Commit Graph

22408 Commits

Author SHA1 Message Date
Matthew Jordan 70c5ac6635 Fix error that caused seek format operations to set max file size to '1' or '0'
A very inappropriate placement of a ')' (introduced in r362151) caused the
maximum size of a file to be set as the result of a comparison operation, as
opposed to the result of the ftello operation.  This resulted in seeking being
restricted to the beginning of the file, or 1 byte into the file.  Thanks to
the Asterisk Test Suite for properly freaking out about this on at least one
test.

(issue ASTERISK-19655)
Reported by: Matt Jordan
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Merged revisions 362304 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 362305 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362306 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-17 18:29:51 +00:00
Michael L. Young 8337ecd38d Turn off warning message when bind address is set to any.
When a bind address is set to an ANY address (udpbindport=::), a warning message
is displayed stating that "Address remapping activated in sip.conf but we're
using IPv6, which doesn't need it.  Please remove 'localnet' and/or 'externaddr'
settings."  But if one is running dual stack, we shouldn't be told to turn those
settings off.

This patch checks if the bind address is an ANY address or not.  The warning
message will now only be displayed if the bind address is NOT an ANY address and
IPv6 is being used.

Also, updated the copyright year.

(closes issue ASTERISK-19456) 
Reported by: Michael L. Young 
Tested by: Michael L. Young 
Patches: 
  chan_sip_ipv6_message.diff uploaded by Michael L. Young (license 5026)
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Merged revisions 362253 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 362264 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362266 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-17 15:00:02 +00:00
Matthew Jordan 2fed9cfa8f Fix negative return handling in channel drivers
In chan_agent, while handling a channel indicate, the agent channel driver
must obtain a lock on both the agent channel, as well as the channel the
agent channel is using.  To do so, it attempts to lock the other channel
first, then unlock the agent channel which is locked prior to entry into
the indicate handler.  If this unlock fails with a negative return value,
which can occur if the object passed to agent_indicate is an invalid ao2
object or is NULL, the return value is passed directly to strerror, which
can only accept positive integer values.

In chan_dahdi, the return value of dahdi_get_index is used to directly
index into the sub-channel array.  If dahd_get_index returns a negative
value, it would use that value to index into the array, which could cause
an invalid memory access.  If dahdi_get_index returns a negative number,
we now default to SUB_REAL.

(issue ASTERISK-19655)
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/1863/
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Merged revisions 362204 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 362205 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362206 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-16 21:58:06 +00:00
Matthew Jordan ebcccf8997 Fix handling of negative return code when storing voicemails in ODBC storage
When storing a voicemail message using an ODBC connection to a database, the
voicemail message is first stored on disk.  The sound file associated with
the message is read into memory before being transmitted to the database.
When this occurs, a failure in the C library's lseek function would cause a
negative value to be passed to the mmap as the size of the memory map to
create.  This would almost certainly cause the creation of the memory map to
fail, resulting in the message being lost.

(issue ASTERISK-19655)
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/1863
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Merged revisions 362201 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 362202 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362203 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-16 21:42:12 +00:00
Michael L. Young abf40d9b28 Add IPv6 address support to security events framework.
The current Security Events Framework API only supports IPv4 when it comes to
generating security events.  This patch does the following:

* Changes the Security Events Framework API to support IPV6 and updates
  the components that use this API.

* Eliminates an error message that was being generated since the current
  implementation was treating an IPv6 socket address as if it was IPv4.

* Some copyright dates were updated on files touched by this patch.

(closes issue ASTERISK-19447) 
Reported by: Michael L. Young 
Tested by: Michael L. Young 
Patches: 
  security_events_ipv6v3.diff uploaded by Michael L. Young (license 5026)

Review: https://reviewboard.asterisk.org/r/1777/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362200 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-16 21:20:50 +00:00
Matthew Jordan ba7032be5f Check for IO stream failures in various format's truncate/seek operations
For the formats that support seek and/or truncate operations, many of
the C library calls used to determine or set the current position indicator
in the file stream were not being checked.  In some situations, if an error 
occurred, a negative value would be returned from the library call.  This
could then be interpreted inappropriately as positional data.

This patch checks the return values from these library calls before
using them in subsequent operations.

(issue ASTERISK-19655)
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/1863/
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Merged revisions 362151 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 362152 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362153 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-16 20:17:03 +00:00
Jonathan Rose ba0f044bde Make ForkCDR e option not set end time of the newly forked CDR log
Prior to this patch, ForkCDR's e option would immediately set the end time of the forked
CDR to that of the CDR that is being terminated. This resulted in the new CDR's end time
being roughly the same as it's beginning time (which is in turn roughly the same as the
original's end time).

