asterisk/asterisk.git
6 years agoUpdate config.guess and config.sub: 2012-10-10
Tzafrir Cohen [Sun, 14 Oct 2012 12:23:46 +0000 (12:23 +0000)]
Update config.guess and config.sub: 2012-10-10

Update config.guess and config.sub to revision
fb456b34ef4aa02b95dc6be69aaa66fa94a844fb from the savannah.gnu.org git
repo. Adds support for e.g. aarch64 (ARM 64bit).

config.guess:timestamp='2012-09-25'
config.sub:timestamp='2012-10-10'
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6 years agoUpdate CREDITS
Andrew Latham [Sat, 13 Oct 2012 19:58:20 +0000 (19:58 +0000)]
Update CREDITS

Update Jean-Denis and add myself

(issue ASTERISK-20259)

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6 years agoMultiplatform Makefile Update
Andrew Latham [Sat, 13 Oct 2012 17:38:41 +0000 (17:38 +0000)]
Multiplatform Makefile Update

Paul Belanger pointed out that using sed in the Makefile is an issue with multiple platforms.  We are cleaning up the Doxygen config as a following step so I just switched the sed inplace changes to be an echo append instead.

(issue ASTERISK-20259)

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6 years agoDoxygen Clean ups
Andrew Latham [Sat, 13 Oct 2012 16:38:48 +0000 (16:38 +0000)]
Doxygen Clean ups

Add app_skel.c as an example in app.c and fix some formating for the "Dial Privacy scripts" so it actually shows up in the Doxygen output.

(issue ASTERISK-20259)

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6 years agoTest for Asterisk Version info
Andrew Latham [Sat, 13 Oct 2012 16:05:12 +0000 (16:05 +0000)]
Test for Asterisk Version info

Doxygen uses the ASTERISKVERSION as a sub header. If a SVN export is done and no .svn or .version file exists it defualts to UNKNOWN__and_probably_unsupported which is honest but not great for the online docs.  During the "make progdocs" I added a test for this and just warned and ommitted the version.

(issue ASTERISK-20259)

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6 years agoCorrect output directory
Andrew Latham [Sat, 13 Oct 2012 15:40:40 +0000 (15:40 +0000)]
Correct output directory

During testing I used an alternate output directory and mistakenly committed it.  Matt Jordan noticed and I reverted.  This is the correct setting for local output to match with all branches.

(issue ASTERISK-20259)

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6 years agoAdd licens/copyright header
Andrew Latham [Sat, 13 Oct 2012 15:14:51 +0000 (15:14 +0000)]
Add licens/copyright header

Begin update of static-http files and general clean ups.  This only adds the standard header to the files.

(issue ASTERISK-20503)

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6 years agoAdd check for Doxygen
Andrew Latham [Sat, 13 Oct 2012 15:06:39 +0000 (15:06 +0000)]
Add check for Doxygen

The autoconf configuration system had a test for DOT but not for Doxygen.  I added the test for Doxygen and did an overhaul of the Makefile check to a much simpler process.

(issue ASTERISK-20259)

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6 years agoAvoid a segfault on invalid format names
Kinsey Moore [Fri, 12 Oct 2012 21:58:29 +0000 (21:58 +0000)]
Avoid a segfault on invalid format names

If a format name was not found by ast_getformatbyname, a NULL pointer
would be passed into ast_format_rate and immediately dereferenced.
This ensures that a valid pointer is used since the structure is
already allocated on the stack.

(closes issue DPH-523)
Reported-by: Steve Pitts
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6 years agoDo not use a FILE handle when doing SIP TCP reads.
Mark Michelson [Fri, 12 Oct 2012 16:31:01 +0000 (16:31 +0000)]
Do not use a FILE handle when doing SIP TCP reads.

This is used to solve an issue where a poll on a file
descriptor does not necessarily correspond to the readiness
of a FILE handle to be read.

This change makes it so that for TCP connections, we do a
recv() on the file descriptor instead.

Because TCP does not guarantee that an entire message or even
just one single message will arrive during a read, a loop has
been introduced to ensure that we only attempt to handle a
single message at a time. The tcptls_session_instance structure
has also had an overflow buffer added to it so that if more
than one TCP message arrives in one go, there is a place to
throw the excess.

Huge thanks goes out to Walter Doekes for doing extensive review
on this change and finding edge cases where code could fail.

(closes issue ASTERISK-20212)
reported by Phil Ciccone

Review: https://reviewboard.asterisk.org/r/2123
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6 years agoAppend Doxygen to Debian packages list
Andrew Latham [Thu, 11 Oct 2012 23:40:44 +0000 (23:40 +0000)]
Append Doxygen to Debian packages list

Add Doxygen to the Debian install list.  I will check for other platforms like Red Hat

(issue ASTERISK-20259)

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6 years agoUpdate JQuery URL to recent version
Andrew Latham [Thu, 11 Oct 2012 22:43:52 +0000 (22:43 +0000)]
Update JQuery URL to recent version

The JQuery URL to version 1.4 will be removed within the life span of Asterisk 11.  This is a compatible upgrade by using the URL for 1.8.

(issue ASTERISK-20503)

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6 years agoContinue to group config files
Andrew Latham [Thu, 11 Oct 2012 22:39:02 +0000 (22:39 +0000)]
Continue to group config files

(issue ASTERISK-20259)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374888 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoCREDITS clean up
Andrew Latham [Thu, 11 Oct 2012 22:35:41 +0000 (22:35 +0000)]
CREDITS clean up

As discussed online http://lists.digium.com/pipermail/asterisk-dev/2012-October/057245.html the credits file needs some cleaning.  This is 95% whitespace with a few additions found in file headers.  Further additions should be added here instead of in the file being updated.

(issue ASTERISK-20259)

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6 years agoRevert Local testing Config
Andrew Latham [Thu, 11 Oct 2012 21:40:02 +0000 (21:40 +0000)]
Revert Local testing Config

Revert a local testing config that I made.  This was not intended to be committed.
Thank you Matt Jordan for noticing this.

(issue ASTERISK-20259)

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6 years agoFix a bug where audio on Google Voice would not work due to ignoring candidates.
Joshua Colp [Thu, 11 Oct 2012 21:19:33 +0000 (21:19 +0000)]
Fix a bug where audio on Google Voice would not work due to ignoring candidates.

Instead of ignoring parts of the message that are not known just ignore the ones
we know may be present and that would cause a problem.
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6 years agoFix an issue where outgoing calls would fail to establish audio due to ICE negotiatio...
Joshua Colp [Thu, 11 Oct 2012 16:06:28 +0000 (16:06 +0000)]
Fix an issue where outgoing calls would fail to establish audio due to ICE negotiation failures.

This change removes the requirement for ufrag and pwd in the transport stanza and also
makes us the controlling agent.

(closes issue ASTERISK-20554)
Reported by: mmichelson

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6 years agoDon't make chan_sip export global symbols.
Mark Michelson [Thu, 11 Oct 2012 15:49:02 +0000 (15:49 +0000)]
Don't make chan_sip export global symbols.

During testing, it was discovered that having chan_sip
export global symbols was problematic.

The biggest problem was that load order was affected.
Trying to use realtime could be problematic since in
all likelihood the necessary realtime driver(s) would
not be loaded before chan_sip.

In addition, it was found that it was impossible to
use the Digium Phone Module for Asterisk since it
must be loaded before chan_sip since it must hook
into chan_sip's configuration parsing.

The solution is to use a virtual table in the same
manner that other modules in Asterisk do, like
app_voicemail.

(closes issue ASTERISK-20545)
Reported by: kmoore

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6 years agoFix incorrect billing duration reported when batch mode is enabled
Matthew Jordan [Thu, 11 Oct 2012 15:44:38 +0000 (15:44 +0000)]
Fix incorrect billing duration reported when batch mode is enabled

Similar to r369351, the billing duration can be skewed when batch mode is
enabled.  This happened much more rarely than the duration, as it only
occured when the call was answered (thereby indicating an actual answer
time) and immediately hung up on (indicating a billsec of 0).  Since
a billing time of '0' can either mean that the call immediately ended
or that the CDR was improperly answered, we have to use additional information
to know whether or not we can trust the CDR billsec value.  Prior to this
patch, we looked to see if we had a valid answer time.  If we did, and
billsec was zero, we used the current time to calculate what billsec value
we could from the CDR being written.  If batch mode is enabled, this will
incorrectly report a billsec value being much greater than the actual
duration of the call.

