Sorry for the delay and thanks Chris, Please find attached logs: http://www.mediafire.com/?8thg9ey7kd26882 sip client ip 192.168.1.202 freeswitch ip 192.168.1.35 mrcp connector/speech server ip 192.168.1.32 i have attached 3 logs - freeswitch log file, wireshark on freeswitch server, wireshark on mrcp connector/speech server. my script simply plays back audio from wave [basic freeswitch function], then plays back audio from TTS [which cannot be heard] Thank you, Michel. ---------- Forwarded message ----------

From: Christopher Rienzo <cmrienzo@gmail.com> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Date: Wed, 9 Mar 2011 22:25:34 -0500 Subject: Re: [Freeswitch-users] ASR from Freeswitch to MS Speech Server [using MRCP Connector] - Audio Problem I still would like to see the logs for your call.

On Tue, Mar 8, 2011 at 7:18 AM, Michel Habib <michelhabib@gmail.com>wrote:

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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>

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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:

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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.