Commit Graph

3332 Commits

Author SHA1 Message Date
Jonathan Rose 19a4928fee INFO/Record request configurable to use dynamic features
Adds two new options to SIP peers allowing them to specify features (dynamic or builtin)
to use when sending INFO/record requests. Recordonfeature activates whatever feature
is specified when recieving a record: on request while recordofffeature activates
whatever feature is specified when receiving a record: off request. Both of these
features can be disabled by setting the feature to an empty string.

(closes issue ASTERISK-16507)
Reported by: Jon Bright
Review: https://reviewboard.asterisk.org/r/1634/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349098 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-23 20:42:21 +00:00
Richard Mudgett 32e35e5fcd Fix extension state callback references in chan_sip.
Chan_sip gives a dialog reference to the extension state callback and
assumes that when ast_extension_state_del() returns, the callback cannot
happen anymore.  Chan_sip then reduces the dialog reference count
associated with the callback.  Recent changes (ASTERISK-17760) have
resulted in the potential for the callback to happen after
ast_extension_state_del() has returned.  For chan_sip, this could be very
bad because the dialog pointer could have already been destroyed.

* Added ast_extension_state_add_destroy() so chan_sip can account for the
sip_pvt reference given to the extension state callback when the extension
state callback is deleted.

* Fix pbx.c awkward statecbs handling in ast_extension_state_add_destroy()
and handle_statechange() now that the struct ast_state_cb has a destructor
to call.

* Ensure that ast_extension_state_add_destroy() will never return -1 or 0
for a successful registration.

* Fixed pbx.c statecbs_cmp() to compare the correct information.  The
passed in value to compare is a change_cb function pointer not an object
pointer.

* Make pbx.c ast_merge_contexts_and_delete() not perform callbacks with
AST_EXTENSION_REMOVED with locks held.  Chan_sip is notorious for
deadlocking when those locks are held during the callback.

* Removed unused lock declaration for the pbx.c store_hints list.

(closes issue ASTERISK-18844)
Reported by: rmudgett

Review: https://reviewboard.asterisk.org/r/1635/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@348953 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-23 02:35:13 +00:00
Matthew Jordan cf0c9830bf Add Asterisk TestSuite event hooks to support ConfBridge testing
This patch adds initial testsuite event hooks so that ConfBridge tests
can be executed in the Asterisk TestSuite.

(issue ASTERISK-19059)
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Merged revisions 348846 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@348848 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-22 20:44:53 +00:00
Leif Madsen eb37d38b7d Update documentation for MESSAGE_SEND_STATUS variable.
(Closes issue ASTERISK-19056)
Reported by: Yuri
Patches:
     348360.diff uploaded by Yuri (license #5242)
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Merged revisions 348605 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@348606 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-19 19:55:18 +00:00
Richard Mudgett be74e6f16e Clean-up on isle five for __ast_request_and_dial() and ast_call_forward().
* Add locking when a channel inherits variables and datastores in
__ast_request_and_dial() and ast_call_forward().  Note: The involved
channels are not active so there was minimal potential for problems.

* Remove calls to ast_set_callerid() in __ast_request_and_dial() and
ast_call_forward() because the set information is for the wrong direction.

* Don't use C++ keywords for variable names in ast_call_forward().

* Run the redirecting interception macro if defined when forwarding a call
in ast_call_forward().  Note: Currently will never execute because the
only callers that supply a calling channel supply a hungup or zombie
channel.

* Make feature_request_and_dial() put the transferee into autoservice when
it calls ast_call_forward() in case a redirection interception macro is
run.  Note: Currently will never happen because the caller channel (Party
B) is always hungup at this time.

* Make feature_request_and_dial() ignore the AST_CONTROL_PROCEEDING frame
to silence a log message.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@348466 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-16 23:58:44 +00:00
Richard Mudgett e71bad4958 Fix cut and past error in ast_call_forward().
(issue ASTERISK-18836)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@348408 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-16 21:30:35 +00:00
Richard Mudgett b05d4603c4 Fix crash during CDR update.
The ast_cdr_setcid() and ast_cdr_update() were shown in ASTERISK-18836 to
be called by different threads for the same channel.  The channel driver
thread and the PBX thread running dialplan.