Instead of relying on the presence of an answer time to know whether or not
we can re-calculate the billsec for the CDR, we now also use the presence
of the CDR's end time to know if we need to re-calculate or whether we can
trust the billsec value that we have.  This prevents erroneous jumps in the
billsec value, while still making sure that in the worst case, some billing
time will be calculated.

(closes issue AST-1016)
Reported by: Thomas Arimont
Tested by: Thomas Arimont
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6 years agoConsider the Google Talk content stanza name (jin:content) valid.
Joshua Colp [Thu, 11 Oct 2012 13:34:52 +0000 (13:34 +0000)]
Consider the Google Talk content stanza name (jin:content) valid.
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6 years agoapp_queue: Made pass connected line updates from the caller to ringing queue members.
Richard Mudgett [Wed, 10 Oct 2012 21:05:51 +0000 (21:05 +0000)]
app_queue: Made pass connected line updates from the caller to ringing queue members.

Party A calls Party B
Party B puts Party A on hold.
Party B calls a queue.
Ringing queue member D sees Party B identification.
Party B transfers Party A to the queue.
Queue member D does not get a connected line update for Party A.
Queue member D answers the call and still sees Party B information.

However, if Party A later transfers the call to Party C then queue member
D gets a connected line update for Party C.

* Made pass connected line updates from the caller to queue members while
the queue members are ringing.

(closes issue AST-1017)
Reported by: Thomas Arimont

(closes issue ABE-2886)
Reported by: Thomas Arimont
Tested by: rmudgett

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6 years agoFix segfault regression from r370681
Kinsey Moore [Wed, 10 Oct 2012 13:40:40 +0000 (13:40 +0000)]
Fix segfault regression from r370681

Due to usage of ast_hook_send_action, AMI action handling code should
be able to handle a NULL mansession->session.  This would cause a crash
on NULL dereference if action_originate was called from
ast_hook_send_action.

(closes issue ASTERISK-20544)
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6 years agoFix execution of 'i' extension due to uninitialized variable.
Richard Mudgett [Tue, 9 Oct 2012 22:24:10 +0000 (22:24 +0000)]
Fix execution of 'i' extension due to uninitialized variable.

The fix for ASTERISK-18243 added code that could potentially use
dst_exten[] uninitialized.  As a result the 'i' exten may not be executed
when it should.

(closes issue ASTERISK-20455)
Reported by: Richard Miller
Patches:
      pbx-1.8.16.0.diff (license #5685) patch uploaded by Richard Miller
      Made some cosmetic modifications.
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6 years agoImprove logging for DTLS-SRTP failure situations.
Joshua Colp [Tue, 9 Oct 2012 21:35:53 +0000 (21:35 +0000)]
Improve logging for DTLS-SRTP failure situations.

(closes issue ASTERISK-20487)
Reported by: mjordan
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6 years agodahdi.conf.sample: Add description for "buffers" setting.
Richard Mudgett [Mon, 8 Oct 2012 22:31:09 +0000 (22:31 +0000)]
dahdi.conf.sample: Add description for "buffers" setting.

This contains an edited version of the patch originally created by John
Bigelow.

(closes issue ASTERISK-14435)
Reported by: John Bigelow
Patches:
      buffers.patch (license #5091) patch uploaded by John Bigelow
      0001-dahdi.conf.sample-Add-description-for-buffers-settin.patch (license #5417) patch uploaded by Shaun Ruffell
      Modified
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6 years agoFix deletion of unopenable spool files.
Richard Mudgett [Mon, 8 Oct 2012 21:24:11 +0000 (21:24 +0000)]
Fix deletion of unopenable spool files.

If scan_service() cannot open the spool file, it logs a message saying
that it will delete the file and calls remove_from_queue() to do it.
However, remove_from_queue() fails to delete the spool file because struct
outgoing has not yet been fully initialized.

* Merged allocating a new struct outgoing and init_outgoing() into
new_outgoing().  Allocation is initialization.

* Made apply_outgoing() not initialize the spool filename in struct
outgoing.

* Made apply_outgoing() call ast_trim_blanks() and ast_skip_blanks()
rather than manually inlining them.

* Reduced indentation levels in apply_outgoing().

* Fixed a garbled comment in remove_from_queue().

* Reworked scan_service() to simplify it.

(closes issue ASTERISK-17231)
Reported by: David Chappell
Patches:
      spool_open_failure.diff (license #4997) patch uploaded by David Chappell
      Started with this patch.
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* Fixed some memory leaks on off nominal paths in init_outgoing() when
merging into the new_outgoing() function dealing with o->capabilities.
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6 years agoDisable ICE support by default
Matthew Jordan [Mon, 8 Oct 2012 20:39:26 +0000 (20:39 +0000)]
Disable ICE support by default

Since there are a number of legacy devices out there that fail to handle ICE
candidates properly (which is a nice way of saying something much uglier),
disable it by default.

Support for ICE candidates can be enabled in rtp.conf using the icesupport
setting.
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6 years agoResolve issues in ConfBridge regarding marked, waitmarked, and unmarked users
Matthew Jordan [Mon, 8 Oct 2012 18:48:34 +0000 (18:48 +0000)]
Resolve issues in ConfBridge regarding marked, waitmarked, and unmarked users

Thank's to Neil Tallim (flan)'s tireless testing, issue reporting, and patches
it became clear that app_confbridge had some complex logic in how it handled
interactions between marked, waitmarked, and unmarked users.  In particular,
there were some areas in which the interactions between the users resulted
in inconsistent behavior, and app_confbridge was missing logic in how to handle
some corner cases.  Some areas included:
 * Poor handling of mixing unmarked and waitmarked users
 * Inconsistencies in how MOH and muting was applied to various users
 * Handling of various announcements for different user profile options
flan's patches seem to fix the various issues, but highlighted how hard the
code could be to maintain.  In an attempt to make things easier to maintain and
to more fully enumerate the various cases that exist, this patch breaks up the
logic into a state machine-like setup.

Please note that the various state transitioned are documented on the Asterisk
wiki:

https://wiki.asterisk.org/wiki/display/AST/Confbridge+state+changes

Review: //https://reviewboard.asterisk.org/r/2072/

Note that for the following issues, mjordan uploaded the patch, although it
was written by twilson.  Any contributor license discrepency is due to that.

(closes issue ASTERISK-19562)
Reported by: flan
Tested by: flan, mjordan, jrose
patches:
  bugASTERISK-19562_ASTERISK-19726_ASTERISK-20181.patch uploaded by twilson (license 6283)

(closes issue ASTERISK-19726)
Reported by: flan
Tested by: flan
patches:
  bugASTERISK-19562_ASTERISK-19726_ASTERISK-20181.patch uploaded by twilson (license 6283)

(closes issue ASTERISK-20181)
Reported by: Jonathan White
Tested by: Jonathan White
patches:
  bugASTERISK-19562_ASTERISK-19726_ASTERISK-20181.patch uploaded by twilson (license 6283)
........

