asterisk/contrib
Nick French 37b2e68628 res_pjsip: Implement additional SIP RFCs for Google Voice trunk compatability
This change implements a few different generic things which were brought
on by Google Voice SIP.

1.  The concept of flow transports have been introduced.  These are
configurable transports in pjsip.conf which can be used to reference a
flow of signaling to a target.  These have runtime configuration that can
be changed by the signaling itself (such as Service-Routes and
P-Preferred-Identity).  When used these guarantee an individual connection
(in the case of TCP or TLS) even if multiple flow transports exist to the
same target.

2.  Service-Routes (RFC 3608) support has been added to the outbound
registration module which when received will be stored on the flow
transport and used for requests referencing it.

3.  P-Associated-URI / P-Preferred-Identity (RFC 3325) support has been
added to the outbound registration module.  If a P-Associated-URI header
is received it will be used on requests as the P-Preferred-Identity.

4.  Configurable outbound extension support has been added to the outbound
registration module.  When set the extension will be placed in the
Supported header.

5.  Header parameters can now be configured on an outbound registration
which will be placed in the Contact header.

6.  Google specific OAuth / Bearer token authentication
(draft-ietf-sipcore-sip-authn-02) has been added to the outbound
registration module.

All functionality changes are controlled by pjsip.conf configuration
options and do not affect non-configured pjsip endpoints otherwise.

ASTERISK-27971 #close

Change-Id: Id214c2d1c550a41fcf564b7df8f3da7be565bd58
2018-10-24 07:51:25 -05:00
..
ast-db-manage res_pjsip: Implement additional SIP RFCs for Google Voice trunk compatability 2018-10-24 07:51:25 -05:00
docker Add initial support to build Docker images 2016-03-17 09:53:57 -05:00
editors contrib/editors: Fix vim syntax highlighting of comments in config files 2015-05-07 19:37:42 +00:00
init.d Remove as much trailing whitespace as possible. 2017-12-22 09:23:22 -05:00
scripts refdebug: Create refstats.py script. 2018-10-15 15:35:35 -05:00
systemd contrib: Update systemd README.txt. 2018-07-18 17:14:44 -05:00
thirdparty remove extraneous svn:executable properties 2005-11-29 18:24:39 +00:00
unistimLang Add French translation for chan_unistim phones on-screen menus. 2012-07-16 07:34:12 +00:00
upstart Remove as much trailing whitespace as possible. 2017-12-22 09:23:22 -05:00
utils Remove as much trailing whitespace as possible. 2017-12-22 09:23:22 -05:00
asterisk-doxygen-header remove extraneous svn:executable properties 2005-11-29 18:24:39 +00:00
asterisk-ices.xml Remove as much trailing whitespace as possible. 2017-12-22 09:23:22 -05:00
dictionary.digium Add support for logging CDR recrods to a radius server (issue #6639, phsultan) 2006-05-20 22:30:05 +00:00
festival-1.4.1-diff remove extraneous svn:executable properties 2005-11-29 18:24:39 +00:00
festival-1.4.2.diff remove extraneous svn:executable properties 2005-11-29 18:24:39 +00:00
festival-1.4.3.diff remove extraneous svn:executable properties 2005-11-29 18:24:39 +00:00
festival-1.95.diff remove extraneous svn:executable properties 2005-11-29 18:24:39 +00:00
i18n.testsuite.conf Merged revisions 170671 via svnmerge from 2009-01-23 20:23:00 +00:00
Makefile refdebug: Create refstats.py script. 2018-10-15 15:35:35 -05:00
README.festival Remove as much trailing whitespace as possible. 2017-12-22 09:23:22 -05:00
valgrind-RedHat-8.0.supp remove extraneous svn:executable properties 2005-11-29 18:24:39 +00:00
valgrind.supp Line 24 missed in compatibility fix in revision 233577 2010-04-26 19:05:47 +00:00

app_festival is an application that allows one to send text-to-speech commands
to a background festival server, and to obtain the resulting waveform which
gets sent down to the respective channel. app_festival also employs a waveform
cache, so invariant text-to-speech strings ("Please press 1 for instructions")
do not need to be dynamically generated all the time.

You need :

1) festival, patched to produce 8khz waveforms on output. Patch for Festival
1.4.2 RELEASE are included. The patch adds a new command to festival
(asterisk_tts).

It is possible to run Festival without patches in the source-code. Just
add this to your /etc/festival.scm or /usr/share/festival/festival/scm:

    (define (tts_textasterisk string mode)
    "(tts_textasterisk STRING MODE)
    Apply tts to STRING. This function is specifically designed for
    use in server mode so a single function call may synthesize the string.
    This function name may be added to the server safe functions."
    (let ((wholeutt (utt.synth (eval (list 'Utterance 'Text string)))))
    (utt.wave.resample wholeutt 8000)
    (utt.wave.rescale wholeutt 5)
    (utt.send.wave.client wholeutt)))

[See the comment with subject "Using Debian
 festival >= 1.4.3-15 (no recompiling needed!)" on
 http://www.voip-info.org/wiki-Asterisk+festival+installation for the
 original mentioning of it]

2) You may wish to obtain and install the asterisk-perl
module by James Golovich <james@gnuinter.net>, from
either CPAN, or his site: http://asterisk.gnuinter.net,
as this contains a good example of how variable text
can be tts'd via asterisk, namely the examples/tts-*.agi
files there. It has been noted that the current expression
evaluation capabilities of asterisk are not best suited
for the generation and manipulation of text. AGI scripting
can be ideal for these sorts of needs. For simpler usage,
fixed, pre-recorded messages may be more amenable for your
purposes.

3) Before running asterisk, you have to run festival-server with a command
like :

/usr/local/festival/bin/festival --server > /dev/null 2>&1 &