table -> <coredump>.locks.txt
* Can tarball corefiles and optionally delete them after processing
* Can tarball results files and optionally delete them after processing
* Converts ':' in coredump and results file names '-' to facilitate
uploading. Jira for instance, won't accept file names with colons
in them.
Tested on Fedora24+, Ubuntu14+, Debian6+, CentOS6+ and FreeBSD9+[1].
[1] For *BSDs, the "devel/gdb" package might have to be installed to
get a recent gdb. The utility will check all instances of gdb
it finds in $PATH and if one isn't found that can run python, it
prints a friendly error.
Joshua Colp [Thu, 5 Jan 2017 12:11:43 +0000 (12:11 +0000)]
res_pjsip_endpoint_identifier_ip: Add support for SRV lookups.
This change implements SRV support for the IP based endpoint
identifier module. All possible addresses through SRV are looked
up and added as matches. If no SRV records are available a
fallback to normal host resolution is done. If an IP address
is provided then no SRV lookup occurs.
This is configured using the "srv_lookups" option on the
identify section and defaults to "yes".
Alexander Traud [Thu, 22 Dec 2016 15:13:46 +0000 (16:13 +0100)]
res_pjsip_session: Access SIPDOMAIN via Dialplan.
This feature was available in the SIP channel driver chan_sip. For example,
Asterisk is the outbound proxy and has to handle all SIP-URIs, even domains not
local to Asterisk. In that case, SIPDOMAIN is used in the Dialplan, to detect
and dial remote SIP-URIs. This change here sets the SIP destination domain of
an inbound call (SIPDOMAIN) in the SIP channel driver res_pjsip as well.
Alexander Traud [Wed, 4 Jan 2017 11:50:11 +0000 (12:50 +0100)]
chan_sip: Remember SDP negotiation on SIP_CODEC_INBOUND.
After a SIP_CODEC_INBOUND in the dialplan, do not continue with cached formats
but remember the joint format. Cached formats contain default parameters,
often create an empty fmtp line. However, a joint format might have passed
format_get_joint(.) in a res_format_attr_* module (like Opus Codec) and
contain the resulting format parameters from a SDP negotiation.
George Joseph [Tue, 3 Jan 2017 21:14:09 +0000 (14:14 -0700)]
pjproject_bundled: Compile pjsua with max log level = 2
A while back, we changed config_site.h to set PJ_LOG_MAX_LEVEL = 6.
This allowed us to control the log level better from inside Asterisk.
An unfortunate side effect of this was that the pjsua binary and
python bindings were also compiled with log level set to 6 so whenever
a testsuite test that uses pjsua runs, it spits out 6795 lines of
debug in an instant even before the test starts. I believe this
overruns the Jenkins capture buffer and prevents the test from
properly terminating. In turn, this results in the testsuite just
hanging until the job is killed. It's more frequent on the higher
end agents because they can spit out the messages faster.
Unfortunately, the messages are all spit out before we have control
of the python pj.Lib instance where we can set logging levels so the
only alternative was to actually compile pjsua and _pjsua.so with an
overridden PJ_LOG_MAX_LEVEL. Although defining a lower max level was
done in the Makefile, the define in config_site.h had to be wrapped
with "#ifndef" so the change would take effect.
Joshua Colp [Thu, 22 Dec 2016 22:00:58 +0000 (22:00 +0000)]
chan_pjsip: Use session for retrieving CHANNEL() information.
The CHANNEL() dialplan function implementation for PJSIP allows
querying of PJSIP specific information. This used the channel
passed in to get the PJSIP session and associated information.
It is possible for this channel to be masqueraded and end
up as a different channel type by the time the information
request is actually acted upon.
This change retrieves the PJSIP session safely and accesses
data from it (including channel). This provides a guarantee
that the session and channel will not be altered when the
request is being acted upon.
Richard Mudgett [Fri, 23 Dec 2016 18:10:40 +0000 (12:10 -0600)]
bridge_native_rtp.c: Fix native rtp bridge data race.
native_rtp_bridge_compatible() didn't lock the bridge channels before
checking the channels for native bridging ability. As a result, one of
the channel's native format capabilities structure got replaced out from
under the native bridge check. Use of a stale pointer to freed memory
causes bad things to happen.
MALLOC_DEBUG, DO_CRASH, and the
tests/channels/pjsip/transfers/blind_transfer/caller_direct_media
testsuite test caught this.
* Add missing channel locking in native_rtp_bridge_compatible().
ast_rtp_remote_address_set() could pass an uninitialized 'us' parameter to
ast_ouraddrfor(). If ast_ouraddrfor() returns an error then the 'us'
parameter may not get initialized. Thus when the code tries to save the
'us' parameter to the local address we could try to copy a ridiculous
sized memory buffer and segfault.
* Made pass an initialized 'us' parameter to ast_ouraddrfor().
Richard Mudgett [Wed, 21 Dec 2016 23:54:42 +0000 (17:54 -0600)]
chan_rtp.c: Fix uninitialized memory crash.
unicast_rtp_request() could pass an uninitialized 'us' parameter to
ast_ouraddrfor(). If ast_ouraddrfor() returns an error then the 'us'
parameter may not get initialized. Thus when the code tries to save the
'us' parameter to the local address we could try to copy a ridiculous
sized memory buffer and segfault.
* Made pass an initialized 'us' parameter to ast_ouraddrfor() and abort
the UnicastRTP channel request if it fails.
Martin Tomec [Fri, 9 Dec 2016 18:23:37 +0000 (19:23 +0100)]
app_queue: Ensure member is removed from pending when hanging up.
In some cases member is added to pending_members, and the channel
is hung up before any extension state change. So the member would
stay in pending_members forever. So when we call do_hang, we
should also remove member from pending.
