asterisk

History

George Joseph 5013d8f5d3 res_pjsip: Symmetric transports A new transport parameter 'symmetric_transport' has been added. When a request from a dynamic contact comes in on a transport with this option set to 'yes', the transport name will be saved and used for subsequent outgoing requests like OPTIONS, NOTIFY and INVITE. It's saved as a contact uri parameter named 'x-ast-txp' and will display with the contact uri in CLI, AMI, and ARI output. On the outgoing request, if a transport wasn't explicitly set on the endpoint AND the request URI is not a hostname, the saved transport will be used and the 'x-ast-txp' parameter stripped from the outgoing packet. * config_transport was modified to accept and store the new parameter. * config_transport/transport_apply was updated to store the transport name in the pjsip_transport->info field using the pjsip_transport->pool on UDP transports. * A 'multihomed_on_rx_message' function was added to pjsip_message_ip_updater that, for incoming requests, retrieves the transport name from pjsip_transport->info and retrieves the transport. If transport->symmetric_transport is set, an 'x-ast-txp' uri parameter containing the transport name is added to the incoming Contact header. * An 'ast_sip_get_transport_name' function was added to res_pjsip. It takes an ast_sip_endpoint and a pjsip_sip_uri and returns a transport name if endpoint->transport is set or if there's an 'x-ast-txp' parameter on the uri and the uri host is an ipv4 or ipv6 address. Otherwise it returns NULL. * An 'ast_sip_dlg_set_transport' function was added to res_pjsip which takes an ast_sip_endpoint, a pjsip_dialog, and an optional pjsip_tpselector. It calls ast_sip_get_transport_name() and if a non-NULL is returned, sets the selector and sets the transport on the dialog. If a selector was passed in, it's updated. * res_pjsip/ast_sip_create_dialog_uac and ast_sip_create_dialog_uas were modified to call ast_sip_dlg_set_transport() instead of their original logic. * res_pjsip/create_out_of_dialog_request was modified to call ast_sip_get_transport_name() and pjsip_tx_data_set_transport() instead of its original logic. * Existing transport logic was removed from endpt_send_request since that can only be called after a create_out_of_dialog_request. * res_pjsip/ast_sip_create_rdata was converted to a wrapper around a new 'ast_sip_create_rdata_with_contact' function which allows a contact_uri to be specified in addition to the existing parameters. (See below) * res_pjsip_pubsub/internal_pjsip_evsub_send_request was eliminated since all it did was transport selection and that is now done in ast_sip_create_dialog_uac and ast_sip_create_dialog_uas. * 'contact_uri' was added to subscription_persistence. This was necessary because although the parsed rdata contact header has the x-ast-txp parameter added (if appropriate), subscription_persistence_update stores the raw packet which doesn't have it. subscription_persistence_recreate was then updated to call ast_sip_create_rdata_with_contact with the persisted contact_uri so the recreated subscription has the correct transport info to send the NOTIFYs. * res_pjsip_session/internal_pjsip_inv_send_msg was eliminated since all it did was transport selection and that is now done in ast_sip_create_dialog_uac. * pjsip_message_ip_updater/multihomed_on_tx_message was updated to remove all traces of the x-ast-txp parameter from the outgoing headers. NOTE: This change does NOT modify the behavior of permanent contacts specified on an aor. To do so would require that the permanent contact's contact uri be updated with the x-ast-txp parameter and the aor sorcery object updated. If we need to persue this, we need to think about cloning permanent contacts into the same store as the dynamic ones on an aor load so they can be updated without disturbing the originally configured value. You CAN add the x-ast-txp parameter to a permanent contact's uri but it would be much simpler to just set endpoint->transport. Change-Id: I4ee1f51473da32ca54b877cd158523efcef9655f		8 years ago
..
ast-db-manage	res_pjsip: Symmetric transports	8 years ago
docker	Add initial support to build Docker images	9 years ago
editors	contrib/editors: Fix vim syntax highlighting of comments in config files	10 years ago
init.d	Update init.d scripts to handle stderr; readd splash screen for remote consoles	12 years ago
scripts	debug_utilities: Add ast_logescalator	8 years ago
thirdparty	…
unistimLang	Add French translation for chan_unistim phones on-screen menus.	13 years ago
upstart	Change all refererences to 1.6.3 to be 1.8, since that will be the next feature release	16 years ago
utils	Fix typo's (retrieve, specified, address).	10 years ago
Makefile	debug_utilities: Install ast_logescalator to /var/lib/asterisk/scripts	8 years ago
README.festival	…
asterisk-doxygen-header	…
asterisk-ices.xml	…
dictionary.digium	…
festival-1.4.1-diff	…
festival-1.4.2.diff	…
festival-1.4.3.diff	…
festival-1.95.diff	…
i18n.testsuite.conf	Merged revisions 170671 via svnmerge from	16 years ago
valgrind-RedHat-8.0.supp	…
valgrind.supp	Line 24 missed in compatibility fix in revision 233577	15 years ago

README.festival

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 &