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The stream topology (list of streams and order) is now stored with the configured PJSIP endpoints and used during the negotiation process. Media negotiation state information has been changed to be stored in a separate object. Two of these objects exist at any one time on a session. The active media state information is what was previously negotiated and the pending media state information is what the media state will become if negotiation succeeds. Streams and other state information is stored in this object using the index (or position) of each individual stream for easy lookup. The ability for a media type handler to specify a callback for writing has been added as well as the ability to add file descriptors with a callback which is invoked when data is available to be read on them. This allows media logic to live outside of the chan_pjsip module. Direct media has been changed so that only the first audio and video stream are directly connected. In the future once the RTP engine glue API has been updated to know about streams each individual stream can be directly connected as appropriate. Media negotiation itself will currently answer all the provided streams on an offer within configured limits and on an offer will use the topology created as a result of the disallow/allow codec lines. If a stream has been removed or declined we will now mark it as such within the resulting SDP. Applications can now also request that the stream topology change. If we are told to do so we will limit any provided formats to the ones configured on the endpoint and send a re-invite with the new topology. Two new configuration options have also been added to PJSIP endpoints: max_audio_streams: determines the maximum number of audio streams to offer/accept from an endpoint. Defaults to 1. max_video_streams: determines the maximum number of video streams to offer/accept from an endpoint. Defaults to 1. ASTERISK-27076 Change-Id: I8afd8dd2eb538806a39b887af0abd046266e14c7
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 &