Age | Commit message (Collapse) | Author |
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Continuing with the Who Hung Up? project for Asterisk 11, this adds
support to IAX2 for the HANGUPCAUSE hash.
Additionally, this breaks out some functionality in frame.c for getting
information about frame types and subclasses.
Review: https://reviewboard.asterisk.org/r/1941/
(issue SWP-4222)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369007 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(issue AST-807)
(issue AST-901)
(issue AST-908)
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merge.
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* Make non-normal dialplan execution routines be able to run on a hung up
channel. This is preparation work for hangup handler routines.
* Fixed ability to support relative non-normal dialplan execution
routines. (i.e., The context and exten are optional for the specified
dialplan location.) Predial routines are the only non-normal routines that
it makes sense to optionally omit the context and exten. Setting a hangup
handler also needs this ability.
* Fix Return application being able to restore a dialplan location
exactly. Channels without a PBX may not have context or exten set.
* Fixes non-normal execution routines like connected line interception and
predial leaving the dialplan execution stack unbalanced. Errors like
missing Return statements, popping too many stack frames using StackPop,
or an application returning non-zero could leave the dialplan stack
unbalanced.
* Fixed the AGI gosub application so it cleans up the dialplan execution
stack and handles the autoloop priority increments correctly.
* Eliminated the need for the gosub_virtual_context return location.
Review: https://reviewboard.asterisk.org/r/1984/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368985 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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The Hangup application used to just return -1 to cause normal dialplan
execution to hangup a channel. For the non-normal execution routines like
predial and connected-line interception routines, the hangup request would
exit the routine early but otherwise be ignored.
* Made the Hangup application not allow setting a cause code of zero. A
zero cause code is not defined.
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r368963 | qwell | 2012-06-14 13:47:03 -0500 (Thu, 14 Jun 2012) | 14 lines
Remove global symbol requirement from app_voicemail.
This uses the existing "function installation" stuff that already existed for
other functions, like getting message counts.
(closes issue AST-807)
(issue AST-901)
(issue AST-908)
Review: https://reviewboard.asterisk.org/r/1965/
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r368965 | qwell | 2012-06-14 14:04:57 -0500 (Thu, 14 Jun 2012) | 11 lines
These functions that were moved need to be static.
Also wrap test functions in a #ifdef.
(issue AST-807)
(issue AST-901)
(issue AST-908)
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AST-2012-008 (r367844) fixed a denial of service attack exploitable in the
Skinny channel driver that occurred when certain messages are sent after a
previously registered station sends an Off Hook message. Unresolved in that
patch is an issue in the Asterisk 10 releases, wherein, if a Station Key
Pad Button Message is processed after an Off Hook message, the channel driver
will inappropriately dereference a NULL pointer.
This patch fixes those places where the message handling or the channel
callback functions would attempt to dereference the line's pointer to the
device.
(issue ASTERISK-19905)
Reported by: Christoph Hebeisen
Tested by: mjordan, Christoph Hebeisen
Patches:
AST-2012-009-10.diff uploaded by mjordan (license 6283)
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The change has resulted in a linking error for certain versions
of GCC. This is much worse than the original issue, so for now,
temporarily revert the change. A more thorough change will be
sought out.
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This adds a callback that only fires when changes have been successfully
applied via the Config Options API.
Review: https://reviewboard.asterisk.org/r/1980/
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This adds the ability to handle a single filename alias for a config
file. This is useful if a config filename has changed, but the old
filename should be supported for backwards compatibility.
Review: https://reviewboard.asterisk.org/r/1981/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368920 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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This was discovered by trying to perform a call forward to an extension
that makes use of func_volume. When the local channel is optimized away,
the datastore on the local;2 channel would have its audiohook destroyed
rather than detaching the audiohook from the channel and then destroying
it.
With this patch, func_volume's datastore destructor takes the proper
route of detaching the audiohook and then destroying it.
(closes issue ASTERISK-19611)
reported by Volker Sauer
Patches:
ASTERISK-19611.patch uploaded by Mark Michelson (license #5049)
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Recently, various issues surrounding weak symbols have caused problems with
modules that rely on that feature to be enabled in menuselect. This includes
app_voicemail and chan_dahdi, as they both rely upon res_smdi and res_adsi,
which, in certain circumstances, may not be enabled by default in menuselect.
