Age | Commit message (Collapse) | Author |
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Kevin P Fleming suggested that r343157 should use __alignof__ instead
of sizeof. For most systems this won't be an issue, but better fix it
now while it's still fresh.
Review: https://reviewboard.asterisk.org/r/1573
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This patch fixes the format type check in ast_closestream and
filestream_destructor. Previously a comparison operator was used, but since
audio formats are no longer contiguous (and AST_FORMAT_AUDIO_MASK includes
formats that have a value greater than the video formats), a bitwise AND
operation is used instead. Duplicated code was also moved to filestream_close.
(closes issue ASTERISK-18682)
Reported by: Aldo Bedrij
Tested by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1580/
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Nick Lewis added them in https://reviewboard.asterisk.org/r/549/diff/1-2/
for no apparent reason. There is no way that params could become NULL in
that piece of code, so I removed these excess checks again.
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The opaque_data was added and enclosed in single quotes, assuming it
would be only a single line. The rest of the lines were appended after
the closing quote.
(closes issue ASTERISK-18852)
Reported by: peep_ on IRC
Review: https://reviewboard.asterisk.org/r/1577
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If capability is adjusted when switching to UDPTL during fax transmission, fax
teardown fails. Make sure capability is only touched if RTP is active. This
regression was introduced in R344385.
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JIRA AST-710
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Holding the channel lock while the CLI "core show channel" command is
executing can slow down the system. It could block the system if the
console output is halted or paused.
* Made capture the CLI "core show channel" output into a buffer to be
output after the channel is unlocked.
* Removed use of C++ keyword as a variable name. out renamed to obuf.
* Checked allocation of obuf for failure so will not crash.
(closes issue ASTERISK-18571)
Reported by: Pavel Troller
Tested by: rmudgett
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Attempting to call an extension which used Caller ID matching with a channel that
has an empty caller id string would result in a segmentation fault.
(closes issue ASTERISK-18392
Reported By: Ales Zelenik
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(closes issue ASTERISK-18848)
Reported by: Tony Mountifield
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Fixed app_queue.c:ring_entry() calling ast_call() with the channel locks
held. Chan_local attempts to do deadlock avoidance in its ast_call()
callback and could deadlock if a channel lock is already held.
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channel.
It was strange that the AgentCalled AMI event would get most of its
information from the incoming channel but then get the CallerID
information from the outgoing channel. Before connected line support was
added, this information was always the same at this point.
(closes issue ASTERISK-18152)
Reported by: Thomas Farnham
Tested by: rmudgett
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https://origsvn.digium.com/svn/asterisk/branches/10
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r344493 | dvossel | 2011-11-10 15:54:42 -0600 (Thu, 10 Nov 2011) | 12 lines
Fixes issue with ConfBridge participants hanging up during DTMF feature menu usage getting stuck in conference forever.
When a conference user enters the DTMF menu they are suspended from the
bridge while the channel is handed off to the DTMF feature code. If a
user entered this state and hungup, there existed a race condition where
the channel could not exit the conference because it was waiting on a
signal that would never arrive. This patch fixes that, because it would
stupid for me to talk about the problem and commit a patch for something else.
(closes issue ASTERISK-18829)
Reported by: zvision
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This fixes an issue where a user of a dynamic conference was asked for a PIN
twice. This also adds documentation to assist in future modifications to the
piece of code responsible for PIN checking.
(closes issue AST-670)
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Fix bug ASTERISK-16558 which dealt with the order of responses to incoming
streams defined by SDP.
Fix unreported bug where offering multiple same-type streams would cause
Asterisk to reply with an incorrect SDP response missing one or more streams
without a proper declination.
Fix bugs related to a single non-audio stream being offered with responses
requesting codecs that were not offered in the initial invite along with an
additional audio stream that was not in the initial invite.
Review: https://reviewboard.asterisk.org/r/1516/
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Patch by: jkonieczny (modified)
ASTERISK-18490
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Another deadlock between the conlock/hints and channels/channel locking
orders.
* Don't hold the channel and private lock in sip_new() when calling
ast_exists_extension().
(closes issue ASTERISK-18740)
Reported by: Byron Clark
Patches:
sip_exists_exten_dlock_3.diff (license #5041) patch uploaded by Gregory Hinton Nietsky
ASTERISK-18740.patch (license #6157) patch uploaded by Byron Clark
Tested by: Byron Clark
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The domain matching code prior to 1.8 used to manually remove the port
from the host:port string when determining if an incoming request
matched the list of domains. When switching to the new parsing
functions, the documentation implied that the "domain" was being
returned by these functions, when instead it was returning the
"hostport" as defined by RFC 3261. This led to confusion and resulted
in 1.8+ rejecting an incoming request from x.x.x.x:xxxxx when
domain=x.x.x.x was set in sip.conf.
