Where i can get help with freeswitch?
Hello from St.Petersburg! My name is Patrick. I try to realise IVR with ASR & TTS. Platform win32. Soft: Freeswith, Unimrcp mod (client), and some local product "Voicenavigator" (mrcp server, ASR, TTS). I have 2 problems: 1. How to realise "barge in" for playback and TTS? Does freeswitch allow that? I need to start playback, or TTS and break it when some speech is detected (by ASR)... 2. When I call from Asterisk to Freeswitch and bridge my call back to Asterisk (to another number), there only noise in first asterisk abonent's phone... I would be grateful for any help ;-) -- С уважением, Равальдини Патрик Стефанович Инженер по тестированию ООО «Центр речевых технологий» Тел.: (812) 325-8848, доб. 6225 Факс: (812) 327-9297 E-mail: patrick@speechpro.com http://www.speechpro.ru
Barge-in will work out of the box for digits... to make it work for ASR is a bit more complicated. I don't know what method you are using to do TTS, but it this is the general idea: 1. set up a handler to deal with input callbacks 2. on DTMF or start of speech, return "break" to cause barge-in. My help can be more specific if you tell me more about what method (dialplan, Lua, javascript, etc) you are using to execute TTS and ASR. Can't help you on the noise issue... someone else needs to chime in. On Fri, Apr 23, 2010 at 5:41 AM, patrick <patrick@speechpro.com> wrote:
Hello from St.Petersburg! My name is Patrick. I try to realise IVR with ASR & TTS. Platform win32. Soft: Freeswith, Unimrcp mod (client), and some local product "Voicenavigator" (mrcp server, ASR, TTS).
I have 2 problems:
1. How to realise "barge in" for playback and TTS? Does freeswitch allow that? I need to start playback, or TTS and break it when some speech is detected (by ASR)...
2. When I call from Asterisk to Freeswitch and bridge my call back to Asterisk (to another number), there only noise in first asterisk abonent's phone...
I would be grateful for any help ;-)
-- С уважением,
Равальдини Патрик Стефанович Инженер по тестированию ООО <<Центр речевых технологий>> Тел.: (812) 325-8848, доб. 6225 Факс: (812) 327-9297 E-mail: patrick@speechpro.com http://www.speechpro.ru
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I use dialplan and this extensions: <extension name="my_ivr_01"> <condition field="destination_number" expression="^777$"> <action application="answer"/> <action application="set" data="tts_engine=unimrcp:vn-mrcp-v1"/> <action application="set" data="tts_voice=Anna8000"/> <action application="speak" data="Here is some text for synthesis"/> <action application="detect_speech" data="unimrcp http://192.168.22.116:8080/vxml/tstgrammar.xml vn-mrcp-v1"/> </condition> </extension> <extension name="my_ivr_02"> <condition field="destination_number" expression="^778$"> <action application="answer"/> <action application="set" data="playback_terminators=#"/> <action application="playback" data="ponce-preludio-in-e-major.wav"/> <action application="detect_speech" data="unimrcp http://192.168.22.116:8080/vxml/tstgrammar.xml vn-mrcp-v1"/> </condition> </extension> And I need to barge-in and start "detect_speech" in both extensions, when synthesis or playback is going on. P.S. Thank you for answer! Christopher Rienzo пишет:
Barge-in will work out of the box for digits... to make it work for ASR is a bit more complicated.
I don't know what method you are using to do TTS, but it this is the general idea:
1. set up a handler to deal with input callbacks 2. on DTMF or start of speech, return "break" to cause barge-in.
My help can be more specific if you tell me more about what method (dialplan, Lua, javascript, etc) you are using to execute TTS and ASR.
Can't help you on the noise issue... someone else needs to chime in.
On Fri, Apr 23, 2010 at 5:41 AM, patrick <patrick@speechpro.com <mailto:patrick@speechpro.com>> wrote:
Hello from St.Petersburg! My name is Patrick. I try to realise IVR with ASR & TTS. Platform win32. Soft: Freeswith, Unimrcp mod (client), and some local product "Voicenavigator" (mrcp server, ASR, TTS).
I have 2 problems:
1. How to realise "barge in" for playback and TTS? Does freeswitch allow that? I need to start playback, or TTS and break it when some speech is detected (by ASR)...
2. When I call from Asterisk to Freeswitch and bridge my call back to Asterisk (to another number), there only noise in first asterisk abonent's phone...
I would be grateful for any help ;-)
-- С уважением,
Равальдини Патрик Стефанович Инженер по тестированию ООО «Центр речевых технологий» Тел.: (812) 325-8848, доб. 6225 Факс: (812) 327-9297 E-mail: patrick@speechpro.com <mailto:patrick@speechpro.com> http://www.speechpro.ru
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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-- С уважением, Равальдини Патрик Стефанович Инженер по тестированию ООО «Центр речевых технологий» Тел.: (812) 325-8848, доб. 6225 Факс: (812) 327-9297 E-mail: patrick@speechpro.com http://www.speechpro.ru
participants (2)
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Christopher Rienzo
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patrick