Commit graph

22662 commits

Author SHA1 Message Date
Matthew Jordan
ee11118695 Fix crash in unloading of res_adsi module
When res_adsi is unloaded, it removes the ADSI functions that it previously installed
by passing a NULL adsi_funcs pointer to ast_adsi_install_funcs.  This function was not
checking whether or not the adsi_funcs pointer passed in was NULL before dereferencing
it to check whether or not the version of the functions matches what the core was
expecting it.

This patch makes it so that the version is only checked if a potentially valid adsi_funcs
pointer was passed in.  Passing in NULL removes the installed functions, bypassing the
version check.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369392 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-26 13:23:12 +00:00
Matthew Jordan
5d31fb2dd2 Update "manager show event" to support tab completion
Thank you rmudgett for pointing out that I was missing this in the initial
check-in for AMI event documentation (r369346)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369386 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-25 20:43:26 +00:00
Matthew Jordan
bebdbf3381 Fix incorrect duration reporting in CDRs created in batch mode
Certain places in core/cdr.c would, if the duration value were 0, calculate the
duration as being the delta between the current time and the time at which the
CDR record was started.  While this does not typically cause a problem in
non-batch mode, this can cause an issue in batch mode where CDR records are
gathered and written long after those calls have ended. In particular, this
affects calls that were never answered, as those are expected to have a duration
of 0.  Often, this would result in CDR logs with a significant number of calls
with lengthy durations, but dispositions of "BUSY".

Note that this does not affect cdr_csv, as that backend does not use
ast_cdr_getvar and instead directly reports the duration value.  The affected
core backends include cdr_apative_odbc and cdr_custom; other extended or
deprecated CDR backends may potentially still directly manipulate the duration
values.

(issue ASTERISK-19860)
Reported by: Thomas Arimont

(issue AST-883)
Reported by: Thomas Arimont
Tested by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/1996/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369370 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-25 19:39:03 +00:00
Mark Michelson
e0883154cf Re-fix how local tag is generated when sending a 481 to an INVITE.
Match our local tag to whatever to-tag was sent in the initial INVITE.
Because the size of the to-tag may not fit in the buffer in the sip_pvt,
it has been changed to a string field.

(closes issue ASTERISK-19892)
reported by Walter Doekes

Review: https://reviewboard.asterisk.org/r/1977
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369367 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-25 19:26:31 +00:00
Matthew Jordan
82a7409c15 Add AMI event documentation
This patch adds the core changes necessary to support AMI event documentation
in the source files of Asterisk, and adds documentation to those AMI events
defined in the core application modules.  Event documentation is built from
the source by two new python scripts, located in build_tools:
get_documentation.py and post_process_documentation.py.

The get_documentation.py script mirrors the actions of the existing AWK
get_documentation scripts, except that it will scan the entirety of a source
file for Asterisk documentation.  Upon encountering it, if the documentation
happens to be an AMI event, it will attempt to extract information about the
event directly from the manager event macro calls that raise the event.  The
post_process_documentation.py script combines manager event instances that
are the same event but documented in multiple source files.  It generates
the final core-[lang].xml file.

As this process can take longer to complete than a typical 'make all', it
is only performed if a new make target, 'full', is chosen.

Review: https://reviewboard.asterisk.org/r/1967/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369346 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-25 17:59:34 +00:00
Richard Mudgett
d0fda07d74 Fix Bridge application occasionally returning to the wrong location.
* Fix do_bridge_masquerade() getting the resume location from the zombie
channel.  The code must not touch a clone channel after it has masqueraded
it.  The clone channel has become a zombie and is starting to hangup.

(closes issue ASTERISK-19985)
Reported by: jamicque
Patches:
      jira_asterisk_19985_v1.8.patch (license #5621) patch uploaded by rmudgett
Tested by: jamicque
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369329 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-25 16:07:02 +00:00
Mark Michelson
453e01725d Multiple revisions 369323-369324
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  r369323 | mmichelson | 2012-06-25 10:35:43 -0500 (Mon, 25 Jun 2012) | 9 lines
  
  Eliminate embedding of res_adsi.so module.
  
  The way this is done is to stop using the optional API.
  Instead, res_adsi.so, when loaded fills in a table of
  function pointers.
  