Merged revisions 374652 from http://svn.asterisk.org/svn/asterisk/branches/10
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6 years agopjproject: Fix for Solaris builds. Do not undef s_addr.
Matthew Jordan [Mon, 8 Oct 2012 00:45:36 +0000 (00:45 +0000)]
pjproject: Fix for Solaris builds. Do not undef s_addr.

pjproject, in order to solve build problems on Windows [1], undefines s_addr in
one of it's headers that is included in res_rtp_asterisk.c. On Solaris s_addr
is not a structure member, but defined to map to the real strucuture member,
therefore when building on Solaris it's possible to get build errors like:

    [CC] res_rtp_asterisk.c -> res_rtp_asterisk.o
    In file included from /export/home/admin/asterisk-11-svn/include/asterisk/stun.h:29,
                     from res_rtp_asterisk.c:51:
    /export/home/admin/asterisk-11-svn/include/asterisk/network.h: In function `inaddrcmp':
    /export/home/admin/asterisk-11-svn/include/asterisk/network.h:92: error: structure has no member named `s_addr'
    /export/home/admin/asterisk-11-svn/include/asterisk/network.h:92: error: structure has no member named `s_addr'
    res_rtp_asterisk.c: In function `ast_rtp_on_ice_tx_pkt':
    res_rtp_asterisk.c:706: warning: dereferencing type-punned pointer will break strict-aliasing rules
    res_rtp_asterisk.c:710: warning: dereferencing type-punned pointer will break strict-aliasing rules
    res_rtp_asterisk.c: In function `rtp_add_candidates_to_ice':
    res_rtp_asterisk.c:1085: error: structure has no member named `s_addr'
    make[2]: *** [res_rtp_asterisk.o] Error 1
    make[1]: *** [res] Error 2
    make[1]: Leaving directory `/export/home/admin/asterisk-11-svn'
    gmake: *** [_cleantest_all] Error 2

Unfortunately, in order to make this work, I also had to make sure pjproject
only used the typdef pj_in_addr and not the struct pj_in_addr so that when
building Asterisk I could "typedef struct in_addr pj_in_addr". It's possible
then that the library and users of those interfaces in Asterisk have a different
idea about the type of the argument, while on the surface it looks like they are
all 32 bit big endian values.

[1] http://trac.pjsip.org/repos/changeset/484

(issues ASTERISK-20366)
Reported by: Ben Klang
Tested by: Ben Klang, mjordan
patches:
  0001-pjproject-Fix-for-Solaris-builds.-Do-not-undef-s.patch uploaded by Shaun Ruffell (license 5417)
........

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6 years agoTrivial patch to make 'best_score' defined for all architectures.
Matthew Jordan [Sun, 7 Oct 2012 17:33:38 +0000 (17:33 +0000)]
Trivial patch to make 'best_score' defined for all architectures.

Fixes trivial build error on Solaris:

  acl.c: In function `get_local_address':
  acl.c:196: error: `best_score' undeclared (first use in this function)
  acl.c:196: error: (Each undeclared identifier is reported only once
  acl.c:196: error: for each function it appears in.)
  make[2]: *** [acl.o] Error 1

(issue ASTERISK-20366)
Reported by: Ben Klang
Tested by: Ben Klang
patches:
  0002-main-acl.c-Trivial.-best_score-should-be-defined-for.patch by Shaun Ruffell (license 5417)
........

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6 years agoHandle capability stanzas that fail to provide node or version information
Matthew Jordan [Sat, 6 Oct 2012 03:22:37 +0000 (03:22 +0000)]
Handle capability stanzas that fail to provide node or version information

While XEP-0115 states that the node and ver attributes are both required, some
devices fail to provide either field.  Prior to this patch, failure to provide
the node or ver attribute would cause a crash in res_xmpp.  While failing to
provide the node or ver attribute is technically invalid, since this
information is not utilized by Asterisk except for reporting purposes, for
interoperability reasons, we continue to process the capability stanza anyways.

(closes issue ASTERISK-20495)
Reported by: Martin W
Tested by: Martin W
patches:
  20495.patch uploaded by Martin W (license #6434)
........

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6 years agoUpdate documentation for MessageSend application/command's From field for XMPP
Matthew Jordan [Sat, 6 Oct 2012 01:47:00 +0000 (01:47 +0000)]
Update documentation for MessageSend application/command's From field for XMPP

When using the channel technology agnostic application/AMI command MessageSend,
the "From" field is technically optional for the SIP channel driver.  However,
if being sent by the XMPP resource module (either res_xmpp or res_jabber), the
"From" field is necessary, and must correspond to a defined account.  This
patch updates the documentation for this application/AMI command to reflect
this.

(closes issue ASTERISK-20405)
Reported by: Leif Madsen
........

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6 years agoMultiple revisions 374570,374581
David M. Lee [Fri, 5 Oct 2012 20:33:56 +0000 (20:33 +0000)]
Multiple revisions 374570,374581

........
  r374570 | dlee | 2012-10-05 15:14:41 -0500 (Fri, 05 Oct 2012) | 22 lines

  Improve AMI long line error handling

  In AMI's parser, when it receives a long line (> 1024 characters), it discards
  that line, but continues to process the message normally.

  Typically, this is not a problem because a) who has lines that long and b)
  usually a discarded line results in an invalid message. But if that line is
  specifying an optional field, then the message will be processed, you get a
  'Response: Success', but things don't work the way you expected them to.

  This patch changes the behavior when a line-too-long parse error occurs.

  * Changes the log message to avoid way-too-long (and truncated anyways) log
    messages
  * Adds a 'parsing' status flag to Response: Success
  * Sets parsing = MESSAGE_LINE_TOO_LONG if, well, a line is too long
  * Responds with an appropriate error if parsing != MESSAGE_OKAY

  (closes issue AST-961)
  Reported by: John Bigelow
  Review: https://reviewboard.asterisk.org/r/2142/
........
  r374581 | dlee | 2012-10-05 15:20:28 -0500 (Fri, 05 Oct 2012) | 1 line

  I've committed too much. Reverting part of r374570.
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6 years agoMerged revisions 374515-374535 from
Richard Mudgett [Fri, 5 Oct 2012 18:42:14 +0000 (18:42 +0000)]
Merged revisions 374515-374535 from
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier

................
  r374515 | rmudgett | 2012-10-04 17:52:36 -0500 (Thu, 04 Oct 2012) | 10 lines

  chan_misdn: Remove some deadcode

  * Made setup_bc() static.

  Patches:
patch1_unused-code.diff (license #6372) patch uploaded by Guenther Kelleter
Modified

  JIRA ABE-2882

................
  r374516 | rmudgett | 2012-10-04 18:01:01 -0500 (Thu, 04 Oct 2012) | 7 lines

  chan_misdn: Remove unused bchan states

  Patches:
patch2_unused-states.diff (license #6372) patch uploaded by Guenther Kelleter

  JIRA ABE-2882

................
  r374517 | rmudgett | 2012-10-04 18:17:51 -0500 (Thu, 04 Oct 2012) | 16 lines

  chan_misdn: Remove unnecessary null pointer checks and checks for stack->nt

  * cleanup_bc() is always called with valid bc (or it would've crashed
  before).

  * Value of stack->nt is known in advance at some places.

  * Rename handle_event() to handle_event_te(), handle_frm() to
  handle_frm_te().

  Patches:
patch3_checks.diff (license #6372) patch uploaded by Guenther Kelleter
Modified

  JIRA ABE-2882

................
  r374518 | rmudgett | 2012-10-04 18:21:59 -0500 (Thu, 04 Oct 2012) | 7 lines

  chan_misdn: Fix spelling in log messages

  Patches:
patch4_spelling.diff (license #6372) patch uploaded by Guenther Kelleter

  JIRA ABE-2882

................
  r374519 | rmudgett | 2012-10-04 18:31:59 -0500 (Thu, 04 Oct 2012) | 15 lines

  chan_misdn: Don't cleanup a bc twice.

  In handle_frm_te() after calling misdn_lib_send_event(bc,
  EVENT_RELEASE_COMPLETE) bc is emptied, cleaned and set not in use,
  although misdn_lib_send_event() already did the same.  This is bad.  When
  it's not in use we are not allowed to touch it.

  * Moved log message in front of the resulting actions and fixed it to
  match the case.

  Patches:
patch5_bccleanup.diff (license #6372) patch uploaded by Guenther Kelleter

  JIRA ABE-2882

................
  r374520 | rmudgett | 2012-10-04 18:43:56 -0500 (Thu, 04 Oct 2012) | 12 lines

  chan_misdn: Fix memory leaks, bc, chan not cleaned up etc., really bad stuff.

  * Fix return codes of cb_events() for EVENT_SETUP to use caller's cleanup
  mechanisms.

  * Move cl_queue_chan() call after bearer check.

  Patches:
patch6_leaks.diff (license #6372) patch uploaded by Guenther Kelleter

  JIRA ABE-2882

................
  r374521 | rmudgett | 2012-10-04 18:48:38 -0500 (Thu, 04 Oct 2012) | 11 lines

  chan_misdn: We must initialize cause on sending a DISCONNECT.