Because res_smdi/res_adsi are dependencies for chan_dahdi/app_voicemail, this
patch marks both as 'core' supported modules. This will allow both
app_voicemail and chan_dahdi to be enabled as well, regardless of whether or
not that system supports weak symbols.
(issue AST-900)
Reported by: Thomas Arimont
(issue AST-885)
Reported by: Denis Alberto Martinez
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In GCC 4.5+ the result is that Asterisk has a phantom
module loaded at startup, claiming to be res_adsi.
(closes issue ASTERISK-19920)
reported by Leif Madsen
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Post Asterisk 10, the MODULES_DIR variable no longer exists, and was replaced
with ASTMODDIR.
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r368830 modified the installation script to only create a directory if that
directory does not exist. If some directory variable was empty, it would attempt
to create the empty location. It also failed to create the ASTLIBDIR directory.
This patch fixes it such that the correct directories are made and only created if
a value specifying them actually exists.
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If a directory already exists, performing a 'make install' will remove the
permissions associated with the current directory and replace them with the
permissions of the user executing the install.
This patch changes this behavior to only perform an install on the directory
if the directory does not exist. Thus, if a user later changes the permissions
on that directory, those permissions will be preserved in subsequent installs.
Review: https://reviewboard.asterisk.org/r/1986
Review: https://reviewboard.asterisk.org/r/1864
(closes issue ASTERISK-19492)
Reported by: Karl Fife
Tested by: Paul Belanger, Tilghman Lesher
patches:
ASTERISK-19492 by pabelanger
(uploaded by mjordan)
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On incoming calls, we were setting the cid_tag on the dialog only if there was
no remote party information (Remote-Party-ID or P-Asserted-Identity) present.
The Caller ID tag is an invented parameter, though, and should be set no matter
the circumstance.
(closes issue ASTERISK-19859)
Reported by Thomas Arimont
(closes issue AST-884)
Reported by Trey Blancher
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In r367163, "send to voicemail" functionality was added to the SIP channel
driver. This required updating the party redirecting information for the
channel based on the headers provided in the REFER request. When the
redirecting party information is updated on the channel, a call to
ast_indicate_data occurs. Because handle_request_refer still had the sip_pvt
locked, a deadlock could occur between the pbx_thread and the do_monitor thread
servicing the REFER request.
This patch preserves the proper locking order between the channel and the
sip_pvt by ensuring that the sip_pvt is unlocked prior to updating the party
redirecting information on the channel.
(closes issue AST-903)
Reported by: Matt Jordan
patches:
jira_ast_903_trunk.patch by rmudgett (license 5621)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368793 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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ANI2 information is now parsed out of SIP From headers when present in
the oli, isup-oli, and ss7-oli parameters and is available via the
CALLERID(ani2) dialplan function.
(closes issue ASTERISK-19912)
Patch-by: Rob Gagnon
Review: https://reviewboard.asterisk.org/r/1947/
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Calling ast_set_hangupsource() with the channel lock held can result in a
deadlock because the function also locks the bridged channel.
(issue ASTERISK-19537)
(closes issue AST-891)
Reported by: Guenther Kelleter
Tested by: Guenther Kelleter
(closes issue ASTERISK-19801)
Reported by: Alec Davis
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Most of these were just saving returned values without using them and
in some cases the variable being saved to could be removed as well.
(issue ASTERISK-19672)
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http://svn.asterisk.org/svn/asterisk/branches/10
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Fix compilation in dev-mode
Backport a compilation fix in md5.c from trunk that only showed up in
dev-mode under certain compiler versions.
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* Check allocation function return values for failure. Crashing is bad.
* Tweak ast_regex_string_to_regex_pattern() parameters for proper ast_str
usage.
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Also fixes few warnings.
(closes issue #19675)
Reported by: dbohling
Patches:
fixmwi.patch uploaded by dbohling (license 6378)
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Original was testing for d->session, setting and testing again (all nested).
Removed duplicate testing and restructured function to test/return and then
the main code.
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Removed d->registered which was mirroring d->session. Changed relevant
references to use d->session instead.
Moved setting and unsetting of l->device from session register to device
configuration. As such, l->device will always be valid unless it is has not
been configured to a device. Revised various test where checking if a device
is registered to use l->device->session.
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parsed.
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Regression from -r367080 ringinuse commit.
(issue ASTERISK-19536)
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Adding multiple file support broke reloading an unchanged file. This
adds an enum for return values for the aco_process_* functions and
ensures that the config is not applied if res is not ACO_PROCESS_OK.