This patch renames the "domain" variables in the parsing functions to
"hostport" to more accurately describe what it is that they are
returning and also properly truncates the resulting hostport strings
when dealing with domain matching.
Review: https://reviewboard.asterisk.org/r/1574/
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Review: https://reviewboard.asterisk.org/r/1575/
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Some PBXes require this for call status checking
(closes issue ASTERISK-18748)
Reported by: Fabrizio Lazzaretti
Patches:
ASTERISK-18748-5.patch (License #5415) patch uploaded by may213
Tested by: Fabrizio Lazzaretti
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The last time this code was touched (by me), a subtlety was missed based on the
difference between needing to check a pin's validity and the need to prompt
for a pin.
(closes issue ASTERISK-18488)
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ASTERISK-18739
Patch by: pawel (modified)
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Residual changes for Asterisk v10 branch from ASTERISK-18747 after
https://reviewboard.asterisk.org/r/1564/ commit and associated dialogs
callid hash key change fix.
* Make check_rtp_timeout() return CMP_MATCH if need to delete dialog from
dialogs_rtpcheck. This is an optimization to avoid an unneeded
lock/unlock and object search when using ao2_unlink.
* Prevent crash in check_rtp_timeout() if dialog->rtp is NULL.
Review: https://reviewboard.asterisk.org/r/1557/
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"dialplan remove include x from y" crashed when the amount of arguments
was less than 6.
(closes issue ASTERISK-18762)
Reported by: Andrey Solovyev
Tested by: Andrey Solovyev
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r343900 | dvossel | 2011-11-08 12:29:33 -0600 (Tue, 08 Nov 2011) | 11 lines
Fixes regression caused by r343635
There was a missing unlock for a function return that is only
present in Asterisk 10 and Asterisk Trunk.
(closes issue ASTERISK-18839)
Reported by: Michael L. Young
Patches:
asterisk-18839-missing-lock-trunk-v2.diff (License #5026) patch uploaded by Michael L. Young
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* Fixed a LOG_ERROR message referencing the config variable list v that
had previously been processed and became NULL.
* Added error return value set that was missing in an ast_append_ha()
error return path.
(closes issue ASTERISK-18743)
Reported by: Michele
Patches:
issueA18743-fix_dynamic_exclude_static_bad_host_log.patch (license #5674) patch uploaded by Walter Doekes
Tested by: Michele
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A hardcoded a branch number was in the prep_tarball which could not work. Changed
it to the variable.
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The "Trust RPID" and "Send RPID" entries in the "sip show settings" CLI command
pulled the flags from the incorrect global flags page. These are now read from
sip global flags page 0.
(closes issue AST-711)
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You can now use the built in variables , , and
within a dynamic weight. For example, this could be useful when you want
to pass requested lookup number to the SHELL() function which could be
used to execute a script to dynamically set the weight of the result.
(Closes issue ASTERISK-13657)
Reported by: Joel Vandal
Tested by: Leif Madsen, Russell Bryant
Patches:
asterisk-1.6-dundi-varhead.patch uploaded by Joel Vandal (License #5374)
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ASTERISK-18669
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Changing an object value used as a container key requires removing the
object from the container and reinserting it.
* Created change_callid_pvt() to call instead of build_callid_pvt(). The
change_callid_pvt() will correctly change the dialog callid so the ao2
conainter can explicitly unlink it.
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Fix a locking inversion in sip_send_mwi_to_peer that was causing deadlocks.
This function now requires that both the peer and associated pvt be unlocked
before it is called for cases where peer and peer->mwipvt form a circular
reference.
(closes issue ASTERISK-18663)
Review: https://reviewboard.asterisk.org/r/1563/
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The udptl port range was defined as 4000-4999 in the udptl.conf.sample,
as 4500-4599 if you didn't have a config and 4500-4999 if your config
was broken. Default is now 4000-4999.
(closes issue ASTERISK-16250)
Reviewed by: Tilghman Lesher
Review: https://reviewboard.asterisk.org/r/1565
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A dialog cannot be destroyed by the ao2_callback dialog_needdestroy
because of a deadlock between the dialogs container lock and the RWLOCK of
the events subscription list.
* Create dialogs_to_destroy container to hold dialogs that will be
destroyed.
* Ensure that the event subscription callback will never happen with an
invalid peer pointer by making the event callback removal the first thing
in the peer destructor callback.
NOTE: This particular deadlock will not happen with Asterisk 10, but some
of the changes still apply.
(closes issue ASTERISK-18747)
Reported by: Gregory Hinton Nietsky
Review: https://reviewboard.asterisk.org/r/1564/
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command "core show codec"
AST-699
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(these memleaks stop development of gk codes, now i can continue)
Fix printHandler 'Unbalanced Structure' issues with locking printHandler
data for single thread.
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The sqlite realtime handler assumed you had a static config configured
as well. The realtime multientry handler assumed that you weren't using
dynamic realtime.