  We must initialize cause on sending a DISCONNECT, so it is later correctly
  indicated to ast_channel in case the answer (RELEASE/RELEASE_COMPLETE)
  does not include one.

  Patches:
patch7_hangupcause.diff (license #6372) patch uploaded by Guenther Kelleter

  JIRA ABE-2882

................
  r374522 | rmudgett | 2012-10-04 19:03:56 -0500 (Thu, 04 Oct 2012) | 7 lines

  chan_misdn: Remove unused code for upqueue

  Patches:
patch8_unused-upqueue.diff (license #6372) patch uploaded by Guenther Kelleter

  JIRA ABE-2882

................
  r374523 | rmudgett | 2012-10-04 19:11:50 -0500 (Thu, 04 Oct 2012) | 7 lines

  chan_misdn: Improve debugging (port number, messages fixed, dups removed)

  Patches:
patch9_debug.diff (license #6372) patch uploaded by Guenther Kelleter

  JIRA ABE-2882

................
  r374533 | rmudgett | 2012-10-05 12:17:18 -0500 (Fri, 05 Oct 2012) | 8 lines

  chan_misdn: Better debug: we can print_bc_info even if there's no ast leg.

  Patches:
patch10_debug-bc-2.diff (license #6372) patch uploaded by Guenther Kelleter
Modified.

  JIRA ABE-2882

................
  r374534 | rmudgett | 2012-10-05 12:34:10 -0500 (Fri, 05 Oct 2012) | 16 lines

  chan_misdn: setup_bc() is called too early for an incoming SETUP on TE.

  This prevents the B channel from being setup for HDLC mode when requested
  by the bearer capability and config option hdlc=yes.  It violates
  ETS300102 Ch.5.2.3.2: "The user, in any case, must not connect to the
  channel until a CONNECT ACKNOWLEDGE message has been received."

  * Call setup_bc() on receipt of CONNECT_ACKNOWLEGDE for PTMP, and on first
  response to SETUP for PTP.

  Patches:
abe-2881-2.diff (license #6372) patch uploaded by Guenther Kelleter
Modified.

  JIRA ABE-2881

................
  r374535 | rmudgett | 2012-10-05 12:41:05 -0500 (Fri, 05 Oct 2012) | 2 lines

  chan_misdn: Remove some more deadcode.

................
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6 years agodsp.c User Configurable DTMF_HITS_TO_BEGIN and DTMF_MISSES_TO_END
Alec L Davis [Thu, 4 Oct 2012 20:21:36 +0000 (20:21 +0000)]
dsp.c User Configurable DTMF_HITS_TO_BEGIN and DTMF_MISSES_TO_END

Instead of a recompile, allow values to be adjusted in dsp.conf

For binary distributions allows easy adjustment for wobbly GSM calls, and other reasons.

Defaults to DTMF_HITS_TO_BEGIN=2 and DTMF_MISSES_TO_END=3

(closes issue ASTERISK-17493)
Reported by: alecdavis
Tested by: alecdavis
alecdavis (license 585)

Review https://reviewboard.asterisk.org/r/2144/
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6 years agodsp.c fix incorrect DTMF Digit_Duration.
Alec L Davis [Thu, 4 Oct 2012 20:08:22 +0000 (20:08 +0000)]
dsp.c fix incorrect DTMF Digit_Duration.

it's always short by 'hits_to_begin*DTMF_GSIZE', or 25.5ms if hitstobegin=2

(issue ASTERISK-16003)
Tested by: alecdavis
alecdavis (license 585)

Review https://reviewboard.asterisk.org/r/2145/
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6 years agoFix DBDelTree error codes for AMI, CLI and AGI
David M. Lee [Thu, 4 Oct 2012 15:48:24 +0000 (15:48 +0000)]
Fix DBDelTree error codes for AMI, CLI and AGI

The AMI DBDelTree command will return Success/Key tree deleted successfully even
if the given key does not exist. The CLI command 'database deltree' had a
similar problem, but was saved because it actually responded with '0 database
entries removed'. AGI had a slightly different error, where it would return
success if the database was unavailable.

This came from confusion about the ast_db_deltree retval, which is -1 in the
event of a database error, or number of entries deleted (including 0 for
deleting nothing).

* Changed some poorly named res variables to num_deleted
* Specified specific errors when calling ast_db_deltree (database unavailable
  vs. entry not found vs. success)
* Fixed similar bug in AGI database deltree, where 'Database unavailable'
  results in successful result

(closes issue AST-967)
Reported by: John Bigelow
Review: https://reviewboard.asterisk.org/r/2138/
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6 years agoAdd support for applying direct media ACLs between differing channel technologies.
Joshua Colp [Thu, 4 Oct 2012 13:49:45 +0000 (13:49 +0000)]
Add support for applying direct media ACLs between differing channel technologies.

Review: https://reviewboard.asterisk.org/r/2122/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374414 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agodsp.c User configuration of DTMF_NORMAL_TWIST and DTMF_REVERSE_TWIST values
Alec L Davis [Thu, 4 Oct 2012 04:50:16 +0000 (04:50 +0000)]
dsp.c User configuration of DTMF_NORMAL_TWIST and DTMF_REVERSE_TWIST values

Asterisk's DTMF Specifications are based on AT&T specs, which may not be compatible in other countries.

Various countries have different specifications for the maximum power level differences
between the DTMF low group and high group of frequencies.

Power level difference between frequencies for different Administrations/RPOAs
 NTT        = Max. 5 dB
 AT&T       = 4dB(reverse) to 8dB(normal)
 Danish     = Max. 6 dB
 Australian = Max. 10 dB
 Brazilian  = Max. 9 dB
 ETSI       = Max. 6 dB from ETSI ES 201 235-3 V1.3.1 (2006-03)

Now allow 4 variables to be individually configured in dsp.conf, with reasonable min/max of 2dB to 20dB.
Default is AT&T specifications

Add's the following variables to dsp.conf
;dtmf_normal_twist=6.31
;dtmf_reverse_twist=2.51
;relax_dtmf_normal_twist=6.31
;relax_dtmf_reverse_twist=3.98

(closes issue ASTERISK-20442)
Reported by: tbsky
Tested by: tbsky,alecdavis

alecdavis (license 585)

Review https://reviewboard.asterisk.org/r/2141/
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Merged revisions 374384 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 374385 from http://svn.asterisk.org/svn/asterisk/branches/10
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Merged revisions 374386 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374387 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoCheck for presence of buddy in info/dinfo handlers
Matthew Jordan [Thu, 4 Oct 2012 02:16:43 +0000 (02:16 +0000)]
Check for presence of buddy in info/dinfo handlers

The res_jabber resource module uses the ASTOBJ library for managing its ref
counted objects.  After calling ASTOBJ_CONTAINER_FIND to locate a buddy object,
the pointer to the object has to be checked to see if the buddy existed.
Prior to this patch, the buddy object was not checked for NULL; with this patch
in both aji_client_info_handler and aji_dinfo_handler the pointer is checked
before used and, if no buddy object was found, the handlers return an error
code.

This patch does not take the approach that our JID can be used to log in from
another resource.  If that approach is desired, an improvement could be made to
this patch to create the buddy on the fly.  This patch seeks only to prevent
Asterisk from crashing.

FYI: In Asterisk 11+, you really should be using res_xmpp.  It does not have
this problem, as it moved to the astobj2 library.

Note that multiple people have proposed patches for this issue; the patch being
committed here is based on those.

(closes issue ASTERISK-19532)
Reported by: Karsten Wemheuer
Tested by: Byron Clark
patches:
  fix-jabber uploaded by Karsten Wemheuer (license #5930)
  xmpp_no_crash_with_ejabberd.patch uploaded by Byron Clark (license #6157)

(closes issue ASTERISK-19557)
Reported by: ulugutz
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Merged revisions 374335 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 374336 from http://svn.asterisk.org/svn/asterisk/branches/10
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Merged revisions 374337 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374338 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoDestroy the generic_monitors container after the core_instances in ccss
Matthew Jordan [Wed, 3 Oct 2012 17:27:53 +0000 (17:27 +0000)]
Destroy the generic_monitors container after the core_instances in ccss

For each item in core_instances disposed of in the shutdown of ccss, any
generic monitor instances referenced by the objects will be removed from
generic_monitors during their destruction.  Hilarity ensues if
generic_monitors no longer exists.