* Add lock protection around CDR API calls that access an ast_channel
pointer.

(closes issue ASTERISK-18836)
Reported by: gpluser

Review: https://reviewboard.asterisk.org/r/1628/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@348364 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-16 21:10:19 +00:00
Jonathan Rose 480d46f92c Add and document PARKEDCALL variable set during timeout
PARKEDCALL variable tracks which parking lot the call was last parked in.  This can be
used afterwards for flow control when returntoorigin is set to off. I went ahead and
documented both this and the existing variable set during timeout (PARKINGSLOT) in
the sample features.conf since there was no prior mention of variables being set during
timeout.

(closes issue ASTERISK-16239)
Reported By: Clod Patry
Patches:
	M17503.diff uploaded by Clod Patry (license 5138)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@348161 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-14 21:08:20 +00:00
Matthew Jordan 9057aa20b6 Backed out core changes from r346391
During testing, it was discovered that there were a number of side effects
introduced by r346391 and subsequent check-ins related to it (r346429,
r346617, and r346655).  This included the /main/stdtime/ test 'hanging',
as well as the remote console option failing to receive the appropriate output
after a period of time.

I only backed out the changes to main/ and utils/, as this was adequate
to reverse the behavior experienced.

(issue ASTERISK-18974)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347997 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-12 19:35:08 +00:00
Richard Mudgett 01d3fd2167 Fix some parsing issues in add_exten_to_pattern_tree().
* Simplify compare_char() and avoid potential sign extension issue.

* Fix infinite loop in add_exten_to_pattern_tree() handling of character
set escape handling.

* Added buffer overflow checks in add_exten_to_pattern_tree() character
set collection.

* Made ignore empty character sets.

* Added escape character handling to end-of-range character in character
sets.  This has a slight change in behavior if the end-of-range character
is an escape character.  You must now escape it.

* Fix potential sign extension issue when expanding character set ranges.

* Made remove duplicated characters from character sets.  The duplicate
characters lower extension matching priority and prevent duplicate
extension detection.

* Fix escape character handling when the escape character is trying to
escape the end-of-string.  We could have continued processing characters
after the end of the exten string.  We could have added the previous
character to the pattern matching tree incorrectly.

(closes issue ASTERISK-18909)
Reported by: Luke-Jr
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347813 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-09 01:33:29 +00:00
Richard Mudgett 3f13a41886 Mark channel running the h exten with the soft-hangup flag.
When a bridge is broken, ast_bridge_call() might execute the h exten on
the calling channel.  However, that channel may not have been the channel
that broke the bridge by hanging up.  The channel executing the h exten
must be in a hung up state so things like AGI run in the correct mode.

* Make sure ast_bridge_call() marks the channel it is executing the h
exten on as hung up.  (The AST_SOFTHANGUP_APPUNLOAD flag is used so as to
match the pbx.c main dialplan execution loop when it executes the h
exten.)

(closes issue ASTERISK-18811)
Reported by: David Hajek
Patches:
      jira_asterisk_18811_v1.8.patch (license #5621) patch uploaded by rmudgett
Tested by: David Hajek, rmudgett
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347601 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-08 17:55:07 +00:00
Richard Mudgett 395814c33e Update AMI Getvar and Setvar documentation about supplying a channel name.
(closes issue ASTERISK-18958)
Reported by: Red
Patches:
      jira_asterisk_18958_v1.8.patch (license #5621) patch uploaded by rmudgett
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347440 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-07 21:42:29 +00:00
Terry Wilson 980ab2d018 Add ASTSBINDIR to the list of configurable paths
This patch also makes astdb2sqlite3 and astcanary use the configured
directory instead of relying on $PATH.

(closes issue ASTERISK-18959)
Review: https://reviewboard.asterisk.org/r/1613/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347345 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-07 20:15:29 +00:00
Jonathan Rose 9b33408ba1 Documents CHANNEL(musicclass) taking priority over m([x]) in waitExten
If waitExten specifies a music class to use with its music on hold option, it will use
CHANNEL(musicclass) instead if that channel variable has been set on the initiating
channel.  This documents that behavior in the waitExten app so that this can be known
without checking the documentation of the code in function local_ast_moh_start.

