Revision 360862 was intended to improve identities sent in dialog-info
NOTIFY requests. Some users reported that hint became broken once this
was done. It's not clear exactly what part of the patch has caused this
regression, but broken hints are bad.
For now, this revision is being reverted so that the next releases of
Asterisk do not have bad behavior in them. The original reported issue
will have to be fixed differently in the next version of Asterisk.
(issue ASTERISK-16735)
........
Merged revisions 364706 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 364707 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364708 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The configuration option to specify a custom sound_leader_has_left file for a
conference bridge was not being parsed. This patch fixes it so that a custom
sound file will now be used.
(closes issue ASTERISK-19771)
Reported by: Pawel Kuzak
Tested by: Pawel Kuzak, Michael L. Young
Patches: leaderhasleft_sound.dpatch uploaded by Pawel Kuzak (license 6380)
Review: https://reviewboard.asterisk.org/r/1884/
........
Merged revisions 364536 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364537 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If enabled using the keepalive option in sip.conf a small packet will be sent
at a regular interval to keep the NAT mapping open. This is lightweight as the
remote side does not need to parse and handle a SIP message.
(closes issue AST-783)
Review: https://reviewboard.asterisk.org/r/1756/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364500 65c4cc65-6c06-0410-ace0-fbb531ad65f3
md5.c: In function ‘MD5Final’:
md5.c:154:2: error: dereferencing type-punned pointer will break strict-aliasing rules [-Werror=strict-aliasing]
md5.c:155:2: error: dereferencing type-punned pointer will break strict-aliasing rules [-Werror=strict-aliasing]
There is an md5 unit test and it still passes.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364462 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The warnings were about argv[0] being used uninitialized, which is correct.
Just remove setting username to this value, since username is set again before
it actually gets used.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364438 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Add two new dialplan functions: FEATURE() and FEATUREMAP(). FEATURE()
lets you set some of the configuration options from the [general] section
of features.conf on a per-channel basis. FEATUREMAP() lets you customize
the key sequence used to activate built-in features, such as blindxfer,
and automon. See the built-in documentation for details.
Review: https://reviewboard.asterisk.org/r/1871/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364437 65c4cc65-6c06-0410-ace0-fbb531ad65f3
........
r364365 | twilson | 2012-04-27 17:31:01 -0500 (Fri, 27 Apr 2012) | 11 lines
Fix ast_parse_arg numeric type range checking and add tests
ast_parse_arg wasn't checking for strto* parse errors or limiting
the results by the actual range of the numeric types. This patch fixes
that and adds unit tests as well.
Review: https://reviewboard.asterisk.org/r/1879/
........
Merged revisions 364340 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
r364369 | twilson | 2012-04-27 17:33:10 -0500 (Fri, 27 Apr 2012) | 2 lines
Add missing test_config.c
........
Merged revisions 364365,364369 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364397 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The method ast_tvdiff_ms attempts to calculate the difference, in milliseconds,
between two timeval structs, and return the difference in a 64-bit integer.
Unfortunately, it assumes that the long tv_sec/tv_usec members in the timeval
struct are large enough to hold the calculated values before it returns. On
64-bit machines, this might be the case, as a long may be 64-bits. On 32-bit
machines, however, a long may be less (32-bits), in which case, the calculation
can overflow.
This overflow caused significant problems in MixMonitor, which uses the method
to determine if an audio factory, which has not presented audio to an audiohook,
is merely late in providing said audio or will never provide audio. In an
overflow situation, the audiohook would incorrectly determine that an audio
factory that will never provide audio is merely late instead. This led to
situations where a MixMonitor never recorded any audio. Note that this happened
most frequently when that MixMonitor was started by the ConfBridge application
itself, or when the MixMonitor was attached to a Local channel.
(issue ASTERISK-19497)
Reported by: Ben Klang
Tested by: Ben Klang
Patches:
32-bit-time-overflow-10-2012-04-26.diff (license #6283) by mjordan
(closes issue ASTERISK-19727)
Reported by: Mark Murawski
Tested by: Michael L. Young
Patches:
32-bit-time-overflow-2012-04-27.diff (license #6283) by mjordan)
(closes issue ASTERISK-19471)
Reported by: feyfre
Tested by: feyfre
(issue ASTERISK-19426)
Reported by: Johan Wilfer
Review: https://reviewboard.asterisk.org/r/1889/
........