(closes issue ASTERISK-19164)
Reported by: Steve Davies
Patches:
	cdr_fork_end.v10.patch uploaded by Steve Davies (license 5012)
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Merged revisions 362082 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 362084 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362085 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-13 16:12:17 +00:00
Jonathan Rose c0b9fe8530 Send relative path named recordings to the meetme directory instead of sounds
Prior to this patch, no effort was made to parse the path name to determine a proper
destination for recordings of MeetMe's r option. This fixes that.

Review: https://reviewboard.asterisk.org/r/1846/
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Merged revisions 362079 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 362080 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362081 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-13 15:38:08 +00:00
Paul Belanger 05eb51bb2c Convert SRV lookup message to debug level
This helps clean up the Asterisk CLI by converting the log message from verbose
to debug


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362043 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-12 20:08:26 +00:00
Richard Mudgett a35c7ba8e7 Add option to invoke the extensions.conf stdexten using the legacy macro method.
ASTERISK-18809 eliminated the legacy macro invocation of the stdexten in
favor of the Gosub method without a means of backwards compatibility.

(issue ASTERISK-18809)
(closes issue ASTERISK-19457)
Reported by: Matt Jordan
Tested by: rmudgett

Review: https://reviewboard.asterisk.org/r/1855/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361998 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-12 16:29:52 +00:00
Kinsey Moore f9155c9c3d Make trunkfreq take effect when set
Previously, setting trunkfreq had no effect on initial load or on reload and
only ever used the default value.  This causes trunkfreq to be used 
appropriately on initial load and reload.

(closes issue ASTERISK-19521)
Patch-by: Jaco Kroon
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Merged revisions 361972 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 361981 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361987 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-12 16:25:09 +00:00
Kinsey Moore 8696daadf8 Simplify build system architecture optimization
This change to the build system rips out any usage of PROC along with
architecture-specific optimizations in favor of using -march=native where it is
supported.  This fixes broken builds on 64bit Intel systems and results in
better optimized code on systems running GCC 4.2+.

Review: https://reviewboard.asterisk.org/r/1852/
(closes issue ASTERISK-19462)
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Merged revisions 361955 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 361956 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361968 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-12 15:25:47 +00:00
Jonathan Rose 683eacb59a Change default value of 'ignorebusy' on Queue members so that behavior is more like 1.8
Prior to this patch, in order to restore that behavior, a function would have
to be used on the QueueMember to make the ringinuse option do anything, which
is pretty unreasonable.

(closes issue ASTERISK-19536)
reported by: Philippe Lindheimer
Review: https://reviewboard.asterisk.org/r/1860/
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Merged revisions 361907 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361909 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-11 17:20:08 +00:00
Richard Mudgett 198046d706 Prevent invalid access of free'd memory if DAHDI channel during an MWI event
In the MWI processing loop, when a valid event occurs the temporary caller ID
information is deallocated.  If a new DAHDI channel is successfully created,
the event is passed up to the analog_ss_thread without error and the loop
exits.  If, however, the DAHDI channel is not created, then the caller ID
struct has been free'd, and the gains reset to their previous level.  This
will almost certainly cause an invalid access to the free'd memory, either
in subsequent calls to callerid_free or calls to callerid_feed.

* Rework the -r361705 patch to better manage the cs and mtd allocated
resources.

* Fixed use of mwimonitoractive flag to be correct if the mwi_thread()
fails to start.
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Merged revisions 361854 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361856 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-10 21:50:46 +00:00
Matthew Jordan 3d7b9e7fb1 Fix crash caused by unloading or reloading of res_http_post
When unlinking itself from the registered HTTP URIs, res_http_post could
inadvertently free all URIs registered with the HTTP server.  This patch
modifies the unregister method to only free the URI that is actually
being unregistered, as opposed to all of them.
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Merged revisions 361803 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 361804 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361805 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-10 19:58:04 +00:00
Matthew Jordan 38c0a62413 Allow func_curl to exit gracefully if list allocation fails during write
If the global_curl_info data structure could not be allocated, the
datastore associated with the operation would be free'd, but the function
would not return.  This would later dereference the datastore, almost
certainly causing Asterisk to crash.  With this patch, if the data
structure is not allocated the method will return an error code, and
not attempt any further operation.
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Merged revisions 361753 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 361754 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361755 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-09 21:47:54 +00:00
Matthew Jordan 97f813f3a4 Prevent invalid access of free'd memory if DAHDI channel during an MWI event
In the MWI processing loop, when a valid event occurs the temporary caller ID
information is deallocated.  If a new DAHDI channel is successfully created, 
the event is passed up to the analog_ss_thread without error and the loop
exits.  If, however, the DAHDI channel is not created, then the caller ID
struct has been free'd, and the gains reset to their previous level.  This
will almost certainly cause an invalid access to the free'd memory, either
in subsequent calls to callerid_free or calls to callerid_feed.

