Do you get audio between FS and your SIP client when not using ASR/TTS?<br><br>Show me the MRCP profile configuration and your FreeSWITCH logs during the call.<br><br><br><br><div class="gmail_quote">On Mon, Mar 7, 2011 at 4:55 AM, Michel Habib <span dir="ltr"><<a href="mailto:michelhabib@gmail.com">michelhabib@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;"><div dir="ltr">Hello All,<br>I have MS OCS Speech Server 2007 [working correctly, as i can make SIP calls and use its ASR and TTS Services successfully]<br>
I am also using MRCP Connector from AumTech - which allows me to use ASR and TTS Services through an MRCP Client .<br>
Now, i am using Freeswitch mod unimrcp to use ASR and TTS.<br><br>for TTS, I can successfully make the call, the Audio RTP of the TTS voice is transferred succesfully from Speech Server [through MRCP Connector] back to the Freeswitch Server.<br>
However, Freeswitch is not sending back the Audio RTP to the SIP client.<br><br>for ASR, I can successfully define the grammar and start recognition, but the audio RTP sent to speech server [through MRCP Connector] is silent [empty].<br>
<br>I am suspecting something is wrong with the RTP Configuration - can you help me?<br><br>Let me now if you need any specific logs/scripts/configuration?<br><br>Thank you,<br>Michel.<br><br></div>
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