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On systems with lots of RAM (e.g. 24GB) /proc/sys/fs/file-max divided
by two can exceed the per-process file limit of 2^20. This patch
ensures the value is capped.
(Patch cleaned up by me.)
ASTERISK-24011 #close
Reported by: Michael Myles
Patches:
safe_asterisk-ulimit.diff uploaded by Michael Myles (License #6626)
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The underlying library, pjnath, that res_rtp_asterisk uses for ICE
support does not have support for ICE-TCP. As candidates are
passed through directly to it this can cause error messages to occur
when it receives something unexpected (such as a TCP candidate).
This change merely ignores all non-UDP candidates so they never
reach pjnath.
ASTERISK-24326 #close
Reported by: Joshua Colp
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This error message primarily applies to development tasks and will now
only show up when dev-mode is enabled via configure.
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In Asterisk 13+, any given message type is not guaranteed to exist even
if Asterisk comes up correctly since creation of the message type could
be declined. The indexer should not prevent Asterisk from starting
under these conditions.
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When message type creation is declined via stasis.conf, certain
operations log errors assuming that the declined type is being used
before initialization or after destruction. These error messages get
quite spammy for oft used message types and should not be logged in the
first place since the message type is validly NULL.
Reported by: Matt DiMeo
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Formats within a capabilities structure are addressed starting at 0, not 1.
Assuming 1 causes it to exceed an array.
ASTERISK-24389 #close
Reported by: Kevin Harwell
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424752 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Prior to this patch, the auth_reject_permanent parameter was not initialized on
the registration client state, leading to the parameter being disabled
regardless of the value specified in pjsip.conf.
This patch initialized the setting on the registration client state to the
provided configuration value.
ASTERISK-24398 #close
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If SendMessage encounters an error (such as incorrect input provided to the
action), it will currently return -1. Actions should only return -1 if the
connection to the AMI client should be closed. In this case, SendMessage
causing the client to disconnect is inappropriate.
This patch causes the action to return 0, which simply causes the action to
fail.
Review: https://reviewboard.asterisk.org/r/4024
ASTERISK-24354 #close
Reported by: Peter Katzmann
patches:
sendMessage.patch uploaded by Peter Katzmann (License 5968)
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Using the Bridge application to bridge a channel that is executing an
applicaiton such as Wait results in a lingering Surrogate channel in the
CLI "core show channels" output even though it has already hungup.
* Fix bridge_exec() to not hold onto the current_dest_chan ref once it has
been put into the bridge.
* Eliminated bridge_exec()'s use of RAII_VAR().
ASTERISK-24224 #close
Reported by: Mark Michelson
Review: https://reviewboard.asterisk.org/r/4041/
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An OPTIONS request that is sent to Asterisk but not to a specific endpoint is
currently sent a 404 in response. This is because, not surprisingly, an empty
extension is never going to be found in the dialplan.
This patch makes it so that we only attempt to look up the endpoint in the
dialplan if it is specified in the OPTIONS request URI.
#SIPit31
ASTERISK-24370 #close
Reported by: Matt Jordan
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Calling PJSIP_MEDIA_OFFER on a non-PJSIP channel is hazardous to your health.
It will treat the channels as a PJSIP channel, eventually hitting an ao2 error,
FRACKing on assertion error, and quite likely crashing.
This patch adds checks to the read/write callbacks that ensure that the channel
technology is of type 'PJSIP' before attempting to operate on the channel.
#SIPit31
ASTERISK-24382 #close
Reported by: Matt Jordan
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When a message that exceeds the PJ_MAX_PKT_SIZE is sent over a reliable
transport, it is possible (although it shouldn't occur) for pjproject to pass
up an rdata object with a NULL msg in the msg_info. Needless to say, things
that attempt to dereference this are in for a rough ride.
In particular, this caused crashes in three different locations, all of which
are 'low level' enough to intercept an rdata object early in processing:
(1) res_pjsip_logger
(2) res_hep_pjsip
(3) res_pjsip/distributor
Anything that can intercept an rdata object before res_pjsip/distributor should
be defensive when looking at the received packet.
#SIPit31
ASTERISK-24369 #close
Reported by: Matt Jordan
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A subscription that has been persisted can - for various reasons - fail to be
re-created on startup. This patch resolves a number of crashes that occurred
when a subscription cannot be re-created on several off-nominal paths.
#SIPit31
ASTERISK-24368 #close
Reported by: Matt Jordan
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424601 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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ASTERISK-24378 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4037/
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chan_sip: Clean leak on error path of process_sdp
Resolve leak in process_sdp that occurs in 2 error path's where
crypto lines are expected but not provided.
ASTERISK-24385 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4045/
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This fix applies to 13 and trunk.
ASTERISK-24384 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4043/
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ASTERISK-24384 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4043/
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ASTERISK-24199
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removed.