  Review: https://reviewboard.asterisk.org/r/1991
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  r369324 | mmichelson | 2012-06-25 10:50:17 -0500 (Mon, 25 Jun 2012) | 2 lines
  
  Forgot to svn add this file in my last commit.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369326 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-25 15:55:25 +00:00
Mark Michelson
87810af23d Be more consistent with the return code for requests received from invalid domain.
When Asterisk receives an INVITE from an external domain when allowexternaldomains=no
send a 403 instead of a 404. This is consistent with Asterisk's behavior when receiving
a REGISTER in this situation.

(Closes issue ASTERISK-19601)
Reported by Matthew Jordan
Patches:
	ASTERISK-19601-no401.patch uploaded by Mark Michelson (License #5049)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369304 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-25 14:30:19 +00:00
Richard Mudgett
b78d79c203 Fix F and F(x) action logic in Bridge application.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369296 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-23 00:33:41 +00:00
Richard Mudgett
b857a633e0 Fix Bridge application and AMI Bridge action error handling.
* Fix AMI Bridge action disconnecting the AMI link on error.

* Fix AMI Bridge action and Bridge application not checking if their
masquerades were successful.

* Fix Bridge application running the h-exten when it should not.

* Made do_bridge_masquerade() return if the masquerade was successful so
the Bridge application and AMI Bridge action could deal with it correctly.

* Made bridge_call_thread_launch() hangup the passed in channels if the
bridge_call_thread fails to start.  Those channels would have been
orphaned.

* Made builtin_atxfer() check the success of the transfer masquerade
setup.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369295 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-23 00:29:18 +00:00
Richard Mudgett
62274463fa Explicitly check caller hangup in app Queue rather than a polluted res2 value.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369264 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-22 22:12:06 +00:00
Richard Mudgett
30a417dc6c Fix F and F(x) action logic in Queue application.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369261 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-22 21:51:05 +00:00
Richard Mudgett
3d1e26d2d2 Check if PBX was started and fix F and F(x) action logic in Dial application.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369260 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-22 21:43:44 +00:00
Richard Mudgett
f3bf3acbfd Check if PBX was started for generic CCSS recall.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369240 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-22 21:06:36 +00:00
Richard Mudgett
e07ba960f9 Change incorrect chan_sip zombie hangup debug message. They are all zombies now.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369237 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-22 20:52:54 +00:00
Terry Wilson
9cdc5468e7 Don't crash on a guest directmedia call
A sip_pvt may not have relatedpeer set if a call doesn't match up
with a peer. If there is no relatedpeer, there is no direct media
ACL to apply, so just return that it is allowed.

(closes issue ASTERISK-20040)
Reported by: Terry Wilson
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369217 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-22 20:05:22 +00:00
Kinsey Moore
eaf8d8a0d8 Fix wrong variable name in the R2 disconnect callback
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369216 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-22 19:54:41 +00:00
Kinsey Moore
35c7b65475 Don't parse media stream state for SIP video streams
The sendonly/recvonly/sendrecv/inactive media stream attributes were
parsed for video, but nothing was ever done with them.  With this code
removed, an UNSUPPORTED message is produced when these attributes are
used in conjunction with a video stream which is the better behavior
since they were never really supported in the first place.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369207 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-22 17:25:06 +00:00
Kinsey Moore
6a1843bbd0 Add HANGUPCAUSE hash implementation for DAHDI MFC/R2 subtech
This adds a minimal implementation of the "Who Hung Up?" Asterisk 11
work to chan_dahdi.c for the MFC/R2 DAHDI subtech.  Given the way that
OpenR2 interfaces with chan_dahdi, it is much harder to expose the
type of protocol information that is available in PRI, SS7, or other
channel technologies.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369190 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-22 15:57:02 +00:00
Kinsey Moore
1ab47ac137 Add HANGUPCAUSE hash support for analog and PRI DAHDI subtechs
This is part of the DAHDI support for the Asterisk 11 "Who Hung Up?"
project and covers the implementation for the technologies implemented
in sig_analog.c and sig_pri.c. Tested on a local machine to verify
protocol and cause information is available.

Review: https://reviewboard.asterisk.org/r/1953/
(issue SWP-4222)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369185 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-22 15:10:38 +00:00
Kinsey Moore
dee5d6b9e5 Add "Who Hung Up?" implementation for DAHDI SS7 subtechnology
Testing was done on a local machine to verify that protocol and
cause information was being sent properly.