(closes issue ASTERISK-18804)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347241 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-06 22:01:00 +00:00
Jonathan Rose c5fe1cfdc0 Resolve duplicate label used in multiple priorities for the same extension.
Prior to this patch, if labels with the same name were used for different priorities in
the same extension, the new label would be accepted, but it would be unusable since
attempts to reach that label would just go to the first one. Now pbx.c detects this,
generates a warning in logs, and culls the label before adding it to the dialplan.

(closes issue ASTERISK-18807)
Reported by: Kenneth Shumard
Patches:
	pbx.c.patch uploaded by Kenneth Shumard (License 5077)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346956 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-05 15:04:12 +00:00
Richard Mudgett 83cd844b82 Re-resolve the STUN address if a STUN poll fails for res_stun_monitor.
The STUN socket must remain open between polls or the external address
seen by the STUN server is likely to change.  However, if the STUN request
poll fails then the STUN server address needs to be re-resolved and the
STUN socket needs to be closed and reopened.

* Re-resolve the STUN server address and create a new socket if the STUN
request poll fails.

* Fix ast_stun_request() return value consistency.

* Fix ast_stun_request() to check the received packet for expected message
type and transaction ID.

* Fix ast_stun_request() to read packets until timeout or an associated
response packet is found.  The stun_purge_socket() hack is no longer
required.

* Reduce ast_stun_request() error messages to debug output.

* No longer pass in the destination address to ast_stun_request() if the
socket is already bound or connected to the destination.

(closes issue ASTERISK-18327)
Reported by: Wolfram Joost
Tested by: rmudgett

Review: https://reviewboard.asterisk.org/r/1595/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346709 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-01 21:19:41 +00:00
Tilghman Lesher 56b21b4683 Remove the few places where we try to ast_verbose() without a newline.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346655 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-11-30 23:38:34 +00:00
Tilghman Lesher 3106f64eac Fix edge case for overflow buffer.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346617 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-11-30 22:40:23 +00:00
Jonathan Rose 9ef171ffe0 r346525 | jrose | 2011-11-30 15:10:38 -0600 (Wed, 30 Nov 2011) | 18 lines
Cleaning up chan_sip/tcptls file descriptor closing.

This patch attempts to eliminate various possible instances of undefined behavior caused
by invoking close/fclose in situations where fclose may have already been issued on a
tcptls_session_instance and/or closing file descriptors that don't have a valid index
for fd (-1). Thanks for more than a little help from wdoekes.

(closes issue ASTERISK-18700)
Reported by: Erik Wallin

(issue ASTERISK-18345)
Reported by: Stephane Cazelas

(issue ASTERISK-18342)
Reported by: Stephane Chazelas

Review: https://reviewboard.asterisk.org/r/1576/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346566 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-11-30 22:03:02 +00:00
Jonathan Rose fb4c483eb7 Reverting 346525 due to accidental patch against trunk instead of 1.8
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346563 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-11-30 21:32:23 +00:00
Jonathan Rose 6fa827b5d0 Cleaning up chan_sip/tcptls file descriptor closing.
This patch attempts to eliminate various possible instances of undefined behavior caused
by invoking close/fclose in situations where fclose may have already been issued on a
tcptls_session_instance and/or closing file descriptors that don't have a valid index
for fd (-1). Thanks for more than a little help from wdoekes.

(closes issue ASTERISK-18700)
Reported by: Erik Wallin

(issue ASTERISK-18345)
Reported by: Stephane Cazelas

(issue ASTERISK-18342)
Reported by: Stephane Chazelas

Review: https://reviewboard.asterisk.org/r/1576/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346525 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-11-30 21:10:38 +00:00
Tilghman Lesher 77b670c4ab Allow each logging destination and console to have its own notion of the verbosity level.
Review: https://reviewboard.asterisk.org/r/1599


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346391 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-11-29 18:43:16 +00:00
David Vossel d7dec4f14f Merged revisions 346349 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r346349 | dvossel | 2011-11-28 18:00:11 -0600 (Mon, 28 Nov 2011) | 10 lines
  
  Fixes memory leak in message API.
  
