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.
ASTERISK-26672
Change-Id: I1ef7a7c09f4da4f15dcb6de660d2bcac5f2a95c0
We access uninitialized memory when the 'ourip' parameter does not
have an initial guess to our IP address.
ASTERISK-26672
Change-Id: I35507ea1ad7455d2be188f6ccdd4add7bd150e15
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.
ASTERISK-26621 #close
Change-Id: Iae476b5c06481db18ebe0fa594b3e80fdc9a7d54
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.
Change-Id: Ie5648fb91bb89b4224b6bf43a0daa1af793c4ce1
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.
ASTERISK-26586
Change-Id: Ie23692041d838fbd35ece61868f4c640960ff882
The AC_ARG_WITH macro's shell variable is withval; not enableval. Purely
coincidentally, the option would work when --enable-dev-mode is given.
Also fixed a portability problem with bootstrap.sh, since -printf is not
a portable option for find.
Change-Id: I0f0e5b1a934b5af5737713834361e9c95b96b376
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.
Change-Id: Idf3a67038c16e3da814cf612ff4d6d18ad29ecd7
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.
ASTERISK-26653 #close
Change-Id: I1b24d454852d80186f60c5a65dc4624ea8a1c359
Consider reviewing the expression of the 'A = B != C' kind.
The expression is calculated as following: 'A = (B != C)'
Change-Id: Ibaa637dfda47d51a20e26069d3103e05ce80003d
RFC says SIP headers look like:
HCOLON = *( SP / HTAB ) ":" SWS
SWS = [LWS] ; sep whitespace
LWS = [*WSP CRLF] 1*WSP ; linear whitespace
WSP = SP / HTAB ; from rfc2234
chan_sip implemented this:
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.
ASTERISK-26433 #close
AST-2016-009
Change-Id: I78086fbc524ac733b8f7f78cb423c91075fd489b
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.
ASTERISK-26579
Change-Id: I01f53e5d9fa9f1925a7365f8d25071b5b3ac2dc3
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.
ASTERISK-26644 #close
Change-Id: I326f12c9ecb52bf37ba03f0748749de4da01490a
Fix the tests for DNS to use older style nameser.h as
in ASTERISK-26608.
Tested on: OpenBSD 6.0, Debian 8
ASTERISK-26647 #close
Change-Id: I285913c44202537c04b3ed09c015efa6e5f9052d
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.
* Simply disable PJ_HASH_USE_OWN_TOLOWER.
ASTERISK-26490 #close
Change-Id: If89bfdb5f301b8b685881a9a2a6e0c3c5af32253
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.
Change-Id: I9e00873b1e9082d05b5549d974534b48a2142556
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.
ASTERISK-26527
Change-Id: If851c7d514bb530d9226e4941ba97dcf52000646
The latest Release candidate fails to create RTP streams when IPv6
is not available. Due to the changes made in September the ast_sockaddr
structure passed around to create these streams is always of AF_INET6
type, causing failure when used for IPv4. This patch adds a utility
function to check for availability of IPv6 and applies such check
at startup to determine how to create the ast_sockaddr structures.
ASTERISK-26617 #close
Change-Id: I627a4e91795e821111e1cda523f083a40d0e0c3e
The response from gmail calendar includes the string name
"caldav:calendar-data". res_calendar_caldav implements
the example included in RFC 4791: string "C:calendar-data".
When reading the calendar, res_calendar_caldav compare the
string and if does not match just discards the event.
This commit compares the response to both strings,
successfully loading gmail calendar events.
Writing to gmail calendar is working prior to this fix.
ASTERISK-26624
Reported by: Eduardo S. Libardi
Change-Id: Ia1eef10552ae616efb645d390f5ffe81260d7d4a
Use of the new logging is as simple as issuing the new CLI command or
setting the new pjproject.conf option.
Other options that can affect the logging are how you have the pjproject
log levels mapped to Asterisk log types in pjproject.conf and if you have
configured Asterisk to log the DEBUG type messages. Altering the
pjproject.conf level mapping shouldn't be necessary for most installations
as the default mapping is sensible. Configuring Asterisk to log the DEBUG
message type is standard practice for collecting debug information.
* Added CLI "pjproject set log level" command to dynamically adjust the
maximum pjproject log message level.
* Added CLI "pjproject show log level" command to see the currently set
maximum pjproject log message level.
* Added pjproject.conf startup section "log_level" option to set the
initial maximum pjproject log message level so all messages could be
captured from initialization.
* Set PJ_LOG_MAX_LEVEL to 6 to compile in all defined logging levels into
bundled pjproject. Pjproject will use the currently set run time log
level to determine if a log message is generated just like Asterisk
verbose and debug logging levels.
* In log_forwarder(), made always log enabled and mapped pjproject log
messages. DEBUG mapped log messages are no longer gated by the current
Asterisk debug logging level.
* Removed RAII_VAR() from res_pjproject.c:get_log_level().
ASTERISK-26630 #close
Change-Id: I6dca12979f482ffb0450aaf58db0fe0f6d2e5389
Richard Mudgett
Joshua Colp
George Joseph
Martin Tomec
Corey Farrell
David M. Lee
zuul
Badalyan Vyacheslav
Kevin Harwell
Walter Doekes
snuffy
Guido Falsi
Eduardo S. Libardi