This patch makes it so that we only free the caller ID structure if a
DAHDI channel is successfully created, and we bump the gains back up
if we fail to make a DAHDI channel.
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Merged revisions 361705 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 361706 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361707 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-09 20:55:53 +00:00
Matthew Jordan f4fd1b2fb0 Change SHARED function to use a safe traversal when modifying a variable
When the SHARED function modifies a variable, it removes it from its list of
variables and reinserts the new value at the head of the list of variables.
Doing this inside a standard list traversal can be dangerous, as the
standard list traversal does not account for the list being changed.  While
the code in question should not cause a use after free violation due to its
breaking out of the loop after freeing the variable, it could lead to a
maintenance issue if the loop was modified.  This also fixes a violation
reported by a static analysis tool, which also makes this code easier to
maintain in the future. 
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Merged revisions 361657 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 361658 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361659 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-09 19:44:35 +00:00
Matthew Jordan 90226b6fd7 Fix memory leak in res_calendar_ews when event email address node is empty
If the XML calendar data returned by a Microsoft Exchange Web Service
specifies an XML Event E-Mail Address ("EmailAddress"), and no e-mail address
is provided, a condition existed where an ast_calendar_attendee struct would
be allocated but not appended to the list of attendees.  Because of that,
the memory associated with the attendee would never be freed.  This patch
frees the memory if no e-mail address is provided.
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Merged revisions 361606 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 361607 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361608 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-06 22:00:58 +00:00
Matthew Jordan aa21d4fc6b Fix memory leak when using MeetMeAdmin 'e' option with user specified
A memory leak/reference counting leak occurs if the MeetMeAdmin 'e' command
(eject last user that joined) is used in conjunction with a specified user.
Regardless of the command being executed, if a user is specified for the
command, MeetMeAdmin will look up that user.  Because the 'e' option kicks
the last user that joined, as opposed to the one specified, the reference to
the user specified by the command would be leaked when the user variable
was assigned to the last user that joined.
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Merged revisions 361558 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361561 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-06 20:32:52 +00:00
Richard Mudgett 4665986fb1 Don't add an empty MESSAGE_DATA(key) header if it doesn't already exist.
Doing Set(MESSAGE_DATA(key)=) would add an empty key header if the key
header did not already exist.  If it already existed it would delete it.

* Made msg_set_var_full() exit early if the named variable did not already
exist and the value to set is empty.
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Merged revisions 361522 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361523 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-06 19:58:44 +00:00
Kinsey Moore a485f44022 Add missing newlines to CLI logging
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Merged revisions 361471 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361476 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-06 18:19:03 +00:00
Paul Belanger fcb7eb3c59 Multiple revisions 361403,361412
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  r361403 | pabelanger | 2012-04-06 12:24:36 -0400 (Fri, 06 Apr 2012) | 2 lines
  
  Fix typo in svn:keywords
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  r361412 | pabelanger | 2012-04-06 12:27:30 -0400 (Fri, 06 Apr 2012) | 2 lines
  
  Fix typo in svn:keywords
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361429 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-06 16:33:24 +00:00
Russell Bryant b2f7b0c649 Remove a few more files related to chan_usbradio and app_rpt.
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Merged revisions 361380 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361382 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-06 15:50:18 +00:00
Matthew Jordan a2e127a651 Fix a typo in the warning messages for an ignored media stream
Added a '\n' to the warning messages when we ignore a media stream due to the
port number being '0'.

(closes issue ASTERISK-19646)
Reported by: Badalian Vyacheslav
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Merged revisions 361332 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361334 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-06 14:02:16 +00:00
Kinsey Moore 51f0e5c53d Remove unnecessary error message in app_dial.c
The error message for failure to stop autoservice after a gosub or macro call
during a dial was removed for macro while Asterisk 1.4 was still being actively
developed. The corresponding gosub error message was never removed.

