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A feature of chan_sip that service providers relied upon was the ability to identify by the Authorization username. This is most often used when customers have a PBX that needs to register rather than identify by IP address. From my own experiance, this is pretty common with small businesses who otherwise don't need a static IP. In this scenario, a register from the customer's PBX may succeed because From will usually contain the PBXs account id but an INVITE will contain the caller id. With nothing recognizable in From, the service provider's Asterisk can never match to an endpoint and the INVITE just stays unauthorized. The fixes: A new value "auth_username" has been added to endpoint/identify_by that will use the username and digest fields in the Authorization header instead of username and domain in the the From header to match an endpoint, or the To header to match an aor. This code as added to res_pjsip_endpoint_identifier_user rather than creating a new module. Although identify_by was always a comma-separated list, there was only 1 choice so order wasn't preserved. So to keep the order, a vector was added to the end of ast_sip_endpoint. This is only used by res_pjsip_registrar to find the aor. The res_pjsip_endpoint_identifier_* modules are called in globals/endpoint_identifier_order. Along the way, the logic in res_pjsip_registrar was corrected to match most-specific to least-specific as res_pjsip_endpoint_identifier_user does. The order is: username@domain username@domain_alias username Auth by username does present 1 problem however, the first INVITE won't have an Authorization header so the distributor, not finding a match on anything, sends a securty_alert. It still sends a 401 with a challenge so the next INVITE will have the Authorization header and presumably succeed. As a result though, that first security alert is actually a false alarm. To address this, a new feature has been added to pjsip_distributor that keeps track of unidentified requests and only sends the security alert if a configurable number of unidentified requests come from the same IP in a configurable amout of time. Those configuration options have been added to the global config object. This feature is only used when auth_username is enabled. Finally, default_realm was added to the globals object to replace the hard coded "asterisk" used when an endpoint is not yet identified. The testsuite tests all pass but new tests are forthcoming for this new feature. ASTERISK-25835 #close Reported-by: Ross Beer Change-Id: I30ba62d208e6f63439600916fcd1c08a365ed69d
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 &