Thanks to the Asterisk Test Suite's generic_ccss test for complaining loudly
when it ran into this.
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Merged revisions 374300 from http://svn.asterisk.org/svn/asterisk/branches/10
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Merged revisions 374301 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374302 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoMissed an astobj2.c debug tag.
Richard Mudgett [Tue, 2 Oct 2012 23:23:30 +0000 (23:23 +0000)]
Missed an astobj2.c debug tag.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374279 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years ago* Add ref debug tags to astobj2.c ref usage.
Richard Mudgett [Tue, 2 Oct 2012 22:39:47 +0000 (22:39 +0000)]
* Add ref debug tags to astobj2.c ref usage.

* Make container nodes not show up in the ref debug log.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374269 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoEnsure Shutdown AMI event is still fired during Asterisk shutdown
Matthew Jordan [Tue, 2 Oct 2012 21:26:27 +0000 (21:26 +0000)]
Ensure Shutdown AMI event is still fired during Asterisk shutdown

Richard pointed out that having the manager dispose of itself gracefully
during shutdown meant that the Shutdown event will no longer get fired.
This patch moves the AMI event just prior to running the atexit callbacks.
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Merged revisions 374230 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 374248 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374259 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoModify hashtest2 to compile after r374213. Someone, somewhere, may care.
Matthew Jordan [Tue, 2 Oct 2012 20:45:22 +0000 (20:45 +0000)]
Modify hashtest2 to compile after r374213.  Someone, somewhere, may care.

Because hashtest2 has to provide symbols for things in asterisk that items
it includes may use, when astobj2 decided to use ast_register_atexit it needed
to provide a declaration for that as well.  Otherwise - no linky.

On a related note, ASTERISK-20505 was filed to convert hashtest/hashtest2 into
actual unit tests, so we don't run into this problem again.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374229 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoFix findings from check-in on r374177
Matthew Jordan [Tue, 2 Oct 2012 17:16:20 +0000 (17:16 +0000)]
Fix findings from check-in on r374177

Richard pointed out two problems with the check-in from r374177:
* The ast_msg_shutdown function declaration doesn't match the prototype
  in main/message.c.
* The ref/alloc function usage in astobj2 (in trunk) can use the ao2_t_*
  variants of the functions to allow the REF_DEBUG flag to enable/disable
  their debug counterparts.
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Merged revisions 374210 from http://svn.asterisk.org/svn/asterisk/branches/10
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Merged revisions 374211 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374213 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoFix a variety of ref counting issues
Matthew Jordan [Tue, 2 Oct 2012 01:47:16 +0000 (01:47 +0000)]
Fix a variety of ref counting issues

This patch resolves a number of ref leaks that occur primarily on Asterisk
shutdown.  It adds a variety of shutdown routines to core portions of
Asterisk such that they can reclaim resources allocate duringd initialization.

Review: https://reviewboard.asterisk.org/r/2137
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Merged revisions 374177 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 374178 from http://svn.asterisk.org/svn/asterisk/branches/10
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Merged revisions 374196 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374197 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoDoxygen Cleanup
Andrew Latham [Mon, 1 Oct 2012 23:39:45 +0000 (23:39 +0000)]
Doxygen Cleanup

Start adding configuration file linking and pages.  Add module loading doxygen block.

Breaking up commits to keep it easy to track

(issue ASTERISK-20259)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374167 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoDoxygen Cleanup
Andrew Latham [Mon, 1 Oct 2012 23:24:35 +0000 (23:24 +0000)]
Doxygen Cleanup

Start adding configuration file linking and pages.  Add module loading doxygen block.

Breaking up commits to keep it easy to track

(issue ASTERISK-20259)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374166 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoDoxygen Cleanup
Andrew Latham [Mon, 1 Oct 2012 23:24:10 +0000 (23:24 +0000)]
Doxygen Cleanup

Start adding configuration file linking and pages.  Add module loading doxygen block.

Breaking up commits to keep it easy to track

(issue ASTERISK-20259)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374165 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoDoxygen Cleanup
Andrew Latham [Mon, 1 Oct 2012 23:22:50 +0000 (23:22 +0000)]
Doxygen Cleanup

Start adding configuration file linking and pages.  Add module loading doxygen block.

(issue ASTERISK-20259)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374164 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoapp_queue: Support persisting and loading of long member lists.
Sean Bright [Mon, 1 Oct 2012 20:36:25 +0000 (20:36 +0000)]
app_queue: Support persisting and loading of long member lists.

Greenlight in #asterisk brought up that he was receiving an error message "Could
not create persistent member string, out of space" when running app_queue in
Asterisk 10.  dump_queue_members() made an assumption that 8K would be enough to
store the generated string, but with queues that have large member lists this is
not always the case.  This patch removes the limitation and uses ast_str instead
of a fixed sized buffer.

The complicating factor comes from the fact that ast_db_get requires a buffer
and buffer size argument, which doesn't let us pull back more than what we pass
in, so I introduced a new ast_db_get_allocated() which returns an ast_strdup()'d
copy of the value from astdb.

As an aside, I did some testing on the maximum size of data that we can store in
the BDB library we distribute and was able to store a 10MB string and retrieve
it with no problems, so I feel this is a safe patch.

Review: https://reviewboard.asterisk.org/r/2136/
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Merged revisions 374108 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 374135 from http://svn.asterisk.org/svn/asterisk/branches/10
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Merged revisions 374150 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374151 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoUse ast_copy_string instead of strncpy to guarantee a NUL terminated string.
Sean Bright [Mon, 1 Oct 2012 17:28:41 +0000 (17:28 +0000)]
Use ast_copy_string instead of strncpy to guarantee a NUL terminated string.
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Merged revisions 374132 from http://svn.asterisk.org/svn/asterisk/branches/10
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Merged revisions 374133 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374134 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoChange core show help output format.
Richard Mudgett [Mon, 1 Oct 2012 17:05:37 +0000 (17:05 +0000)]
Change core show help output format.

The CLI "core show help" output leaves something to be desired.
1) The command is truncated to a maximum of 30 characters.
2) The output columns are mirrored from the 31st column.

Current output format:
                   logger mute Toggle logging output to a console
                 logger reload Reopens the log files
                 logger rotate Rotates and reopens the log files
logger set level {DEBUG|NOTICE Enables/Disables a specific logging level for this console
          logger show channels List configured log channels

New format:
logger mute                    -- Toggle logging output to a console
logger reload                  -- Reopens the log files
logger rotate                  -- Rotates and reopens the log files
logger set level {DEBUG|NOTICE|WARNING|ERROR|VERBOSE|DTMF} {on|off} -- Enables/Disables a specific logging level for this console
logger show channels           -- List configured log channels

Review: https://reviewboard.asterisk.org/r/2133/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374109 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoDon't destroy confbridge config when error is encountered during a reload.
Mark Michelson [Mon, 1 Oct 2012 16:26:23 +0000 (16:26 +0000)]
Don't destroy confbridge config when error is encountered during a reload.

Not panicking means that the old config is kept.

(closes issue ASTERISK-20458)
Reported by: Leif Madsen
Patches:
ASTERISK-20458.patch uploaded by Mark Michelson(license #5049)
Tested by Leif Madsen
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Merged revisions 374106 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374107 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoAdd support for retrieving engine specific settings using the speech API and from...
Joshua Colp [Mon, 1 Oct 2012 12:29:04 +0000 (12:29 +0000)]
Add support for retrieving engine specific settings using the speech API and from dialplan.

(closes issue ASTERISK-17136)
Reported by: kenner

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374096 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoFix ref leak when adding ICE candidates to an SDP
Matthew Jordan [Sat, 29 Sep 2012 03:56:49 +0000 (03:56 +0000)]
Fix ref leak when adding ICE candidates to an SDP

There was a missing decrement to the reference count for the current ICE
candidate when local candidates are being added to an outbound SDP.  This
patch corrects that.
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Merged revisions 374085 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374086 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoInclude channel uniqueid in "AsyncAGI" and "AGIExec" events.
Richard Mudgett [Fri, 28 Sep 2012 22:11:19 +0000 (22:11 +0000)]
Include channel uniqueid in "AsyncAGI" and "AGIExec" events.