Merged revisions 364277 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 364285 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364287 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Unref the SIP pvt stored in the refer structure as soon as it is no longer
needed so that the pvt and associated file descriptors can be freed sooner.
This change makes a reference decrement unnecessary in code that handles SIP
BYE/Also transfers which should not touch the reference anyway.
(Closes issue ASTERISK-19579)
Reported by: Maciej Krajewski
Tested by: Maciej Krajewski
........
Merged revisions 364258 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 364259 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364260 65c4cc65-6c06-0410-ace0-fbb531ad65f3
As a continuation of the patch in r356604, which allowed for the
reloading of SRTP keys in re-INVITE transfer scenarios, this patch
addresses the more common case where a new key is requested within
the context of a current SIP dialog. This can occur, for example, when
certain phones request a SIP hold.
Previously, once a dialog was associated with an SRTP object, any
subsequent attempt to process crypto keys in any SDP offer - either
the current one or a new offer in a new SIP request - were ignored. This
patch changes this behavior to only ignore subsequent crypto keys within
the current SDP offer, but allows future SDP offers to change the keys.
(issue ASTERISK-19253)
Reported by: Thomas Arimont
Tested by: Thomas Arimont
Review: https://reviewboard.asteriskorg/r/1885/
........
Merged revisions 364203 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 364204 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364205 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When party B does an attended transfer of party A to party C, the
attending bridge between party B and C should not be running an h exten
when the bridge ends. Running an h exten now sets a softhangup flag to
ensure that an AGI will run in dead AGI mode.
* Set the AST_FLAG_BRIDGE_HANGUP_DONT on the party B channel for the
attending bridge between party B and C.
(closes issue AST-870)
(closes issue ASTERISK-19717)
Reported by: Mario
(closes issue ASTERISK-19633)
Reported by: Andrey Solovyev
Patches:
jira_asterisk_19633_v1.8.patch (license #5621) patch uploaded by rmudgett
Tested by: rmudgett, Andrey Solovyev, Mario
........
Merged revisions 364060 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 364065 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364082 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The reference held for SIP blind transfers using the Replaces header in an
INVITE was never freed on success and also failed to be freed in some error
conditions. This caused a file descriptor leak since the RTP structures in use
at the time of the transfer were never freed. This reference leak and another
relating to subscriptions in the same code path have now been corrected.
(closes issue ASTERISK-19579)
........
Merged revisions 363986 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 363987 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363988 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Some switches may not handle the call-deflection/call-rerouting message if
the call is disconnected too soon after being sent. Asteisk was not
waiting for any reply before disconnecting the call.
* Added a 5 second delay before disconnecting the call to wait for a
potential response if the peer does not disconnect first.
(closes issue ASTERISK-19708)
Reported by: mehdi Shirazi
Patches:
jira_asterisk_19708_v1.8.patch (license #5621) patch uploaded by rmudgett
Tested by: rmudgett
........
Merged revisions 363730 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 363734 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363740 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Some ISDN switches occasionally fail to send a RESTART ACKNOWLEDGE in
response to a RESTART request.
* Made the second SETUP received after sending a RESTART request clear the
channel resetting state as if the peer had sent the expected RESTART
ACKNOWLEDGE before continuing to process the SETUP. The peer may not be
sending the expected RESTART ACKNOWLEDGE.
(issue ASTERISK-19608)
(issue AST-844)
(issue AST-815)
Patches:
jira_ast_815_v1.8.patch (license #5621) patch uploaded by rmudgett (modified)
........
Merged revisions 363687 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 363688 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363689 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Asterisk has a setting for the minimum allowed DTMF. If we get shorter
DTMF tones, these will be changed to the minimum on the outbound call
leg.
(closes issue ASTERISK-19772)
Review: https://reviewboard.asterisk.org/r/1882/
Reported by: oej
Tested by: oej
Patches by: oej
Thanks to the reviewers.