(closes issue ASTERISK-18354)
(closes issue ASTERISK-18355)
Review: https://reviewboard.asterisk.org/r/1561
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The original REGISTERTRYING flag, in addition to being impossible to
check, also encroached on the space for the flag above it. This
patch moves the flags that were below REGISTERTRYING back to where
they were as though we had just removed the REGISTERTRYING option.
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This option is not only useless, but has been broken since inception since
the flag was never copied from the peer where it is set to the pvt where
it was checked. RFC 3261 specificially states that you should not send a
provisional response to a non-INVITE request, and if we did fix the code
so that it worked, it would cause the same kind of user enumeration
vulnerability that we've discussed with the nat= setting. This patch
removes registertrying option and any code that would have sent a 100
response to a register.
Review: https://reviewboard.asterisk.org/r/1562/
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Changeset r342927 introduced a warning which was only supposed to be
emitted when a found realtime peer had an empty (or no) name. It turned
out that there were some inconsistencies left. Now found peers with an
empty name are explicitly ignored like before r342927 but better.
Reviewed by: Stefan Schmidts, Terry Wilson
Review: https://reviewboard.asterisk.org/r/1560
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Integers should always be aligned. For some platforms (ARM, SPARC) this
is more important than for others. This changeset ensures that the
string field string lengths are aligned on *all* platforms, not just on
the SPARC for which there was a workaround. It also fixes that the
length integer can be resized to 32 bits without problems if needed.
(closes issue ASTERISK-17310)
Reported by: radael, S Adrian
Reviewed by: Tzafrir Cohen, Terry Wilson
Tested by: S Adrian
Review: https://reviewboard.asterisk.org/r/1549
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(closes issue ASTERISK-17422)
Reported by: Leif Madsen
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In queues.conf.sample the leastrecent strategy was incorrectly described. Now updated
to reflect how the strategy actually checks peers.
(closes issue ASTERISK-17854)
Reported by: Sebastian Denz
Patches:
queues.conf-doc_issue.patch (License #6139)
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The MeetMe application documentation has some comments about usage of DAHDI,
and they were a bit outdated relative to modern DAHDI releases. This patch
changes the comment to just tell the user that a functional DAHDI timing
source is required, and no longer mention 'dahdi_dummy', since that module
does not exist in current DAHDI releases.
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There were several problems with the dynamic realtime peer/user lookup
code. The lookup logic had become rather hard to read due to lots of
incremental changes to the realtime_peer function. And, during the
addition of the sipregs functionality, several possibilities for memory
leaks had been introduced. The insecure=port matching has always been
broken for anyone using the sipregs family. And, related, the broken
implementation forced those using sipregs to *still* have an ipaddr
column on their sippeers table.
Thanks Terry Wilson for comprehensive testing and finding and fixing
unexpected behaviour from the multientry realtime call which caused
the realtime_peer to have a completely unused code path.
This changeset fixes the leaks, the lookup inconsistenties and that
you won't need an ipaddr column on your sippeers table anymore (when
you're using sipregs). Beware that when you're using sipregs, peers
with insecure=port will now start matching!
(closes issue ASTERISK-17792)
(closes issue ASTERISK-18356)
Reported by: marcelloceschia, Walter Doekes
Reviewed by: Terry Wilson
Review: https://reviewboard.asterisk.org/r/1395
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Somewhere between 1.4 and 1.8 the sipusers family has become completely
unused. Before that, the sipfriends family had been obsoleted in favor
of separate sipusers and sippeers families. Apparently, they have been
merged back again into a single family which is now called "sippeers".
Reviewed by: irroot, oej, pabelanger
Review: https://reviewboard.asterisk.org/r/1523
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* Fixed typo in format_cap.c:joint_copy_helper() using the wrong variable.
* Fix potential race between checking if an interface exists and adding it
to the container in format.c:ast_format_attr_reg_interface().
* Fixed double rwlock destroy in format.c:ast_format_attr_init() error
exit path.
* Simplified format.c:find_interface() and format.c:has_interface().
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When an extension is removed from a context, its entry in the pattern match
tree is not deleted. Instead, the extension is marked as deleted. When an
extension is removed and re-added, if that extension is also a prefix of
another extension, several log messages would report an error and did not
check whether or not the extension was deleted before accessing the memory.
Additionally, if the extension was already in the tree but previously
deleted, and the pattern was at the end of a match, the findonly flag was
not honored and the extension would be erroneously undeleted.
Additionaly, it was discovered that an IAX2 peer could be unregistered
via the CLI, while at the same time it could be scheduled for unregistration
by Asterisk. The unregistration method now checks to see if the peer
was already unregistered before continuing with an unregistration.
(closes issue ASTERISK-18135)
Reported by: Jaco Kroon, Henry Fernandes, Kristijan Vrban
Tested by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1526
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