* Added AMI event documentation for AsyncAGI and AGIExec events.

(closes issue ASTERISK-20318)
Reported by: Dan Cropp
Patches:
      res_agi_patch.txt (license #6422) patch uploaded by Dan Cropp
      modified for trunk.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374075 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agores_jabber: Remove CLI command 'jabber test'
Jonathan Rose [Fri, 28 Sep 2012 19:37:22 +0000 (19:37 +0000)]
res_jabber: Remove CLI command 'jabber test'

The opinion of development was that it is both improper to have Matt's
personal email address used in the source and that the command wouldn't
be useful without it.

(closes issue AST-467)
Reported by: Malcolm Davenport
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Merged revisions 374032 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 374045 from http://svn.asterisk.org/svn/asterisk/branches/10
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Merged revisions 374059 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374060 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoAdd pause one second W dial modifier.
Richard Mudgett [Fri, 28 Sep 2012 18:27:02 +0000 (18:27 +0000)]
Add pause one second W dial modifier.

* The following dialplan applications now recognize 'W' to pause sending
DTMF for one second in addition to the previously existing 'w' that paused
sending DTMF for half a second.  Dial, ExternalIVR, and SendDTMF.

* The chan_dahdi analog port dialing and deferred DTMF dialing for PRI now
distinguishes between 'w' and 'W'.  The 'w' pauses dialing for half a
second.  The 'W' pauses dialing for one second.

* Created dahdi_dial_str() in chan_dahdi that eliminated a lot of
duplicated dialing code and diagnostic messages for the channel driver.

(closes issue ASTERISK-20039)
Reported by: Jeremiah Gowdy
Patches:
      jgowdy-wait-6-22-2012.diff (license #5621) patch uploaded by Jeremiah Gowdy
      Expanded patch to add support in chan_dahdi.
Tested by: rmudgett

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374030 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoReset hangup flags on channels created through messages and cleanup globals
Brent Eagles [Fri, 28 Sep 2012 13:04:11 +0000 (13:04 +0000)]
Reset hangup flags on channels created through messages and cleanup globals
in res_xmpp on unload.

This patch fixes an issue where hangup flags were not being reset on a
channel, affecting subsequent use of that channel. The patch also adds some
additional cleanup to res_xmpp to fix an issue with reloading the module.

(closes ASTERISK-20360)
Reported by: Noah Engelberth
Tested by: beagles
Review: https://reviewboard.asterisk.org/r/2134/
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Merged revisions 374019 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374020 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoUpdate documentation to make it explicit that "stream file" will not restart musiconhold.
Joshua Colp [Fri, 28 Sep 2012 12:17:41 +0000 (12:17 +0000)]
Update documentation to make it explicit that "stream file" will not restart musiconhold.

(issue ASTERISK-17367)
Reported by: oej
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Merged revisions 373989 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 373990 from http://svn.asterisk.org/svn/asterisk/branches/10
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Merged revisions 373991 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373992 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoAdd Duration header for PlayDTMF AMI Action
Matthew Jordan [Fri, 28 Sep 2012 03:06:53 +0000 (03:06 +0000)]
Add Duration header for PlayDTMF AMI Action

This patch adds an optional header to the PlayDTMF AMI action, Duration.
It allows the duration of the DTMF digit to be played on the channel to be
specified in milliseconds.

(closes issue ASTERISK-18172)
Reported by: Renato dos Santos

patches:
  send-dtmf.patch uploaded by Renato dos Santos (license #6267)

Modified slightly for this commit for Asterisk 12.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373979 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoTweak app_dial documentation.
Richard Mudgett [Thu, 27 Sep 2012 22:43:27 +0000 (22:43 +0000)]
Tweak app_dial documentation.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373967 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoCleanup ast_dtmf_stream()
Richard Mudgett [Thu, 27 Sep 2012 22:33:15 +0000 (22:33 +0000)]
Cleanup ast_dtmf_stream()

* Made ast_dtmf_stream() wait after starting the silence generator rather
than before.

* Made ast_dtmf_stream() put the peer in autoservice for the whole time
things are being done to the chan.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373966 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoFix SendDTMF crash and channel reference leak using channel name parameter.
Richard Mudgett [Thu, 27 Sep 2012 22:25:34 +0000 (22:25 +0000)]
Fix SendDTMF crash and channel reference leak using channel name parameter.

The SendDTMF channel name parameter has two issues.
1) Crashes if the channel name does not exist.
2) Leaks a channel reference if the channel is the current channel.
Problem introduced by ASTERISK-15956.

* Updated SendDTMF documentation.

* Renamed app to senddtmf_name and tweaked the type.
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6 years agoMake res_http_websocket an optional dependency on supported platforms for chan_sip.
Joshua Colp [Thu, 27 Sep 2012 17:12:08 +0000 (17:12 +0000)]
Make res_http_websocket an optional dependency on supported platforms for chan_sip.

(closes issue ASTERISK-20439)
Reported by: sruffell
Patches:
     0001-chan_sip-websocket-support-is-an-optional-API.patch uploaded by sruffell (license 5417)
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6 years agoAdd VoicemailRefresh AMI Action
Kinsey Moore [Thu, 27 Sep 2012 17:02:13 +0000 (17:02 +0000)]
Add VoicemailRefresh AMI Action

Currently, if there are modifications to mailboxes that Asterisk is
not aware of, the user needs to add "pollmailboxes" to their mailbox
configuration, which repeatedly polls the subscribed mailboxes for
changes. This results in a lot of extra work for the CPU. This patch
introduces the AMI command VoicemailRefresh which permits external
applications to trigger the refresh themselves. The refresh can apply
to a specified mailbox only, an entire context, or all configured
mailboxes. Even a refresh performed on every mailbox would not consume
as much CPU as the pollmailboxes option, given that pollmailboxes runs
continuously and this only runs on demand.

(closes issue ASTERISK-17206)
(closes issue ASTERISK-19908)
Reported-by: Jeff Hutchins
Reported-by: Tilghman Lesher
Patch-by: Tilghman Lesher

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373913 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoloader: Ensure dependent modules are properly initialized.
Joshua Colp [Thu, 27 Sep 2012 16:53:19 +0000 (16:53 +0000)]
loader: Ensure dependent modules are properly initialized.

If an Asterisk module specifies a dependency in ast_module_info.nonoptreq, it
is possible for Asterisk to skip calling the modules's .load function.
Asterisk was loading and linking the module via load_dynamic_module() but was
not adding the module to the resource_heap. Therefore the module was not
initialized based on it's priority along with the other modules in the heap.

Now use load_resource() instead of load_dynamic_module() for non-optional
requirement. This will add the module to the resource_heap so the module can
be properly initialized in the correct order.

This is required if there are any module global data structures initialized in
the .load() callback for the module on platforms which do not support weak
references.

(issue ASTERISK-20439)
Reported by: sruffell
Patches:
     0001-loader-Ensure-dependent-modules-are-properly-initial.patch uploaded by sruffell (license 5417)
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6 years agoFix an issue where Local channels dialed by app_queue are considered in use immediately.
Joshua Colp [Thu, 27 Sep 2012 11:33:54 +0000 (11:33 +0000)]
Fix an issue where Local channels dialed by app_queue are considered in use immediately.

The chan_local channel driver returns a device state of in use even if a created Local
channel has not yet been dialed. This fix changes the logic to return a state of not
in use until the channel itself has been dialed.

(closes issue ASTERISK-20390)
Reported by: tim_ringenbach

Review: https://reviewboard.asterisk.org/r/2116/
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6 years agoMove handling of 408 response so there is no misleading warning message.
Mark Michelson [Wed, 26 Sep 2012 21:17:16 +0000 (21:17 +0000)]
Move handling of 408 response so there is no misleading warning message.

(closes issue ASTERISK-20060)
Reported by: Walter Doekes
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373852 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoFixed meetme tab completion and command documentation.
Richard Mudgett [Wed, 26 Sep 2012 18:23:37 +0000 (18:23 +0000)]
Fixed meetme tab completion and command documentation.

* Removed unnecessary case sensitivity in meetme list, lock, unlock, mute,
unmute, and kick commands.