(closes issue ASTERISK-19551)
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Merged revisions 361329 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361331 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-06 13:32:34 +00:00
Jonathan Rose fc45af331b Fix MusicOnHold in MeetMe so that it always uses the class if it's been defined
There were a few instances of restarting music on hold in meetme that would cause
Asterisk to revert to the default class of music on hold for no adequate reason.

Review: https://reviewboard.asterisk.org/r/1844/
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Merged revisions 361269 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361279 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-05 17:22:30 +00:00
Jonathan Rose 8d3855229a Fix some stuff involving calls to memcpy and memset
The important parts of the patch were already applied through other updates.

(closes issue ASTERISK-19445)
Reported by: Makoto Dei
Patches:
	memset-memcpy-length.patch uploaded by Makoto Dei (license 5027)
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Merged revisions 361210 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361212 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-04 20:10:08 +00:00
Jonathan Rose 6cd462cca8 Make 'help devstate change' display properly (get rid of excess comma)
(closes issue ASTERISK-19444)
Reported by: Makoto Dei
Patches:
	devstate-change-usage-truncate.patch uploaded by Makoto Dei (license 5027)
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Merged revisions 361201 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 361208 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361209 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-04 19:32:57 +00:00
Jonathan Rose e96a59acfd Replace GNU old-style field designator extensions to fix clang warnings
(issue ASTERISK-19540)
Reported by: Makoto Dei
Patches:
	clang-gnu-designator.patch uploaded by Makoto Dei (license 5027)
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Also add from the patch the portion in res_fax_spandsp that didn't apply to 1.8

Merged revisions 361142 from http://svn.asterisk.org/svn/asterisk/branches/1.8
(closes issue ASTERISK-19540)
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Merged revisions 361143 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361155 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-04 18:08:28 +00:00
Jonathan Rose 97b2fa8de1 Make the MeetMeAdmin N command (mute all nonadmins) not mute admins
(Closes Issue ASTERISK-19335)
Reported by: Johan Wilfer
Review: https://reviewboard.asterisk.org/r/1843/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361092 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-04 13:51:45 +00:00
Kinsey Moore 93781fa161 Fix the display of documentation for Transfer
This came up while fixing documentation generation for many other cases where
the argument separator was not being displayed properly.  Now that it is
displayed properly, it shows up in the wrong place for Transfer since the '/'
is only required if Tech is present.

(related to issue ASTERISK-18168)
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Merged revisions 361040 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361042 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-03 20:14:01 +00:00
Mark Murawki 745fcdbffe Fix dev-mode compiler warning about gnu_printf
(related to ASTERISK-19575)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361039 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-03 20:03:44 +00:00
Mark Murawki e4252eac10 Allow the Hangup manager action to match channels by regex
* Hangup now can take a regular expression as the Channel option.  If you want
  to hangup multiple channels, use /regex/ as the Channel option.  Existing
  behavior to hanging up a single channel is unchanged, but if you pass a regex,
  the manager will send you a list of channels back that were hung up.

(closes issue ASTERISK-19575)
Reported by: Mark Murawski
Tested by: Mark Murawski



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361038 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-03 19:31:25 +00:00
Kinsey Moore 9cc6f2c59e Stop sending out RTCP if RTP is inactive
This change prevents Asterisk from sending RTCP receiver reports during a
remote bridge since it is no longer receiving media and should not be
reporting anything.

(related to ASTERISK-19366)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360994 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-02 22:27:13 +00:00
Richard Mudgett 6a540e9087 Fix logger deadlock on Asterisk shutdown.
The logger_thread() had an exit path that failed to release the logmsgs
list lock.

* Make logger_thread() exit path unlock the logmsgs list lock.

* Made ast_log() not queue any messages to the logmsgs list if the
close_logger_thread flag is set.

(issue ASTERISK-19463)
Reported by: Matt Jordan
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360935 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-30 21:38:16 +00:00
Mark Michelson 314d459317 Fix potential race condition during call pickup.
Prior to this patch, a connected line update was queued during
call pickup and then an answer frame was queued. The original
caller would presumably then have his connected line updated
and then the call would be answered.

In actuality, the answer frame was not how the call ended up
being answered. Rather, an odd section in app_dial that checks
if the called channel's state is up.

The result is that the order of the connected line update and
the answer were variable. In most cases, this wasn't actually
a bad thing. However, if the 'I' option was passed to dial, the
connected line update would be inhibited.

The fix is to queued the connected line after the answer frame is
queued. This way the race in app_dial is between two
conditions resulting in an answer. This way the connected line
update occurs after the answer every time.