Review: https://reviewboard.asterisk.org/r/1955/
(issue SWP-4222)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369184 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-22 14:57:07 +00:00
Richard Mudgett
a4b545222f Don't waste time initializing the whole call_identifer_str[].
The array is either setup with a callid string or only the first element
needs to be initialized.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369167 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-20 21:33:11 +00:00
Richard Mudgett
c11c6b6cb0 Fix chan_misdn compile error.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369166 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-20 21:32:40 +00:00
Alexandr Anikin
b5e299ca2e fix locking issue on empty callList
(issue ASTERISK-19298)
Reported by:
        Dmitry Melekhov
Patches:
        ASTERISK-18322-2.patch
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369148 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-20 17:48:20 +00:00
Sean Bright
5837ef8124 Remove declaration of eivr_connect_socket because it no longer exists.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369142 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-20 11:47:12 +00:00
Alexandr Anikin
5db7a15a0b use right definition for channel name
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369141 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-20 11:20:05 +00:00
Michael L. Young
babc0983e8 Add IPv6 Support To Manager
This patch adds IPv6 support to AMI.

(Closes issue ASTERISK-19965)
Reported by: Michael L. Young
Tested by: Michael L. Young
Patches:
    ami_ipv6_v3.diff uploaded by Michael L. Young (license 5026)

Review: https://reviewboard.asterisk.org/r/1968/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369126 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-20 03:18:50 +00:00
Michael L. Young
c843dddf80 Fix NULL pointer segfault in ast_sockaddr_parse()
While working with ast_parse_arg() to perform a validity check, a segfault
occurred.  The segfault occurred due to passing a NULL pointer to
ast_sockaddr_parse() from ast_parse_arg().  According to the documentation in
config.h, "result pointer to the result.  NULL is valid here, and can be used to
perform only the validity checks."

This patch fixes the segfault by checking for a NULL pointer.  This patch also
adds documentation to netsock2.h about why it is necessary to check for a NULL
pointer.

(Closes issue ASTERISK-20006)
Reported by: Michael L. Young
Tested by: Michael L. Young
Patches:
asterisk-20006-netsock-null-ptr.diff uploaded by Michael L. Young (license 5026)

Review: https://reviewboard.asterisk.org/r/1990/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369110 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-20 02:07:00 +00:00
Alexandr Anikin
2428e19436 check rtptimeouts in ooh323 channels as per config file
(rtp voice, video, udptl except rtcp)

(closes issue ASTERISK-19179)
Reported by: TSAREGORODTSEV Yury
Patches:
        19179-ooh323-ast10.patch
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369092 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-19 23:36:43 +00:00
Kinsey Moore
f080be134e Ensure that pvt cause information does not break native bridging
Channel drivers that allow native bridging need to handle
AST_CONTROL_PVT_CAUSE_CODE frames and previously did not handle them
properly, usually breaking out of the native bridge. This change
corrects that behavior and exposes the available cause code information
to the dialplan while native bridges are in place. This required
exposing the HANGUPCAUSE hash setter outside of channel.c, so
additional documentation has been added.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369086 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-19 21:13:41 +00:00
Mark Michelson
91157d5c2b Fix request routing issue when outboundproxy is used.
Asterisk was incorrectly setting the destination of CANCELs
and ACKs for error responses to the URI of the initial INVITE.
This resulted in further requests, such as INVITEs with authentication
credentials, to be routed incorrectly. Instead, when these CANCEL
or ACKs are to be sent, we should simply keep the destination the
same as what it previously was. There is no need to alter it any.

(closes issue ASTERISK-20008)
Reported by Marcus Hunger
Patches:
	ASTERISK-20008.patch uploaded by Mark Michelson (license #5049)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369068 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-19 15:44:42 +00:00
Kinsey Moore
d73a1de0b1 Fix AST_CONTROL_PVT_CAUSE_CODE handling
When the IAX2 Who Hung Up? changes were added, they uncovered a bug in
the way AST_CONTROL_PVT_CAUSE_CODE was handled in
feature_request_and_dial().  This particular frame subtype was being
treated like more terminal control frames causing the function to be
exited prematurely.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369061 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-18 22:56:01 +00:00
Richard Mudgett
c30cc8fbd5 Fix monitoring calls put in a parking lot.
* Fix a regression that was introduced by -r366167 which effectively
disabled monitoring parked calls.