* Separated meetme lock/unlock, mute/unmute, and kick commands into their
own registered commands to simplify tab completion and parameter checking.
meetme_lock_cmd(), meetme_mute_cmd(), and meetme_kick_cmd()

* Simplified meetme_show_cmd()

(closes issue AST-1006)
Reported by: John Bigelow
Tested by: rmudgett
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6 years agoapp_queue: 'agent available' hint, cleanup restart, and initial state
Alec L Davis [Wed, 26 Sep 2012 08:31:46 +0000 (08:31 +0000)]
app_queue: 'agent available' hint, cleanup restart, and initial state

Fix previously untested senarios;

1). On queue initialisation set queue_avail devstate to INUSE.
    Previously was unavailable, which indicated an agent was available.

2). When removing members, if there are no other members available, set queue_avail to INUSE.
    Previously, if a member interface had become 'unavailable', they were never going to be removed, particularly when persistant queues is enabled.

3). When adding a member, check that they are available, if they are set queue_avail to NOT_INUSE.
 Previously on reloaded, members may have been 'unavailable'.

4). When pausing or unpausing a member, set appropriate queue availability.

alecdavis (license 585)
Reported by: Alec Davis
Tested by: alecdavis

Review: https://reviewboard.asterisk.org/r/2129/
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6 years agoFix saying of date in Dutch.
Mark Michelson [Tue, 25 Sep 2012 23:10:22 +0000 (23:10 +0000)]
Fix saying of date in Dutch.

The Dutch say the date before the month.

(closes issue ASTERISK-20353)
Reported by: Teun Ouwehand
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6 years agoRemove dead code and documentation for nonexistent feature.
Mark Michelson [Tue, 25 Sep 2012 22:57:56 +0000 (22:57 +0000)]
Remove dead code and documentation for nonexistent feature.

multiplelogin was removed from chan_agent back in 1.6.0 when
AgentCallbackLogin() was removed.

(closes issue AST-948)
reported by Steve Pitts
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373771 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoFix error where improper IMAP greetings would be deleted.
Mark Michelson [Tue, 25 Sep 2012 21:14:21 +0000 (21:14 +0000)]
Fix error where improper IMAP greetings would be deleted.

(closes issue ASTERISK-20435)
Reported by: fhackenberger
Patches:
asterisk-20435-imap-del-greeting.diff uploaded by Michael L. Young (License #5026)
(with suggested modification made by me)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373740 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoFix T.38 support when used with chan_local in between.
Joshua Colp [Tue, 25 Sep 2012 20:14:13 +0000 (20:14 +0000)]
Fix T.38 support when used with chan_local in between.

Users of the T.38 API can indicate AST_T38_REQUEST_PARMS on a channel to request that the
channel indicate a T.38 negotiation with the parameters present on the channel. The return
value of this indication is expected to be AST_T38_REQUEST_PARMS upon success but with
chan_local involved this could never occur.

This fix changes chan_local to always return AST_T38_REQUEST_PARMS for this situation. If
the underlying channel technology on the other side does not support T.38 this would have
been determined ahead of time using ast_channel_get_t38_state and an indication would
not occur.

(closes issue ASTERISK-20229)
Reported by: wdoekes
Patches:
     ASTERISK-20229.patch uploaded by wdoekes (license 5674)

Review: https://reviewboard.asterisk.org/r/2070/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373708 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoAllow for redirecting reasons to be set to arbitrary strings.
Mark Michelson [Tue, 25 Sep 2012 19:29:14 +0000 (19:29 +0000)]
Allow for redirecting reasons to be set to arbitrary strings.

This allows for the REDIRECTING dialplan function to be
used to set the reason to any string.

The SIP channel driver has been modified to set the redirecting
reason string to the value received in a Diversion header. In
addition, SIP 480 response reason text will set the redirecting
reason as well.

(closes issue AST-942)
reported by Malcolm Davenport

(closes issue AST-943)
reported by Malcolm Davenport

Review: https://reviewboard.asterisk.org/r/2101

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373701 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoProperly handle UAC/UAS roles for SIP session timers
Terry Wilson [Tue, 25 Sep 2012 19:08:02 +0000 (19:08 +0000)]
Properly handle UAC/UAS roles for SIP session timers

The SIP session timer mechanism contains a mandatory 'refresher' parameter
(included in the Session-Expires header) which is used in the session timer
offer/answer signaling within a SIP Invite dialog. It looks like asterisk is
interpreting the uac resp. uas role only as the initial role of client and
server (caller is uac, callee is uas). The standard rfc 4028 however assigns
the client role to the ((RE)-Invite) requester, the server role to the
((RE)-Invite) responder.

This patch has Asterisk track the actual refresher as "us" or "them" as opposed
to relying on just the configured "uas" or "uac" properties.

(closes issue AST-922)
Reported by: Thomas Airmont

Review: https://reviewboard.asterisk.org/r/2118/
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6 years ago"show" completion option for "queue" shouldn't appear twice
Kinsey Moore [Tue, 25 Sep 2012 18:33:59 +0000 (18:33 +0000)]
"show" completion option for "queue" shouldn't appear twice

When tab-completing CLI commands starting with "queue", "show" appeared
twice in the list due to the way that Asterisk's tab completion
functions and the order in which the commands were registered. The
registration order has been altered to resolve this issue.

(closes issue AST-940)
Reported-by: Steve Pitts
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6 years agoFix valgrind found memcpy issues in codec_ilbc.
Richard Mudgett [Tue, 25 Sep 2012 17:22:25 +0000 (17:22 +0000)]
Fix valgrind found memcpy issues in codec_ilbc.

Valgrind found codec_ilbc using memcpy instead of memmove for overlapping
memory blocks.

(issue ASTERISK-19890)
(closes issue ASTERISK-20231)
Reported by: Walter Doekes
Patches:
      ASTERISK-20231.patch (license #5674) patch uploaded by Walter Doekes
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6 years agoMake rebuild GSM, ilbc, or lpc10 codecs if the respective sources change.
Richard Mudgett [Tue, 25 Sep 2012 17:02:21 +0000 (17:02 +0000)]
Make rebuild GSM, ilbc, or lpc10 codecs if the respective sources change.
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6 years agochan_sip: Set Quality of Service for video rtp instance
Jonathan Rose [Tue, 25 Sep 2012 16:45:02 +0000 (16:45 +0000)]
chan_sip: Set Quality of Service for video rtp instance

(closes issue ASTERISK-20201)
Reported by: ddkprog
Patches:
    chan_sip.c.diff uploaded by ddkprog (license 6008)
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6 years agores_agi: async_agi responsiveness improvement on datastore problems
Jonathan Rose [Tue, 25 Sep 2012 14:53:42 +0000 (14:53 +0000)]
res_agi: async_agi responsiveness improvement on datastore problems

This patch changes get_agi_cmd so that the return can be checked
to differentiate between an empty list success and something that
triggered an error. This in turn allows launch_asyncagi to detect
these errors and break free from the command processing loop so
that the async agi can be ended more cleanly

(closes issue ASTERISK-20109)
Reported by: Jeremiah Gowdy
Patches: jgowdy-7-9-2012.diff uploaded by Jeremiah Gowdy (license 6358)
           (Modified by me to fix some logical issues and apply to trunk)
Review: https://reviewboard.asterisk.org/r/2117/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373608 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years ago"He who go through turnstile sideways is going to Bangkok"
Mark Michelson [Tue, 25 Sep 2012 14:13:08 +0000 (14:13 +0000)]
"He who go through turnstile sideways is going to Bangkok"
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6 years agoFix documentation for default username in res_odbc
Kinsey Moore [Tue, 25 Sep 2012 13:29:37 +0000 (13:29 +0000)]
Fix documentation for default username in res_odbc

This was previously stated to be "root", but is actually the name of
the context if unspecified.

(closes issue ASTERISK-20258)
Reported by: Stefan x
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6 years agoFix an issue where a caller to ast_write on a MulticastRTP channel would determine...
Joshua Colp [Tue, 25 Sep 2012 12:12:20 +0000 (12:12 +0000)]
Fix an issue where a caller to ast_write on a MulticastRTP channel would determine it failed when in reality it did not.