Adding a mixmonitor to a channel causes the bridge to change technologies
from native to simple_bridge so the call can be recorded. However, when
the mixmonitor is stopped the bridge does not switch back to the native
technology.
* Added unbridge requests to reevaluate the bridge when a channel
audiohook is removed.
* Moved the unbridge request into ast_audiohook_attach() ensure that the
bridge reevaluates whenever an audiohook is attached. This simplified the
mixmonitor and chan_spy start code as well.
* Added defensive code to stop_mixmonitor_full() in case additional
arguments are ever added to the StopMixMonitor application.
* Made ast_framehook_detach() not do an unbridge request if the framehook
does not exist.
* Made ast_framehook_list_fixup() do an unbridge request if there are any
framehooks. Also simplified the loop.
ASTERISK-24195 #close
Reported by: Jonathan Rose
Review: https://reviewboard.asterisk.org/r/4046/
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Performing a directed call pickup resulted in a deadlock when PJSIP
channels were involved.
A masquerade needs to hold onto the channel locks while it swaps channel
information between the two channels involved in the masquerade. With
PJSIP channels, the fixup routine needed to push a fixup task onto the
PJSIP channel's serializer. Unfortunately, if the serializer was also
processing a task that needed to lock the channel, you get deadlock.
* Added a new control frame that is used to notify the channels that a
masquerade is about to start and when it has completed.
* Added the ability to query taskprocessors if the current thread is the
taskprocessor thread.
* Added the ability to suspend/unsuspend the PJSIP serializer thread so a
masquerade could fixup the PJSIP channel without using the serializer.
ASTERISK-24356 #close
Reported by: rmudgett
Review: https://reviewboard.asterisk.org/r/4034/
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When you call ast_sorcery_create() you don't necessarily know which wizard is
going to be invoked. If it happens to be a wizard like 'config' that doesn't
have a 'create' virtual function you get a segfault in the
sorcery_wizard_create callback. This patch catches the null function pointer,
does an ast_assert, and logs an error.
Review: https://reviewboard.asterisk.org/r/4044/
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The pjsip config option default fixups from r424263 altered the
functional default from "allowed_not_screened" to "allowed". This
change restores the functional default value when none is provided.
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This corrects some issues introduced in the responses to the
CoreShowChannels AMI command as well as adding documentation for the
responses. The command in Asterisk 12 was missing the following fields:
Duration, Application, ApplicationData, and BridgedChannel and
BridgedUniqueID (replaced with BridgeId).
ASTERISK-24262 #close
Reported by: Mitch Claborn
Review: https://reviewboard.asterisk.org/r/4040/
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Due to the architecture of how media streams are handled each individual
handler adds connection details (IP address) for it. The first media stream
is then used as the top level SDP connection line. In practice each
line ends up being the same so to reduce the SDP size stream-level connection
information is also added to the SDP if it differs from the top level SDP
connection line.
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cipher names.
Improvements to the res_pjsip transport cipher option.
* Made the cipher option accept a comma separated list of OpenSSL cipher
names. Users of realtime will be glad if they have more than one name to
list.
* Added the CLI command 'pjsip list ciphers' so a user can know what
OpenSSL names are available for the cipher option.
* Updated the cipher option online XML documentation to specify what is
expected for the value.
* Updated pjsip.conf.sample to not indicate that ALL is acceptable since
ALL does not imply a preference order for the ciphers and PJSIP does not
simply pass the string to OpenSSL for interpretation.
ASTERISK-24199 #close
Reported by: Joshua Colp
Review: https://reviewboard.asterisk.org/r/4018/
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The 'outgoing' value was left off of the enumerator when first creating the
column. This patch adds it, and should gracefully upgrade keeping the existing
data in tact.
ASTERISK-23781 #close
Reported by: Stephen More
Review: https://reviewboard.asterisk.org/r/4013/
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Without setting rewrite_contact, an invite to an endpoint
behind NAT will not reach it - unless the endpoint itself
uses STUN or TURN to discover it's public URI. Thus, the
use of this should be in the sample documentation.
Review: https://reviewboard.asterisk.org/r/4036/
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ASTERISK-24222 #close
Reported by: Mark Michelson
Review: https://reviewboard.asterisk.org/r/4017/
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ASTERISK-24362 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4026/
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During the latest update to DTLS-SRTP support the ability to configure
the hash used for fingerprints was added. This gave us two supported ones:
SHA-1 and SHA-256. The default was accordingly updated to SHA-256.
Unfortunately this configuration ability was not exposed within res_pjsip.
This change adds a dtls_fingerprint option that controls it.
#SIPit31
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#SIPit31
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This updates the code behind PJSIP configuration options with custom
handlers to deal with the assigned default values properly where it
makes sense and adjusting the default value where it doesn't. Before
applying this patch, there were several cases where the default value
for an option would prevent that config section from loading properly.
