Do you get audio between FS and your SIP client when not using ASR/TTS?

Show me the MRCP profile configuration and your FreeSWITCH logs during the call.



On Mon, Mar 7, 2011 at 4:55 AM, Michel Habib <michelhabib@gmail.com> wrote:
Hello All,
I have MS OCS Speech Server 2007 [working correctly, as i can make SIP calls and use its ASR and TTS Services successfully]
I am also using MRCP Connector from AumTech - which allows me to use ASR and TTS Services through an MRCP Client .
Now, i am using Freeswitch mod unimrcp to use ASR and TTS.

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.
However, Freeswitch is not sending back the Audio RTP to the SIP client.

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].

I am suspecting something is wrong with the RTP Configuration - can you help me?

Let me now if you need any specific logs/scripts/configuration?

Thank you,
Michel.


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