When sending RTP packets via multicast the amount of data sent is stored in a variable and returned
from the write function. This is incorrect as any non-zero value returned is considered a failure while
a return value of 0 is success. For callers (such as ast_streamfile) that checked the return value
they would have considered it a failure when in reality nothing went wrong and it was actually a success.

The write function for the multicast RTP engine now returns -1 on failure and 0 on success, as it should.

(closes issue ASTERISK-17254)
Reported by: wybecom
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6 years agoBe consistent, send From: "Anonymous" <sip:anonymous@anonymous.invalid>
Richard Mudgett [Mon, 24 Sep 2012 22:14:28 +0000 (22:14 +0000)]
Be consistent, send From: "Anonymous" <sip:anonymous@anonymous.invalid>

When setting CALLERID(pres)=unavailable in the dialplan, the From header
in the SIP message contains "Anonymous" <sip:Anonymous@anonymous.invalid>.
For consistency, Asterisk should use a lowercase a in the userpart of the
URI.

* Make the From header use a lowercase A in the userpart of the anonymous
URI.

(closes issue ASTERISK-19838)
Reported by: Antti Yrjola
Patches:
      chan_sip_patch_ASTERISK-19838.patch (license #6383) patch uploaded by Antti Yrjola
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6 years agofunc_audiohookinherit: Document some missed sources.
Jonathan Rose [Mon, 24 Sep 2012 21:19:49 +0000 (21:19 +0000)]
func_audiohookinherit: Document some missed sources.

This patch also mentions that AUDIOHOOK_INHERIT can be used to
transfer MixMonitor audiohooks. There is also wiki that addresses
audiohooks and the use of AUDIOHOOK_INHERIT at the following link:
https://wiki.asterisk.org/wiki/display/AST/Audiohooks

(closes issue ASTERISK-18220)
Reported by: Ishfaq Malik
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6 years agoFix potential reentrancy problems in chan_sip.
Richard Mudgett [Mon, 24 Sep 2012 21:15:26 +0000 (21:15 +0000)]
Fix potential reentrancy problems in chan_sip.

Asterisk v1.8 and later was not as vulnerable to this issue.

* Made find_call() lock each private as it processes the found dialogs.
(Primary cause of ABE-2876)

* Made the other functions that traverse the dialogs container lock each
private as it examines them.

* Fix race condition in sip_call() if the thread that sent the INVITE is
held up long enough for a response to be processed.  The p->initid for the
INVITE retransmission could be added after it was canceled by the response
processing.

* Made __sip_destroy() clean up resource pointers after freeing.  This is
primarily defensive in case someone has a stale private pointer.

* Removed redundant memset() in reqprep().  The call to init_req() already
does the memset() and is the first reference to req in reqprep().

* Removed useless set of req.method in transmit_invite().  The calls to
initreqprep() and reqprep() have to do this because they memset() the req.

JIRA ABE-2876

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6 years agoFix a deadlock caused by a race condition between removing a hint and reloading the...
Joshua Colp [Mon, 24 Sep 2012 19:23:32 +0000 (19:23 +0000)]
Fix a deadlock caused by a race condition between removing a hint and reloading the dialplan and subscribing to the removed hint.

If conditions were right it was possible for both the PBX core and chan_sip to deadlock by both having a lock that the other
wants. In the case of the PBX core it had the contexts lock and wanted a SIP dialog lock, while in the case of chan_sip it
had the SIP dialog lock and wanted the contexts lock.

This fix unlocks the SIP dialog before getting the extension state so that the other thread will not block on trying to lock
it. Once the extension state is retrieved the SIP dialog is locked again and life carries on.

As the SIP dialog is reference counted it is not possible for it to go away after unlocking.

(closes issue ASTERISK-20437)
Reported by: jhutchins
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6 years agoFix an issue with H.264 format attribute comparison and fix an issue with improper...
Joshua Colp [Mon, 24 Sep 2012 14:27:17 +0000 (14:27 +0000)]
Fix an issue with H.264 format attribute comparison and fix an issue with improper SDP being produced.

The H.264 format attribute module compares two format attribute structures to determine if they are
compatible or not. In some instances it was possible for this check to determine that both structures
were incompatible when they actually should be considered compatible. This check has now been made even
more permissive by assuming that if no attribute information is available the two structures are compatible.
If both structures contain attribute information a base level comparison of the H.264 IDC value is done to
see if they are compatible or not.

The above issue uncovered a secondary issue in chan_sip where the SDP being produced would be incorrect if
the formats were considered incompatible. This has now been fixed by checking that all information required
to produce the SDP is available instead of assuming it is.

(closes issue ASTERISK-20464)
Reported by: Leif Madsen
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6 years agores_rtp_asterisk: Make TURN and STUN server configurations consistent.
Brent Eagles [Mon, 24 Sep 2012 12:42:19 +0000 (12:42 +0000)]
res_rtp_asterisk: Make TURN and STUN server configurations consistent.

This patch removes the turnport configuration property and changes the
turnaddr property to be a combined host[:port] configuration string. The
patch also modifies the documentation in the example configuration to
reflect the property changes and adds some additional text indicating how
the STUN port is configured.

(closes issue ASTERISK-20344)
Reported by: beagles
Tested by: beagles
Review: https://reviewboard.asterisk.org/r/2111/
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6 years agoDoxygen Updates Janitor Work
Andrew Latham [Sat, 22 Sep 2012 20:43:30 +0000 (20:43 +0000)]
Doxygen Updates Janitor Work

* Whitespace, doc-blocks, spelling, case, missing and incorrect tags.
* Add cleanup to Makefile for the Doxygen configuration update
* Start updating Doxygen configuration for cleaner output
* Enable inclusion of configuration files into documentation
* remove mantisworkflow...
* update documentation README
* Add markup to Tilghman's email and talk with him about updating his email, he knows...
* no code changes on this commit other than the mentioned Makefile change

(issue ASTERISK-20259)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373384 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoiax2-provision: Fix improper return on failed cache retrieval
Jonathan Rose [Fri, 21 Sep 2012 19:35:37 +0000 (19:35 +0000)]
iax2-provision: Fix improper return on failed cache retrieval

(closes issue ASTERISK-20337)
reported by: John Covert
Patches:
    iax2-provision.c.patch uploaded by John Covert (license 5512)
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6 years agoUpdate Doxygen Config Comments
Andrew Latham [Fri, 21 Sep 2012 18:22:05 +0000 (18:22 +0000)]
Update Doxygen Config Comments

This annoying update is almost totally whitespace and updated config comments. I did add Python to the documented file types.

(issue ASTERISK-20259)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373341 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoDoxygen Updates - janitor work
Andrew Latham [Fri, 21 Sep 2012 17:14:59 +0000 (17:14 +0000)]
Doxygen Updates - janitor work

Doxygen updates including mistakes, misspellings, missing parameters, updates for Doxygen style.  Some missing txt file links are removed but their content or essense will be included in some later updates.  A majority of the txt files were removed in the 1.6 era but never noted. The HR and EXTREF are simple changes that make the documentation more compatable with more versions of Doxygen.

Further updates coming.

(issue ASTERISK-20259)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373330 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoStart work on documentation janitor project with a little commit. This adds a link...
Andrew Latham [Fri, 21 Sep 2012 16:06:30 +0000 (16:06 +0000)]
Start work on documentation janitor project with a little commit. This adds a link to the Asterisk wiki at https://wiki.asterisk.org to the README file.

(issue ASTERISK-20259)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373320 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoapp_queue: Make queue reload members and variants of that work
Jonathan Rose [Fri, 21 Sep 2012 15:41:09 +0000 (15:41 +0000)]
app_queue: Make queue reload members and variants of that work

Prior to this patch, 'queue reload members' cli command did not
work at all. This also affects the manager function 'QueueReload'
when supplied with the 'members: yes' field.

(closes issue AST-956)
Reported by: John Bigelow
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6 years agodsp.c: remove more whitespace mentioned in review2107
Alec L Davis [Fri, 21 Sep 2012 09:11:39 +0000 (09:11 +0000)]
dsp.c: remove more whitespace mentioned in review2107

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373284 65c4cc65-6c06-0410-ace0-fbb531ad65f3