  The ast_msg_get_var function did not properly decrement
  the ref count of the var it retrieves.  The way this is
  implemented is a bit tricky, as we must decrement the var and then
  return the var's value.  As long as the documentation for the
  function is followed, this will not result in a dangling pointer as
  the ast_msg structure owns its own reference to the var while it
  exists in the var container.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346350 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-11-29 00:03:36 +00:00
Richard Mudgett 7d9ba4875b Fix calls to ast_get_ip() not initializing the address family.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346241 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-11-23 23:03:32 +00:00
Richard Mudgett 2b3e28f88c Fix dnsmgr entries to ask for the same address family each time.
The dnsmgr refresh would always get the first address found regardless of
the original address family requested.  So if you asked for only IPv4
addresses originally, you might get an IPv6 address on refresh.

* Saved the original address family requested by ast_dnsmgr_lookup() to be
used when the address is refreshed.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345978 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-11-22 23:06:11 +00:00
Paul Belanger 298d015828 Add #tryinclude statement
This provides the same functionality as #include however an asterisk module will
still load if the filename does not exist.

Review: https://reviewboard.asterisk.org/r/1476/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345735 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-11-21 16:40:17 +00:00
Tilghman Lesher 6e7359f594 Update the documentation to better clarify how the existing commands work.
Review: https://reviewboard.asterisk.org/r/1593/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345684 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-11-19 15:11:45 +00:00
Tilghman Lesher 6b8b13ec2d Fix a change in behavior in 'database show' from 1.8.
In 1.8 and previous versions, one could use any fullword portion of the key
name, including the full key, to obtain the record.  Until this patch, this
did not work for the full key.

Closes issue ASTERISK-18886

Patch: by tilghman
Review: by twilson (http://pastebin.com/7rtu6bpk) on #asterisk-dev
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345643 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-11-18 22:20:47 +00:00
Richard Mudgett 1cef6cf8cd Fix Progress spelling error in main/pbx.c.
(closes issue ASTERISK-18857)
Reported by: David M
Patches:
      mainpbx-trivial.patch (License #6326) patch uploaded by David M
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345221 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-11-14 20:48:19 +00:00
Terry Wilson 59d6db63bd Don't read past end of input when calling write()
int blah = 1;
...
write(chan->alertpipe[1], &blah, new_frames * sizeof(blah)) !=
(new_frames * sizeof(blah)))

is only valid when new_frames == 1. Otherwise we start reading into adjacent
variables declared on the stack. The read end discards what is read, so the
values don't matter but it's not a good idea to read past where we want even
though new_frames is almost always 1 and should never be large. This patch is
basically taken out of kpfleming's eventfd branch, as he mentioned that he
remembered fixing it there when I talked to him about this issue.

Review: https://reviewboard.asterisk.org/r/1583/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345165 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-11-14 19:12:49 +00:00
Walter Doekes 735e48f92f Use __alignof__ instead of sizeof for stringfield length storage.
Kevin P Fleming suggested that r343157 should use __alignof__ instead
of sizeof. For most systems this won't be an issue, but better fix it
now while it's still fresh.

Review: https://reviewboard.asterisk.org/r/1573
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344846 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-11-11 22:00:14 +00:00
Matthew Jordan 60f51c002a Video format was treated as audio when removed from the file playback scheduler
This patch fixes the format type check in ast_closestream and 
filestream_destructor.  Previously a comparison operator was used, but since
audio formats are no longer contiguous (and AST_FORMAT_AUDIO_MASK includes
formats that have a value greater than the video formats), a bitwise AND
operation is used instead.  Duplicated code was also moved to filestream_close.

(closes issue ASTERISK-18682)
Reported by: Aldo Bedrij
Tested by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/1580/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344844 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-11-11 21:57:46 +00:00
Walter Doekes bac9ff62ef Fix bad quoting of multiline mxml opaque_data that caused invalid xml.
The opaque_data was added and enclosed in single quotes, assuming it
would be only a single line. The rest of the lines were appended after
the closing quote.

(closes issue ASTERISK-18852)
Reported by: peep_ on IRC

Review: https://reviewboard.asterisk.org/r/1577
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344838 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-11-11 21:33:54 +00:00
Richard Mudgett 39beaff425 Make CLI "core show channel" not hold the channel lock during console output.
Holding the channel lock while the CLI "core show channel" command is
executing can slow down the system.  It could block the system if the
console output is halted or paused.

