Yes, I get the Audio from FS in regular calls - I already disabled all possible firewalls - all 3 machines [softphone, freeswitch, Speech Server (and mrcp connector) ] are on a switch.
192.168.5.107 is the freeswitch server
192.168.5.110 is the MRCP-Connector/Speech Server 2007 server
I made too many iterations on the configuration below:

<include>
  <profile name="mrcp-connector" version="2">
    <param name="client-ip" value="192.168.5.107"/>
    <param name="client-port" value="5090"/>
    <param name="server-ip" value="192.168.5.110"/>
    <param name="server-port" value="5070"/>
    <!--param name="force-destination" value="1"/-->
    <param name="sip-transport" value="udp"/>
    <!--param name="ua-name" value="FreeSWITCH"/-->
    <!--param name="sdp-origin" value="FreeSWITCH"/-->
    <param name="rtp-ext-ip" value="192.168.5.107"/>
    <param name="rtp-ip" value="192.168.5.107"/>
    <param name="rtp-port-min" value="4000"/>
    <param name="rtp-port-max" value="5000"/>
    <!-- enable/disable rtcp support -->
    <param name="rtcp" value="1"/>
    <!-- rtcp bye policies (rtcp must be enabled first)
             0 - disable rtcp bye
             1 - send rtcp bye at the end of session
             2 - send rtcp bye also at the end of each talkspurt (input)
    -->
    <param name="rtcp-bye" value="2"/>
    <!-- rtcp transmission interval in msec (set 0 to disable) -->
    <param name="rtcp-tx-interval" value="5000"/>
    <!-- period (timeout) to check for new rtcp messages in msec (set 0 to disable) -->
    <param name="rtcp-rx-resolution" value="1000"/>

    <!--param name="playout-delay" value="50"/-->
    <!--param name="max-playout-delay" value="200"/-->
    <!--param name="ptime" value="20"/-->
    <param name="codecs" value="PCMU PCMA L16/96/8000"/>

    <!-- Add any default MRCP params for SPEAK requests here -->
    <synthparams>
    </synthparams>

    <!-- Add any default MRCP params for RECOGNIZE requests here -->
    <recogparams>
      <!--param name="start-input-timers" value="false"/-->
    </recogparams>
  </profile>
</include>


---------- Forwarded message ----------
From: Christopher Rienzo <cmrienzo@gmail.com>
To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org>
Date: Mon, 7 Mar 2011 09:32:51 -0500
Subject: Re: [Freeswitch-users] ASR from Freeswitch to MS Speech Server [using MRCP Connector] - Audio Problem
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.