1.8 branch for this patch: agave-dtmf-duration-asterisk-conf-1.8
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363558 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Found a small amount of curly brackets in my hotel room here in Denmark.
I hereby donate them to the Asterisk project.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363480 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1) B calls A with Dial option T
2) B DTMF atxfer to C
3) B hangs up
4) C does not answer
5) B is called back
6) B answers
7) B cannot initiate transfers anymore
* Add dial features datastore to recalled party B channel that is a copy
of the original party B channel's dial features datastore.
* Extracted add_features_datastore() from add_features_datastores().
* Renamed struct ast_dial_features features_caller and features_callee
members to my_features and peer_features respectively. These better names
eliminate the need for some explanatory comments.
* Simplified code accessing the struct ast_dial_features datastore.
(closes issue ASTERISK-19383)
Reported by: lgfsantos
........
Merged revisions 363428 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 363429 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363430 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Simplify some code in app_dial and app_queue by calling
ast_app_exec_macro() and ast_app_exec_sub().
* Fix minor locking issue in app_dial for post-answer macro/gosub
MACRO/GOSUB_RESULT=GOTO: handling.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363269 65c4cc65-6c06-0410-ace0-fbb531ad65f3
As detailed in the advisory, AMI users without write authorization for SYSTEM class AMI
actions were able to run system commands by going through other AMI commands which did
not require that authorization. Specifically, GetVar and Status allowed users to do this
by setting their variable/s options to the SHELL or EVAL functions.
Also, within 1.8, 10, and trunk there was a similar flaw with the Originate action that
allowed users with originate permission to run MixMonitor and supply a shell command
in the Data argument. That flaw is fixed in those versions of this patch.
(closes issue ASTERISK-17465)
Reported By: David Woolley
Patches:
162_ami_readfunc_security_r2.diff uploaded by jrose (license 6182)
18_ami_readfunc_security_r2.diff uploaded by jrose (license 6182)
10_ami_readfunc_security_r2.diff uploaded by jrose (license 6182)
........
Merged revisions 363117 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2
........
Merged revisions 363141 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 363156 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363159 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If Asterisk receives a SIP UPDATE request after a call has been terminated and
the channel has been destroyed but before the SIP dialog has been destroyed, a
condition exists where a connected line update would be attempted on a
non-existing channel. This would cause Asterisk to crash. The patch resolves
this by first ensuring that the SIP dialog has an owning channel before
attempting a connected line update. If an UPDATE request is received and no
channel is associated with the dialog, a 481 response is sent.
(closes issue ASTERISK-19770)
Reported by: Thomas Arimont
Tested by: Matt Jordan
Patches:
ASTERISK-19278-2012-04-16.diff uploaded by Matt Jordan (license 6283)
........
Merged revisions 363106 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 363107 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363108 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When handling a keypad button message event, the received digit is placed into
a fixed length buffer that acts as a queue. When a new message event is
received, the length of that buffer is not checked before placing the new digit
on the end of the queue. The situation exists where sufficient keypad button
message events would occur that would cause the buffer to be overrun. This
patch explicitly checks that there is sufficient room in the buffer before
appending a new digit.
(closes issue ASTERISK-19592)
Reported by: Russell Bryant
........
Merged revisions 363100 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2
........
Merged revisions 363102 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 363103 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363105 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reimplement the "corosync show members" CLI command using a CPG iterator
instead of the cpg_membership_get API call. This will also show all
CPG members, including those in groups other than 'asterisk', which may
be useful at some point for debugging purposes.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363045 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Redo ast_app_run_sub()/ast_app_exec_sub() to use a known return point so
execution will stop after the routine returns there.
(s@gosub_virtual_context:1)
* Create ast_app_exec_macro() and ast_app_exec_sub() to run the macro and
gosub application respectively with the parameter string already created.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362962 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Move debug message in ast_rtp_instance_early_bridge_make_compatible() to
be output when what it states has actually happened.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362920 65c4cc65-6c06-0410-ace0-fbb531ad65f3