* Made capture the CLI "core show channel" output into a buffer to be
output after the channel is unlocked.

* Removed use of C++ keyword as a variable name.  out renamed to obuf.

* Checked allocation of obuf for failure so will not crash.

(closes issue ASTERISK-18571)
Reported by: Pavel Troller
Tested by: rmudgett
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344663 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-11-11 18:02:52 +00:00
Jonathan Rose 8d994bed55 Fix a segmentation fault when using an extension with CID matching and no CID.
Attempting to call an extension which used Caller ID matching with a channel that
has an empty caller id string would result in a segmentation fault.

(closes issue ASTERISK-18392
Reported By: Ales Zelenik
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2011-11-11 15:47:39 +00:00
David Vossel 0a2a79c94b Merged revisions 344493 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r344493 | dvossel | 2011-11-10 15:54:42 -0600 (Thu, 10 Nov 2011) | 12 lines
  
  Fixes issue with ConfBridge participants hanging up during DTMF feature menu usage getting stuck in conference forever.
  
  When a conference user enters the DTMF menu they are suspended from the
  bridge while the channel is handed off to the DTMF feature code.  If a
  user entered this state and hungup, there existed a race condition where
  the channel could not exit the conference because it was waiting on a
  signal that would never arrive.  This patch fixes that, because it would
  stupid for me to talk about the problem and commit a patch for something else.
  
  (closes issue ASTERISK-18829)
  Reported by: zvision
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344494 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-11-10 21:56:16 +00:00
Richard Mudgett cee432c9d8 Fixed reference to incorrect variable if unknown host configured crash.
* Fixed a LOG_ERROR message referencing the config variable list v that
had previously been processed and became NULL.

* Added error return value set that was missing in an ast_append_ha()
error return path.

(closes issue ASTERISK-18743)
Reported by: Michele
Patches:
      issueA18743-fix_dynamic_exclude_static_bad_host_log.patch (license #5674) patch uploaded by Walter Doekes
Tested by: Michele
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@343853 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-11-08 18:02:51 +00:00
Walter Doekes 00a522c000 Correct the default udptl port range.
The udptl port range was defined as 4000-4999 in the udptl.conf.sample,
as 4500-4599 if you didn't have a config and 4500-4999 if your config
was broken. Default is now 4000-4999.

(closes issue ASTERISK-16250)
Reviewed by: Tilghman Lesher

Review: https://reviewboard.asterisk.org/r/1565
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2011-11-07 19:58:44 +00:00
Matthew Nicholson f3c03ac1ec list all of the codecs associated with a particular format id for CLI command "core show codec"
AST-699
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2011-11-07 18:42:04 +00:00
Olle Johansson 816dc295c2 Formatting and doxygen improvements
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@343492 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-11-06 09:51:09 +00:00
Walter Doekes 7fd2a68b0e Ensure that string field lengths are properly aligned
Integers should always be aligned. For some platforms (ARM, SPARC) this
is more important than for others. This changeset ensures that the
string field string lengths are aligned on *all* platforms, not just on
the SPARC for which there was a workaround. It also fixes that the
length integer can be resized to 32 bits without problems if needed.

(closes issue ASTERISK-17310)
Reported by: radael, S Adrian
Reviewed by: Tzafrir Cohen, Terry Wilson
Tested by: S Adrian

Review: https://reviewboard.asterisk.org/r/1549
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2011-11-02 22:02:07 +00:00
Walter Doekes b41b49ea0e Several fixes to the chan_sip dynamic realtime peer/user lookup
There were several problems with the dynamic realtime peer/user lookup
code. The lookup logic had become rather hard to read due to lots of
incremental changes to the realtime_peer function. And, during the
addition of the sipregs functionality, several possibilities for memory
leaks had been introduced. The insecure=port matching has always been
broken for anyone using the sipregs family. And, related, the broken
implementation forced those using sipregs to *still* have an ipaddr
column on their sippeers table.

