git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355495 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Back in r646, TRUNK_CALL_START was added and defined as 0x4000.  That same value
was also hard-coded in one part of the IAX2 code instead of using the #define.

TRUNK_CALL_START has changed over the years (for dealing with LOW_MEMORY), but
the hard-coded usage was never updated to match.  This patch fixes that.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355450 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This change permits each verbose destination (consoles, logger) to have its
own concept of what the verbosity level is.  The big feature here is that
the logger will now be able to capture a particular verbosity level without
condemning each console to need to suffer that level of verbosity.
Additionally, a stray 'core set verbose' will no longer change what will go
to the log.

Review:  https://reviewboard.asterisk.org/r/1599/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355413 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Ogg/vorbis was fairly useless as a voicemail file format because it did
not implement the seek and tell format callbacks among other problems.

Since we were already using the libvorbis and libvorbisenc libraries we
can use libvorbisfile as it is also part of the vorbis library package.

* Made use the libvorbisfile to handle the ogg/vorbis file stream.  The
format_ogg_vorbis.c is now mostly a wrapper around libvorbisfile.

(closes issue ASTERISK-16926)
Reported by: sque
Patches:
      ogg_vorbis_use_libvorbisfile.patch (license #6108) patch uploaded by sque
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355376 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue ASTERISK-19356)
Reported by: Alex Villacis Lasso
Patches:
      asterisk-1.8.9.2-cel_sqlite3_custom-fix-reload-locking-typo.patch (license #5617) patch uploaded by Alex Villacis Lasso

Review: https://reviewboard.asterisk.org/r/1740/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355321 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This was causing identification that should have been
made private to be public.

(closes issue AST-814)
reported by Patrick Anderson

Patches:
	chan_sip.c.diff uploaded by Patrick Anderson (license 5430)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355230 65c4cc65-6c06-0410-ace0-fbb531ad65f3

send_apathetic_reply takes the incoming frame's source call number as the
destination call number for the outgoing frame.  If the incoming frame was a
full frame, then the high order bit of the source call number is set and will be
interpreted as a retransmit when sent back out as the destination call number.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355184 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(Closes issue ASTERISK-19298)
Reported by: robinfood
Patches:
        issue19298.patch uploaded by may213 (License #5415)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355138 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch adds a variable AGIEXITONHANGUP for res_agi.  If this variable is
set to "yes" on a channel, AGI() will exit immediately once a channel hangup
has been detected.  This was the behavior of AGI() in Asterisk 1.4 and earlier
and is still desired by some people.

Review: https://reviewboard.asterisk.org/r/1734/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355102 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Since the dir timestamp is available at one second resolution, we cannot
know if it was updated within the same second after we scanned it.
Therefore, we will force another scan if the dir was just modified.

* Changed to force another scan if the directory was just modified.

(closes issue ASTERISK-19081)
Reported by: Knut Bakke

Review: https://reviewboard.asterisk.org/r/1688/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355058 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355055 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Only allow one 'dialplan reload' to execute at a time as otherwise they would share the same common local context list.

(closes issue AST-758)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355011 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Review: https://reviewboard.asterisk.org/r/1733/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354968 65c4cc65-6c06-0410-ace0-fbb531ad65f3

There can only be one database connection in res_config_pgsql just like
res_config_sqlite.  If the connection is lost, the connection may not get
reestablished to the same database if the res_pgsql.conf and
extconfig.conf files are inconsistent.

* Made only use the configured database from res_pgsql.conf.

* Fixed potential buffer overwrite of last[] in config_pgsql().

(closes issue ASTERISK-16982)
Reported by: german aracil boned

Review: https://reviewboard.asterisk.org/r/1731/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354964 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue ASTERISK-19188)
Reported by: slesru
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354939 65c4cc65-6c06-0410-ace0-fbb531ad65f3

open_mailbox() was changed quite a long time ago to allocate this memory.
close_mailbox() should have been changed to be responsible for freeing it.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354903 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The astman_get_header() never returns NULL so the check by the code for
NULL would never fail.

(closes issue ASTERISK-16974)
Reported by: Nuno Borges
Patches:
      0018325.patch (license #6116) patch uploaded by Nuno Borges (modified)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354837 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This simply fixes the compilation issue introduced in r354429 by
re-adding the 'quote' variable.

(closes issue ASTERISK-19337)
Reported by: John Taylor


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354799 65c4cc65-6c06-0410-ace0-fbb531ad65f3

CDRs cannot be modified after a bridge is torn down, (e.g. after
Dial() returns) even though the CDR() function may be called. Since
modifying the CDR code to change this behavior could very easily
break all kinds of things, this patch just documents this limitation.

(closes issues ASTERISK-16923)
Review: https://reviewboard.asterisk.org/r/1720/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354751 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Change parsing of SIP headers so that compactness of the header no longer
influences which header will be chosen.  Previously, a non-compact header
would be chosen instead of a preceeding compact-form header.

(closes issue ASTERISK-17192)
Review: https://reviewboard.asterisk.org/r/1728/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354704 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The config parser in Asterisk does not currently remove a backslash that is
used to escape a semicolon which would otherwise be interpreted as the start
of a comment.

The change here causes that backslash to be removed, but does not create a
real escape system in the config parser.  The biggest complication with a real
escape system would be breaking existing configs everywhere (parsing \\ as \
and breaking on escaped non-semicolon characters) even though it would be the
"right" way to do things.