(closes issue ASTERISK-20012)
Reported by: sdolloff
Tested by: rmudgett
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369057 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-18 18:25:22 +00:00
Damien Wedhorn
3d38998b70 Various small chan_skinny fixes and cleanup
Added test to skinny_register to only allow device to register against
a device that is not already registered.

Addback l->device test for skinny_show_lines. Fixes segfault if a line
is configured but not configured to a device. Reverses part of r368680.

Removed redundant l->device tests in subsubstate and dumpsub. l->device
will always be valid if these routines are called. Reverses 368948 -
discussed with mjordan on irc.

Some indentation cleanup.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369034 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-15 21:18:56 +00:00
Kinsey Moore
bf6ef69702 Allow chan_sip to decline unwanted media streams
This change replaces the static array of four representable media
streams with an AST_LIST so that chan_sip can keep track of offered
media streams.  This allows chan_sip to deal with offers containing
multiple same-type streams and many other situations without rejecting
the SDP offer in its entirety, yet still generating a valid response.
This also covers cases where Asterisk can not comprehend the offer if
it is in the correct format.

Previously, chan_sip would reject SDP offers or entirely ignore
individual stream offers in an effort to be more compatible which
would often result in invalid SDP responses.

Review: https://reviewboard.asterisk.org/r/1988/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369028 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-15 17:13:20 +00:00
Jason Parker
88c9c6bef8 Fix voicemail API tests by using the correct argument order for create/destroy.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369027 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-15 16:30:58 +00:00
Kevin P. Fleming
166b4e2b30 Multiple revisions 369001-369002
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  r369001 | kpfleming | 2012-06-15 10:56:08 -0500 (Fri, 15 Jun 2012) | 11 lines
  
  Add support-level indications to many more source files.
  
  Since we now have tools that scan through the source tree looking for files
  with specific support levels, we need to ensure that every file that is
  a component of a 'core' or 'extended' module (or the main Asterisk binary)
  is explicitly marked with its support level. This patch adds support-level
  indications to many more source files in tree, but avoids adding them to
  third-party libraries that are included in the tree and to source files
  that don't end up involved in Asterisk itself.
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  r369002 | kpfleming | 2012-06-15 10:57:14 -0500 (Fri, 15 Jun 2012) | 3 lines
  
  Add a script to enable finding source files without support-levels defined.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369013 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-15 16:20:16 +00:00
Kinsey Moore
bdab2763ac Add HANGUPCAUSE hash support to IAX2
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
2012-06-15 16:17:12 +00:00
Jason Parker
ce44b98358 Remove some symbol exports that got missed in the removal of global symbols.
(issue AST-807)
(issue AST-901)
(issue AST-908)
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Merged revisions 368999 from http://svn.asterisk.org/svn/asterisk/branches/10-digiumphones


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369000 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-15 15:33:41 +00:00
Richard Mudgett
75484af169 Remove remaining properties mmichelson left laying around from phones branch merge.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368991 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-15 00:55:43 +00:00
Richard Mudgett
f8746d0009 Allow non-normal execution routines to be able to run on hungup channels.
* 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
2012-06-14 23:22:53 +00:00
Richard Mudgett
aaa591447d Make the Hangup application set a softhangup flag.
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.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368979 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-14 22:57:21 +00:00
Richard Mudgett
c5256059b8 Move vm defines to group them better.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368972 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-14 20:49:28 +00:00
Jason Parker
6334142050 Multiple revisions 368963,368965
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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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  Merged revisions 368964 from http://svn.asterisk.org/svn/asterisk/certified/branches/1.8.11
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368966 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-14 19:40:11 +00:00
Matthew Jordan
1efe727ed8 AST-2012-009: Fix crash in chan_skinny due to Key Pad Button Message handling
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368948 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-14 17:34:10 +00:00
Mark Michelson
5819278c46 Revert Makefile change to remove embedding res_adsi.so
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368929 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-14 15:28:02 +00:00
Terry Wilson
cfa0826c49 Add a post_apply callback to the Config Options API
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/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368921 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-14 13:41:47 +00:00
Terry Wilson
01307e4b7b Add filename alias support to the Config Options API
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
2012-06-14 13:35:07 +00:00
Mark Michelson
21997aa7bb Fix a deadlock that occurs when func_volume is used on a local channel.
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368900 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-13 21:17:13 +00:00
Matthew Jordan
ff0b561045 Mark res_smdi/res_adsi as 'core' supported modules
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368896 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-13 20:28:07 +00:00