Thanks Terry Wilson for comprehensive testing and finding and fixing
unexpected behaviour from the multientry realtime call which caused
the realtime_peer to have a completely unused code path.

This changeset fixes the leaks, the lookup inconsistenties and that
you won't need an ipaddr column on your sippeers table anymore (when
you're using sipregs). Beware that when you're using sipregs, peers
with insecure=port will now start matching!

(closes issue ASTERISK-17792)
(closes issue ASTERISK-18356)
Reported by: marcelloceschia, Walter Doekes
Reviewed by: Terry Wilson

Review: https://reviewboard.asterisk.org/r/1395
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2011-11-01 21:02:56 +00:00
Walter Doekes 25ee5f83b5 Cleanup references to sipusers and sipfriends dynamic realtime families
Somewhere between 1.4 and 1.8 the sipusers family has become completely
unused. Before that, the sipfriends family had been obsoleted in favor
of separate sipusers and sippeers families. Apparently, they have been
merged back again into a single family which is now called "sippeers".

Reviewed by: irroot, oej, pabelanger

Review: https://reviewboard.asterisk.org/r/1523
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2011-11-01 19:53:26 +00:00
Richard Mudgett ed1e02a4d3 Misc format capability fixes.
* Fixed typo in format_cap.c:joint_copy_helper() using the wrong variable.

* Fix potential race between checking if an interface exists and adding it
to the container in format.c:ast_format_attr_reg_interface().

* Fixed double rwlock destroy in format.c:ast_format_attr_init() error
exit path.

* Simplified format.c:find_interface() and format.c:has_interface().
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2011-10-31 17:51:22 +00:00
Matthew Jordan 9333071c1f Fixed invalid memory access when adding extension to pattern match tree
When an extension is removed from a context, its entry in the pattern match
tree is not deleted.  Instead, the extension is marked as deleted.  When an
extension is removed and re-added, if that extension is also a prefix of
another extension, several log messages would report an error and did not
check whether or not the extension was deleted before accessing the memory.
Additionally, if the extension was already in the tree but previously
deleted, and the pattern was at the end of a match, the findonly flag was
not honored and the extension would be erroneously undeleted.  

Additionaly, it was discovered that an IAX2 peer could be unregistered
via the CLI, while at the same time it could be scheduled for unregistration
by Asterisk.  The unregistration method now checks to see if the peer
was already unregistered before continuing with an unregistration.

(closes issue ASTERISK-18135)
Reported by: Jaco Kroon, Henry Fernandes, Kristijan Vrban
Tested by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/1526
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2011-10-31 16:10:32 +00:00
Matthew Nicholson 849992fde9 tweak the v21 detector to detect an additional pattern of hits and misses
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2011-10-27 20:11:11 +00:00
Richard Mudgett ebf860e157 Check fopen return value for ao2 reference debug output.
Reported by: wdoekes
Patched by: wdoekes

Review: https://reviewboard.asterisk.org/r/1539/
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2011-10-25 22:06:43 +00:00
Terry Wilson 700f0a2752 Return NULL when no results returned for realtime_multientry
It was not documented what the return value should be when no entries
were returned with the multientry realtime callback. This change forces
consistent behavior even if the backends return an empty ast_config.

Review: https://reviewboard.asterisk.org/r/1521/
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2011-10-25 01:29:32 +00:00
Jonathan Rose f6cd5af36b Fixes a segfault caused by referencing null frames introduced in r338623
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@342148 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-10-24 22:09:11 +00:00
Richard Mudgett b961d57c4c Fix AGI exec Park to honor the Park application parameters.
The fix for ASTERISK-12715 and ASTERISK-12685 added a check for the Park
application because the channel needed to be masqueraded to prevent a
crash.  Since the Park application now always masquerades the channel into
the parking lot, the special check is no longer needed.  The fix also
resulted in AGI exec Park attempting to double park the call and not honor
the Park application parameters.

* Removed no longer necessary call to ast_masq_park_call() by AGI exec for
the Park application.  (Reverts -r146923)

* Fix Park application to only return 0 or -1.  The AGI exec Park was
causing broken pipe error messages because the Park application returned 1
on successful park.

(closes issue ASTERISK-18737)
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2011-10-20 22:03:35 +00:00