(closes issue ASTERISK-17121)
Review: https://reviewboard.asterisk.org/r/1724/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354657 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch adds the auto_force_rport and auto_comedia NAT options. It
also converts the nat= setting to a list of comma-separated combinable
options: no, force_rport, comedia, auto_force_rport, and auto_comedia.
nat=yes remains as an undocumented option equal to
"force_rport,comedia". The first instance of 'yes' or 'no' in the list
stops parsing and overrides any previously set options. If an auto_*
option is specified with its non-auto_ counterpart, the auto setting
takes precedence.

This patch builds upon the patch posted to ASTERISK-17860 by JIRA user
pedro-garcia.

(closes issue ASTERISK-17860)
Review: https://reviewboard.asterisk.org/r/1698/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354597 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue ASTERISK-16712)
reported by Frank DiGennaro

Review: https://reviewboard.asterisk.org/r/1713
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354552 65c4cc65-6c06-0410-ace0-fbb531ad65f3

rmudgett pointed out some formatting issues in the check-in for
ASTERISK-19290.  This cleans those up.

Review: https://reviewboards.asterisk.org/r/1722/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354549 65c4cc65-6c06-0410-ace0-fbb531ad65f3

In ASTERISK-18924, SIP INFO DTMF handlingw as changed to account for both
lowercase alphatbetic DTMF events, as well as uppercase alphabetic DTMF
events.  When this occurred, the comparison of the character buffer containing
the event code was changed such that the buffer was first compared again '0'
and '9' to determine if it was numeric.  Unfortunately, since the first
character in the buffer will typically be '1' in the case of non-numeric
event codes (10-16), this caused those codes to be converted to a DTMF event
of '1'.  This patch fixes that, and cleans up handling of both
application/dtmf-relay and application/dtmf content types.

Review: https://reviewboard.asterisk.org/r/1722/

(closes issue ASTERISK-19290)
Reported by: Ira Emus
Tested by: mjordan
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354544 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354498 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Well, thats embarrasing.  I forgot to initialize the caller_id storage.

(closes issue ASTERISK-19311)
Reported by: tootai
Tested by: rmudgett
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354497 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch removes some unnecessary locking of the channels container in
ast_hangup().  The reason this came up is that this lock can very quickly block
the entire system.  If any of the channel cleanup code decides to block, it
causes a problem for the whole system.  For example, when audiohooks get
destroyed, if that blocks for a while waiting on the mixmonitor thread to exit
because it's busy blocking on some I/O, it causes a problem for many other
threads in the meantime.

Review: https://reviewboard.asterisk.org/r/1712/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354494 65c4cc65-6c06-0410-ace0-fbb531ad65f3

actually remove res_ais. This commit removes it.

In addition, the 'install_prereq' script has been updated to no longer install
AIS dependency packages, and instead install Corosync packages instead.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354459 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch is based on the one by David Vossel, developer extrodinaire, at
https://reviewboard.asterisk.org/r/344/. If multiple peers are defined with the
same host/port, but differing callbackextensions, it chooses the peer with the
matching callbackextension. Since callbackextension creates an outbound
registration with the callbackextension as the Contact address, matching an
incoming request by that (in addition to the host/port) makes a lot of sense.

This patch also adds support for callbackextension to realtime by querying all
peers with callbackextensions on reload and adding registrations for them.

(closes issue ASTERISK-13456)
Review: https://reviewboard.asterisk.org/r/344/
Review: https://reviewboard.asterisk.org/r/1717/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354458 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354450 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Patch by: Clod Patry
Review: https://reviewboard.asterisk.org/r/1651


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354429 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(related to ASTERISK-17159)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354395 65c4cc65-6c06-0410-ace0-fbb531ad65f3

1. Set lastms to 0 when clearing instead of ""
2. Don't set ipaddr or port to the string "(null)" when they are empty
3. Add missing required fields, set default for lastms to 0, and modify
   the length of the ipaddr field to 45 in the Postgresql realtime.sql
   file.

(closes issue ASTERISK-19172)
Review: https://reviewboard.asterisk.org/r/1703/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354360 65c4cc65-6c06-0410-ace0-fbb531ad65f3

be empty, don't check for it, instead of check if LD_LIBRARY_PATH is
already set and if so, append LIVE_AST_LD_PATH_EXTRA properly.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354314 65c4cc65-6c06-0410-ace0-fbb531ad65f3

libasteriskssl.so at run time when using live_ast.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354313 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354312 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Prior to this patch, attempts to reload cdr_pgsql.so would cause the column list to keep
its current data and then add a second copy during the reload. This would cause attempts
to log the CDR to the database to fail. This patch also cleans up some unnecessary null
checks for ast_free and deals with a few potential locking problems.

(closes issue ASTERISK-19216)
Reported by: Jacek Konieczny
Review: https://reviewboard.asterisk.org/r/1711/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354275 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Documented dialplan add extension <exten>,<priority>,<app(<app-data>)>
format.

* Allow acceptance of command without the app-data value.  There are many
applications that do no need any parameters so it is silly to require that
field for all commands.

* Fixed a couple ast_malloc/ast_free mismatches with ast_add_extension2()
calls.

(closes issue ASTERISK-19222)
Reported by: Andrey Solovyev
Tested by: rmudgett
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354218 65c4cc65-6c06-0410-ace0-fbb531ad65f3