George Joseph [Sun, 18 Dec 2016 21:23:17 +0000 (14:23 -0700)]
pjproject_bundled: Make build single threaded
There were just too many issues in various environments with
multi threaded building of pjproject. It doesn't really speed
things up anyway since asterisk is already being compiled in
parallel.
Corey Farrell [Fri, 9 Dec 2016 02:00:02 +0000 (21:00 -0500)]
chan_sip: Reorder unload_module to deal with stuck TCP threads.
In some situations TCP threads may become frozen. This creates the
possibility that Asterisk could segfault if they become unfrozen after
chan_sip has been dlclose'd. This reorders the unload_module process to
allow abort if threads do not exit within 5 seconds.
High level order as follows:
1) Unregister from the core to stop new requests.
2) Signal threads to stop
3) Clear config based tables (but do not free the table itself).
4) Verify that threads have shutdown, cancel unload if not.
5) Clean all remaining resources.
George Joseph [Tue, 13 Dec 2016 20:06:34 +0000 (13:06 -0700)]
res_sorcery_memory_cache: Change an error to a debug message
When a sorcery user calls ast_sorcery_delete on an object that
may have already expired from the cache, res_sorcery_memory_cache
spits out an ERROR. Since this can happen frequently and validly when
an inbound registration expires after the cache entry expired, the
errors are unnecessary and misleading. Changed to a debug/1.
George Joseph [Fri, 9 Dec 2016 14:14:09 +0000 (07:14 -0700)]
pjproject_bundled: Retry download if previously saved tarball is bad
If a tarball is corrupted during download, the makefile will attempt to
download it again. If the tarball somehow gets corrupted after it's
downloaded however, the makefile was just failing. We now
retry the download.
HCOLON = *( LOWCTL / SP ) ":" SWS
LOWCTL = %x00-1F ; CTL without DEL
This discrepancy meant that SIP proxies in front of Asterisk with
chan_sip could pass on unknown headers with \x00-\x1F in them, which
would be treated by Asterisk as a different (known) header. For
example, the "To\x01:" header would gladly be forwarded by some proxies
as irrelevant, but chan_sip would treat it as the relevant "To:" header.
Those relying on a SIP proxy to scrub certain headers could mistakenly
get unexpected and unvalidated data fed to Asterisk.
This change fixes so chan_sip only considers SP/HTAB as valid tokens
before the colon, making it agree on the headers with other speakers of
SIP.
Joshua Colp [Tue, 15 Nov 2016 00:18:21 +0000 (00:18 +0000)]
res_format_attr_opus: Fix crash when fmtp contains spaces.
When an opus offer or answer was received that contained an
fmtp line with spaces between the attributes the module would
fail to properly parse it and crash due to recursion.
This change makes the module handle the space properly and
also removes the recursion requirement.
George Joseph [Tue, 6 Dec 2016 20:54:25 +0000 (13:54 -0700)]
res_pjsip_registrar: AMI Add RegistrationInboundContactStatuses command
The PJSIPShowRegistrationsInbound AMI command was just dumping out
all AORs which was pretty useless and resource heavy since it had
to get all endpoints, then all aors for each endpoint, then all
contacts for each aor.
PJSIPShowRegistrationInboundContactStatuses sends ContactStatusDetail
events which meets the intended purpose of the other command and has
significantly less overhead. Also, some additional fields that were
added to Contact since the original creation of the ContactStatusDetail
event have been added to the end of the event.
For compatibility purposes, PJSIPShowRegistrationsInbound is left
intact.
Richard Mudgett [Tue, 6 Dec 2016 22:45:38 +0000 (16:45 -0600)]
Bundled pjproject: Fix finding SIP transactions.
Occasionally SIP message transactions are not found when they should be.
In the particular case an incoming INVITE transaction is CANCELed but the
INVITE transaction cannot be found so a 481 response is returned for the
CANCEL. The problematic calls have a '_' character in the Via branch
parameter.
The problem is in the pjproject PJ_HASH_USE_OWN_TOLOWER feature's code.
The problem with the "own tolower" code is that it does not calculate the
same hash value as when the pj_tolower() function is used. The "own
tolower" code will erroneously modify the ASCII characters '@', '[', '\\',
']', '^', and '_'. Calls to pj_hash_calc_tolower() can use the
PJ_HASH_USE_OWN_TOLOWER substitute algorithm when enabled. Calls to
pj_hash_get_lower(), pj_hash_set_lower(), and pj_hash_set_np_lower() call
find_entry() which never uses the PJ_HASH_USE_OWN_TOLOWER algorithm. As a
result you may not be able to find a hash tabled entry because the
calculated hash values would differ.
George Joseph [Tue, 6 Dec 2016 18:06:45 +0000 (11:06 -0700)]
pjproject_bundled: Fix missing inclusion of symbols
Added back in a -g3, and an -O3 when DONT_OPTIMIZE is not set, to
the CFLAGS. Not sure how they went missing.
Also fixed an uninstall problem where we weren't removing the
symlink from libasteriskpj.so.2 to libasteriskpj.so. While I was
there, I fixed it for libasteriskssl as well.
Increasing the testsuite shutdown timeout before forcibly killing
Asterisk allowed more events to be sent out. Some tests failed as
a result. The tests/channels/pjsip/statsd/registrations failed
because we now get the statsd events that a comment in the test
configuration stated couldn't be intercepted. Unfortunately, we
get a variable number of events because of internal status state
transition races generating redundant statsd events.
We were reporting redundant statsd PJSIP.registrations.state changes
for internal state changes that equated to the same thing publicly.
* Made update_client_state_status() filter out redundant statsd
updates.
projects / thirdparty / asterisk.git / log