Reported by: Thomas Thompson
Review: https://reviewboard.asterisk.org/r/4019/
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If contact rewriting is enabled but the contact differs in transport
from what is actually being used, messages after the initial INVITE
transaction can be sent to an incorrect transport/port combination. In
the case where this bug occurred the remote party never received a BYE
since it was sent to the remote party's TCP port over UDP.
Review: https://reviewboard.asterisk.org/r/4032/
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ASTERISK-22945 #related
Reported by: ibercom
Patches:
asterisk11-chan_sip-simplifies.patch uploaded by ibercom (License #6599)
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The peer is referenced at the end of sip_poke_peer, it should not get
an extra ref before the call to sip_poke_peer. This fixes a memory
leak.
ASTERISK-22945 #close
Reported by: ibercom
Tested by: Yuriy Gorlichenko
Patches:
asterisk11.patch uploaded by ibercom (License #6599)
Review: https://reviewboard.asterisk.org/r/4031/
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put IPv6 addresses in brackets.
#SIPit31
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present in SDP.
This change fixes an issue where ICE candidates put into the SDP did not contain
the 'raddr' and 'rport' information for server reflexive and relay candidates.
#SIPit31
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If there's an error on the pjsip command line or there are no objects, don't
print the column headers.
ASTERISK-24350 #close
Reported-by: Brad Latus
Tested-by: George Joseph
Tested-by: Brad Latus
Review: https://reviewboard.asterisk.org/r/4025/
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'==' is bashism (bashspecific, fails when dash is /bin/sh). Anyway, a
'case' works better there.
Originally committed in r375059 and r375060 on 2012-10-16 21:13:08.
ASTERISK-20567 #close
Reported by: Tzafrir Cohen
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Replace code using ast_uuid_generate() with simpler and faster code using
ast_uuid_generate_str(). The new code avoids a malloc(), free(), and
copy.
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* Fix threadpool_alloc() prototype.
* Add missing off-nominal NULL check of pool in threadpool_alloc().
* searializer_create() does not need to create the object with a lock as
the lock is not used.
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There are certain situations which no checks existed for which need to prevent
session refreshes. This includes sending a session refresh with SDP before SDP
negotiation has completed and sending a session refresh before the dialog itself
has been established. Checks for these have been added.
Additionally COLP related UPDATEs were including SDP when it is not needed.
Review: https://reviewboard.asterisk.org/r/4008/
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ASTERISK-24357 #close
Reported by: Jeremy Laine
Patches:
res_fax_bounds.patch (license #6561) patch uploaded by Jeremy Laine
Modified patch to not use magic numbers.
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ASTERISK-23768 #close
Reported by: Jeremy Lainé
Patches:
escape_manpage_hyphen.patch uploaded by Jeremy Lainé (License #6561)
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* Made memset the std struct in ast_sip_push_task_synchronous() because if
DEBUG_THREADS is enabled then uninitialized lock tracking data is used.
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request.
The crash on the issues is a result of an invalid transport configuration
change when asterisk is restarted. The attempt to send the qualify
request fails and we cleaned up. However, the callback is also called
which results in a double unref of the objects involved.
* Put a wrapper around pjsip_endpt_send_request() to detect when the
passed in callback is called because of an error so callers can know to
not cleanup.
* Made send_request_cb() able to handle repeated challenges (Up to 10).
* Fix periodic endpoint qualify OPTIONS sched deletion race by avoiding
it. The sched entry will no longer self stop and must be externally
stopped.
* Added REF_DEBUG description tags to struct sched_data in
pjsip_options.c.
* Fix some off-nominal ref leaks in schedule_qualify(),
qualify_and_schedule().
* Reordered pjsip_options.c module start/stop code to cleanup better on
error.
ASTERISK-24295 #close
Reported by: Rogger Padilla
Review: https://reviewboard.asterisk.org/r/3954/
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Make sure outbound proxy refs are always unreffed on dialog destruction.
Review: https://reviewboard.asterisk.org/r/4016/
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Prior to this commit, CDR and CEL tests were expected to trigger
FRACKs (i.e. assertions) due to the fact that the channels they
create have no formats on them. Some code was independently added
recently that attempts to prevent FRACKs from occurring by failing
early when attempting to set up translation paths if one or both
channels support no formats. Unfortunately, this attempt to be helpful
made the CDR and CEL tests go from simply FRACKing to outright
failing and in some cases, failing so badly as to crash Asterisk.
This commit seeks to correct past mistakes by adding the ulaw format
to channels created by the CDR and CEL unit tests. This makes setting
up translation paths succeed, eliminates previously-seen FRACKs, and
ultimately causes the unit tests to succeed again.
Review: https://reviewboard.asterisk.org/r/4014
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