(closes issue ASTERISK-19183)
Reported by: Thomas Arimont
Tested by: Thomas Arimont
    Mark Michelson
Patches:
    ASTERISK-19183.patch uploaded by Mark Michelson (license 5049)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360886 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-29 23:36:37 +00:00
Mark Michelson cc2366bca0 Improve accuracy of identifying information sent in dialog-info SIP NOTIFY requests.
This change makes use of connected party information in addition to caller ID in order
to populate local and remote XML elements in the dialog-info NOTIFYs.

(closes issue ASTERISK-16735)
Reported by: Maciej Krajewski
Tested by: Maciej Krajewski
Patches:
    local_remote_hint2.diff uploaded by Mark Michelson (license 5049)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360872 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-29 23:22:01 +00:00
Richard Mudgett fb796aac06 Misc changes to make astobj2 enhancement diffs easier to follow.
* Rename astobj2 API parameter funcname to func.

* Rename astobj2 API iterator parameter to iter.

* Update some documentation for OBJ_MULTIPLE.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360827 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-29 21:57:23 +00:00
Jonathan Rose 655a8d4420 Introducing the log message unique call identifiers feature
Log messages will now display a call number that they are tied to (ordered for calls
based on when they started). This feature is made to be minimally invasive without
requiring changes to many of the existing log messages. These IDs  won't show up for
verbose messages on CLI (but they will in log files) This is currently in phase II
of production, see more about this feature on the wiki --
https://wiki.asterisk.org/wiki/display/AST/Unique+Call-ID+Logging

Review: https://reviewboard.asterisk.org/r/1823/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360787 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-29 20:01:20 +00:00
Jonathan Rose d501c2ea2d undoing 360785 due to merging mistake
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360786 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-29 19:59:30 +00:00
Jonathan Rose bf994f0e04 Introducing the log message unique call identifiers feature
Log messages will now display a call number that they are tied to (ordered for calls
based on when they started). This feature is made to be minimally invasive without
requiring changes to many of the existing log messages. These IDs  won't show up for
verbose messages on CLI (but they will in log files) This is currently in phase II
of production, see more about this feature on the wiki --
https://wiki.asterisk.org/wiki/display/AST/Unique+Call-ID+Logging

Review: https://reviewboard.asterisk.org/r/1823/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360785 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-29 19:54:35 +00:00
Terry Wilson dd9405db05 Fix setting CDR variables in the hangup extension
A previous CDR fix for setting CDR variables during a bridge via
custom dialplan features broke setting CDR variables in the
hangup extension. This patch fixes the issue.

Review: https://reviewboard.asterisk.org/r/1794/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360724 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-28 19:39:24 +00:00
Mark Michelson 01cc64585e Make a debug message regarding subscription changes more accurate.
I was getting confused during some testing why Asterisk was saying that
a subscription was being added when it was clearly being removed. This
fixes that confusion.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360673 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-27 18:44:53 +00:00
Richard Mudgett 38e892b370 Add global ao2 array container.
Global ao2 objects must always exist after initialization because there is
no access control to obtain another reference to the global object.

It is expected that module configuration could use these new API calls to
replace an active configuration parameter object with an updated
configuration parameter object.

With these new API calls, the global object could be replaced, removed, or
referenced without the risk of someone using a stale global object
pointer.

Review: https://reviewboard.asterisk.org/r/1824/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360627 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-27 17:13:32 +00:00
Richard Mudgett 8611bea122 Attempt to be more helpful when using a bad ao2 object pointer.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360626 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-27 17:00:13 +00:00
Jonathan Rose 65f56cda7a Updates config with bootstrap where I changed configure.ac in r360488
(issue ASTERISK-17842)
Reported by: Bryon Clark
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360576 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-27 14:43:00 +00:00
Paul Belanger 8331a89892 Blocked revisions 360476
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Update CHANGES for r360471
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360537 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-26 21:36:24 +00:00
Paul Belanger dea8936f89 Convert ast_verb() to ast_debug() and increase log level
Rather then flood the CLI with verbose messages, we've changed the level to
debug. This will help keep the CLI clean.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360536 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-26 21:22:23 +00:00
Jonathan Rose 337f933977 Fix BETTER_BACKTRACES library detection for Fedora/RedHat/CentOS
(closes ASTERISK-17842)
Reported by: Bryon Clark
Patches:
	20110512__issue19278.diff.txt uploaded by Tilghman Lesher (license 5003)
	configure_bfd_with_dl_and_iberty.patch uploaded by Bryon Clark (license 6157)
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2012-03-26 19:49:40 +00:00