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Add rtp_port_start and rtp_port_end options to PJSIP endpoint configuration, allowing each endpoint to use a dedicated RTP port range instead of the global rtp.conf setting. This is useful for scenarios where different endpoints need isolated port ranges, such as firewall rules per trunk, multi-tenant systems, or network QoS policies tied to port ranges. The implementation adds ast_rtp_instance_new_with_port_range() to the RTP engine API, which sets the port range on the instance before the engine allocates the transport. The default RTP engine (res_rtp_asterisk) checks for per-instance overrides in rtp_allocate_transport() and falls back to the global range when none is set. Both options must be set together, with values >= 1024 and rtp_port_end > rtp_port_start. Setting both to 0 (the default) preserves existing behavior. Resolves: https://github.com/asterisk/asterisk-feature-requests/issues/71 UserNote: PJSIP endpoints now support rtp_port_start and rtp_port_end options to configure a dedicated RTP port range per endpoint, overriding the global rtp.conf setting. UpgradeNote: An alembic database migration has been added to add the rtp_port_start and rtp_port_end columns to the ps_endpoints table. Run "alembic upgrade head" to apply the schema change. DeveloperNote: New public API: ast_rtp_instance_new_with_port_range() creates an RTP instance with a per-instance port range. ast_rtp_instance_get_port_start() and ast_rtp_instance_get_port_end() allow RTP engines to query the override. Third-party RTP engines can use these getters to support per-instance port ranges.
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 &