Correct Documentation For DEC Function

The documentation for DEC in func_math.c was incorrect.  Looks like a copy and
paste error.

(Closes issue ASTERISK-20095)
Reported by: Billy Chia
Tested by: Michael L. Young
Patches:
    func_math.patch uploaded by Billy Chia (license 6381)
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Merged revisions 369970 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Reverting last merge since it wasn't completed properly.
Correct Documentation For DEC Function

The documentation for DEC in func_math.c was incorrect.  Looks like a copy and
paste error.

(Closes issue ASTERISK-20095)
Reported by: Billy Chia
Tested by: Michael L. Young
Patches:
    func_math.patch uploaded by Billy Chia (license 6381)
........

Merged revisions 369970 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 369971 from http://svn.asterisk.org/svn/asterisk/branches/10
Prevent double uri_escaping in chan_sip when pedantic is enabled

If pedantic mode is enabled, outbound invites will have double-escaped
contacts.  This avoids setting an already-escaped string into a field
where it is expected to be unescaped.

(closes issue ASTERISK-20023)
Reported-by: Walter Doekes
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Merged revisions 369993 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 369994 from http://svn.asterisk.org/svn/asterisk/branches/10
Include Expires header for SIP PUBLISH requests

RFC3903 requres SIP PUBLISH requests to have Expires headers, so add
them.

Review: https://reviewboard.asterisk.org/r/2003/
Patch-by: gareth
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Merged revisions 370014 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 370015 from http://svn.asterisk.org/svn/asterisk/branches/10
Add missing ast_hangup() calls on some analog exception paths.

Make starting analog_ss_thread() or __analog_ss_thread() failure paths
hangup the channel.
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Merged revisions 370017 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 370025 from http://svn.asterisk.org/svn/asterisk/branches/10
Handle deprecated (aliased) option names with the config options api

Add a simple way to register "deprecated" option names that alias to a
different "current" name.

Review: https://reviewboard.asterisk.org/r/2026/
live_ast: don't set working directory

contrib/scripts/live_ast currently assumes that it is being run from the
top-level directory of the source tree. It creates a script that will
also set the working directory.

This fix avoids the need to set the working directory if the caller sets
LIVE_AST_BASE_DIR instead.

It relies on realpath for that. If realpath is not available, it will
fall back to the original behaviour.

Review: https://reviewboard.asterisk.org/r/2027/
Add support for parsing SDP attributes, generating SDP attributes, and passing it through.

This support includes codecs such as H.263, H.264, SILK, and CELT. You are able to set up a call and have attribute information pass. This should help considerably with video calls.

Review: https://reviewboard.asterisk.org/r/2005/
Reduce memory consumption and add the H.264 and H.263 modules I shamefully neglected to add.

Add French translation for chan_unistim phones on-screen menus.


Deactivate timer for dialing entered number on hook switch hang up.

(closes issue ASTERISK-19554)
Reported by: Stefano Villani
Add support for SIP over WebSocket.

This allows SIP traffic to be exchanged over a WebSocket connection which is useful for rtcweb.

Review: https://reviewboard.asterisk.org/r/2008
Fix an issue where specifying the resource in the username would cause authentication to fail.
Add comments about the BUILD_NATIVE change

This is a significant change and mention of it should have gone into
UPGRADE.txt and CHANGES.
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Merged revisions 370081 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 370082 from http://svn.asterisk.org/svn/asterisk/branches/10
Fix a bug exposed by the testsuite where text streams would no longer be parsed correctly.
Add missing namespace for old non-SASL based authentication.
Fix a bug where some XMPP servers would reject authentication. We need to use the user portion of the JID and not the full configured username.
Fix an issue where a service discovery request could crash Asterisk.

A server sending a service discovery request to us may or may not put a from attribute in the message. If the from attribute is present use it in the to attribute for the result. If the from attribute is not present do not add a to attribute.

(issue ASTERISK-16203)
Reported by: wubbla
Code cleanup and bugfix in chan_sip outboundproxy parsing.

The bug was clearing the global outboundproxy when a peer-specific
outboundproxy was bad. The cleanup reduces duplicate code.

Review: https://reviewboard.asterisk.org/r/2034/
Reviewed by: Mark Michelson
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Merged revisions 370131 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 370132 from http://svn.asterisk.org/svn/asterisk/branches/10