Review: https://reviewboard.asterisk.org/r/1979/
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Review: https://reviewboard.asterisk.org/r/1970/
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Added documentation describing what flags and arguments to pass to
aco_option_register for default option types. Also changed the ACL
handler to use the flags parameter to differentiate between "permit"
and "deny" instead of adding an additional vararg parameter.
Review: https://reviewboard.asterisk.org/r/1969/
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A deadlock can occur when a POTS phone tries to flash hook to originate a
second call for 3-way or transfer. If another process is scanning the
channels container when the POTS line flash hooks then a deadlock will
occur.
* Release the channel and private locks when creating a new channel as a
result of a flash hook.
(closes issue ASTERISK-19842)
Reported by: rmudgett
Tested by: rmudgett
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If a dialog-starting INVITE contains a to-tag, then Asterisk
will respond with a 481. In this case, the resulting incoming
ACK would not be matched, so Asterisk would continue retransmitting
the 481 until the transaction times out.
There were two issues. Asterisk, upon creating a sip_pvt would generate
a local tag. However, when the time came to transmit the 481, since there
was a to-tag in the INVITE, Asterisk would place this original to-tag
in the 481 response. When the ACK came in, Asterisk would attempt to
match the to-tag in the ACK to the generated local tag. Unfortunately,
Asterisk never actually transmitted a response with the generated local
tag, so the to-tag in the ACK would not match.
The other problem was that when the 481 was sent, nothing was set
on the sip_pvt to indicate what CSeq is expected in the ACK.
To fix the first problem, we zero out the to-tag seen in the incoming
INVITE. This way, Asterisk, when time to send a response, will send
its generated local tag instead.
To fix the second problem, we set the sip_pvt's pendinginvite to the
CSeq of the INVITE when we send a 481.
(closes issue ASTERISK-19892)
Reported by Mark Michelson
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Certain branches, such as Certified Asterisk, may have a modifier added to
them that specifies the features available in that branch. For branches, this
modifier is expected to be reflected in the location of the branch in
subversion. For example, a subversion of URL of /certified/branches/1.8.11
would have a feature modifier of 'certified'. This is slightly different then
how features are determined for tags, where the feature is part of the actual
tag name, e.g., "10.5.0-digiumphones".
In keeping with the nomenclature used for tags, the feature specifier for
branches is translated and placed after the revision numbers. For the example
given previously, this would result in a branch version of
"Asterisk SVN-branch-1.8.11-cert-rXXXXXX".
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When changing between different modes of hold, the flags were not being
cleared out properly causing a failure to change hold states.
(closes issue ASTERISK-19919)
Patch-by: Morten Tryfoss
Reported-by: Morten Tryfoss
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When a parked call was retrieved from the parking lot, it could not do a
blind transfer because it caused the involved calls to be hung up
unconditionally.
* Made the ParkedCall application return the ast_bridge_call() return
value.
(closes issue ABE-2862)
Reported by: Vlad Povorozniuc
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Resolve some build warnings
My newly upgraded compiler caught these usages of uninitialized values.
They weren't actually used.
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When localization was added to app_voicemail, these headers were altered
when they should have remained in en_US format for RFC compliance. This
reverts the changes to those two lines.
(closes issue ASTERISK-19876)
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This was essentially duplicated functionality where normal channels used
AST_CAUSE_ANSWERED_ELSEWHERE while local channels and queues used
AST_FLAG_ANSWERED_ELSEWHERE. This removes the flag and converts that usage
into AST_CAUSE_ANSWERED_ELSEWHER usage.
Review: https://reviewboard.asterisk.org/r/1944
(closes issue ASTERISK-19865)
Patch-by: Birger Harzenetter
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Revision 351130 broke corect HANGUPCAUSE setting
for the 404 case in chan_sip. Other cases were also
potentially broken. This patch fixes the relaying
of causes to be what they used to be.
(closes issue ASTERISK-19914)
Reported by Pavel Troller
Tested by Walter Doekes (via a reviewboard test to be committed later)
Patches:
chan_sip.diff uploaded by Pavel Troller (license #6302)
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(issue ASTERISK-19322)
(closes issue ASTERISK-19875)
Reported by: call
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Merged revisions 368469 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 368470 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368472 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368467 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue ASTERISK-19800)
Reported by: Billy Chia
Patches:
asterisk.vim.patch uploaded by Billy Chia (license #6381)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368466 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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support.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368455 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368441 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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