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
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
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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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
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
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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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
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
* 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
This changes the debuglevel of 'pri set debug' to a bit mask allowing the user
to independently select bits of output:
1 libpri internals including state machine
2 Decoded Q.931 messages
4 Decoded Q.921 headers
8 raw hex dump of the full frames
Additionally, this ensures that the meaning of "on" does not change and
intrudces intense and hex to simplify usage.
(closes issue ASTERISK-17159)
Original-patch-by: wimpy
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354165 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch removes res_ais and introduces a new module, res_corosync.
The OpenAIS project is deprecated and is now just a wrapper around
Corosync. This module provides the same functionality using the same
core infrastructure, but without the use of the deprecated components.
Technically res_ais could have been used with an AIS implementation other
than OpenAIS, but that is the only one I know of that was ever used.
Review: https://reviewboard.asterisk.org/r/1700/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354046 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Asterisk now supports using ODBC with databases where a single schema must be
selected. Previously, INSERTs would fail because they did not take into
account extra fields cause by having multiple schemas. This also corrects
some SQL resource leaks.
(closes issue ASTERISK-17106)
Patch-by: Alexander Frolkin
Patch-by: Tilgnman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353964 65c4cc65-6c06-0410-ace0-fbb531ad65f3
After R340970 Asterisk was still polling the RTCP file descriptor after RTCP is
shut down and removed. If the descriptor happened to have data ready when the
removal occured then Asterisk would go into an infinite loop trying to read
data that it can never actually access. This change disables the audio RTCP
file descriptor for the duration of the T.38 transaction.
(closes issue ASTERISK-18951)
Reported-by: Kristijan Vrban
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353917 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This feature also causes the sending complete ie to be sent for switch
types that do not automatically send the ie. (EuroISDN/ETSI)
The main difference between dialing Dial(DAHDI/g0/1234w888) and
Dial(DAHDI/g0/1234,,D(888)) is the sending of the sending complete ie.
(closes issue ASTERISK-19176)
Reported by: rmudgett
Tested by: rmudgett
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* Removes references to tlsbindport from http.conf.sample and manager.conf.sample
* Properly bind to port specified in tlsbindaddr, using the default port if specified.
* On a reload, properly close socket if the service has been disabled.
A note has been added to UPGRADE.txt to indicate how ports must be set for TLS.
(closes issue ASTERISK-16959)
reported by Olaf Holthausen
(closes issue ASTERISK-19201)
reported by Chris Mylonas
(closes issue ASTERISK-19204)
reported by Chris Mylonas
Review: https://reviewboard.asterisk.org/r/1709
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353821 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A previous patch I committed from ASTERISK-16930 unexpectedly changed some output for
the AMI action "sippeers" which this patch changes back. Also, this aligns the output
for the cli command "sip show peers" and fixes another issue that patch introduced by
using ast_sockaddr_stringify calls multiple times without immediately using the pointer.
I also went ahead and did a little janitorial work to clean up whitespace in
_sip_show_peers.
(issue ASTERISK-16930)
(closes issue ASTERISK-19281)
Reported by: Patrick El Youssef
Patches:
ASTERISK-19281.diff uploaded by Walter Doekes (license 5674)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353772 65c4cc65-6c06-0410-ace0-fbb531ad65f3
There are a number of cleaner looking wrappers for ast_sockaddr_stringify_fmt
available which are slightly more readable than using a direct call to
ast_sockaddr_stringify_fmt. This patch switches a number of those calls in
chan_sip to use those wrappers and is generally harmless.
(Closes issue ASTERISK-16930)
Reported by: Michael L. Young
Patches:
chan_sip-broken-registration-1.8.diff uploaded by Michael L. Young (license 5026)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353725 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Richard Mudgett
Mark Michelson
Jason Parker
Sean Bright
Alexandr Anikin
Russell Bryant
Joshua Colp
Terry Wilson
Matthew Jordan
Kinsey Moore
Kevin P. Fleming
Walter Doekes
Jonathan Rose