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