I still would like to see the logs for your call. <br><br><br><div class="gmail_quote">On Tue, Mar 8, 2011 at 7:18 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">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.<br>
192.168.5.107 is the freeswitch server<br>
192.168.5.110 is the MRCP-Connector/Speech Server 2007 server<br>
I made too many iterations on the configuration below:<br><br><include><br> <profile name="mrcp-connector" version="2"><br> <param name="client-ip" value="192.168.5.107"/><br>
<param name="client-port" value="5090"/><br> <param name="server-ip" value="192.168.5.110"/><br> <param name="server-port" value="5070"/><br>
<!--param name="force-destination" value="1"/--><br> <param name="sip-transport" value="udp"/><br> <!--param name="ua-name" value="FreeSWITCH"/--><br>
<!--param name="sdp-origin" value="FreeSWITCH"/--><br> <param name="rtp-ext-ip" value="192.168.5.107"/><br> <param name="rtp-ip" value="192.168.5.107"/><br>
<param name="rtp-port-min" value="4000"/><br> <param name="rtp-port-max" value="5000"/><br> <!-- enable/disable rtcp support --><br> <param name="rtcp" value="1"/><br>
<!-- rtcp bye policies (rtcp must be enabled first)<br> 0 - disable rtcp bye<br> 1 - send rtcp bye at the end of session<br> 2 - send rtcp bye also at the end of each talkspurt (input)<br>
--><br> <param name="rtcp-bye" value="2"/><br> <!-- rtcp transmission interval in msec (set 0 to disable) --><br> <param name="rtcp-tx-interval" value="5000"/><br>
<!-- period (timeout) to check for new rtcp messages in msec (set 0 to disable) --><br> <param name="rtcp-rx-resolution" value="1000"/><br><br> <!--param name="playout-delay" value="50"/--><br>
<!--param name="max-playout-delay" value="200"/--><br> <!--param name="ptime" value="20"/--><br> <param name="codecs" value="PCMU PCMA L16/96/8000"/><br>
<br> <!-- Add any default MRCP params for SPEAK requests here --><br> <synthparams><br> </synthparams><br><br> <!-- Add any default MRCP params for RECOGNIZE requests here --><br> <recogparams><br>
<!--param name="start-input-timers" value="false"/--><br> </recogparams><br> </profile><br></include><div class="im"><br><br><br>---------- Forwarded message ----------<br>
<div class="gmail_quote">
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">From: Christopher Rienzo <<a href="mailto:cmrienzo@gmail.com" target="_blank">cmrienzo@gmail.com</a>><br>
To: FreeSWITCH Users Help <<a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a>><br>Date: Mon, 7 Mar 2011 09:32:51 -0500<br>Subject: Re: [Freeswitch-users] ASR from Freeswitch to MS Speech Server [using MRCP Connector] - Audio Problem<br>
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" target="_blank">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>
<br></blockquote></div></blockquote></div><br></div></div>
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