Files
asterisk/contrib
Michael Kuron fee9012fe1 res_pjsip_aoc: New module for sending advice-of-charge with chan_pjsip
chan_sip supported sending AOC-D and AOC-E information in SIP INFO
messages in an "AOC" header in a format that was originally defined by
Snom. In the meantime, ETSI TS 124 647 introduced an XML-based AOC
format that is supported by devices from multiple vendors, including
Snom phones with firmware >= 8.4.2 (released in 2010).

This commit adds a new res_pjsip_aoc module that inserts AOC information
into outgoing messages or sends SIP INFO messages as described below.
It also fixes a small issue in res_pjsip_session which didn't always
call session supplements on outgoing_response.

* AOC-S in the 180/183/200 responses to an INVITE request
* AOC-S in SIP INFO (if a 200 response has already been sent or if the
  INVITE was sent by Asterisk)
* AOC-D in SIP INFO
* AOC-D in the 200 response to a BYE request (if the client hangs up)
* AOC-D in a BYE request (if Asterisk hangs up)
* AOC-E in the 200 response to a BYE request (if the client hangs up)
* AOC-E in a BYE request (if Asterisk hangs up)

The specification defines one more, AOC-S in an INVITE request, which
is not implemented here because it is not currently possible in
Asterisk to have AOC data ready at this point in call setup. Once
specifying AOC-S via the dialplan or passing it through from another
SIP channel's INVITE is possible, that might be added.

The SIP INFO requests are sent out immediately when the AOC indication
is received. The others are inserted into an appropriate outgoing
message whenever that is ready to be sent. In the latter case, the XML
is stored in a channel variable at the time the AOC indication is
received. Depending on where the AOC indications are coming from (e.g.
PRI or AMI), it may not always be possible to guarantee that the AOC-E
is available in time for the BYE.

Successfully tested AOC-D and both variants of AOC-E with a Snom D735
running firmware 10.1.127.10. It does not appear to properly support
AOC-S however, so that could only be tested by inspecting SIP traces.

ASTERISK-21502 #close
Reported-by: Matt Jordan <mjordan@digium.com>

Change-Id: Iebb7ad0d5f88526bc6629d3a1f9f11665434d333
2022-12-09 07:57:21 -06:00
..
2021-11-16 06:02:11 -06:00
2021-11-16 06:02:11 -06:00
2018-10-15 15:35:35 -05:00

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 &