<div>Hi Michael/Chris</div><div> </div><div>Did you find out what the problem was? We're seeing something similar.</div><div> </div><div>Thanks in advance<br></div><div>David<br></div><div class="gmail_quote">On Mon Mar 14, Michel Habib <michelhabib at <a href="http://gmail.com/" target="_blank">gmail.com</a>>wrote:<blockquote style="margin:0px 0px 0px 0.8ex;padding-left:1ex;border-left-color:rgb(204,204,204);border-left-width:1px;border-left-style:solid" class="gmail_quote">
<p>Sorry for the delay and thanks Chris,<br>Please find attached logs: <a href="http://www.mediafire.com/?8thg9ey7kd26882" target="_blank">http://www.mediafire.com/?8thg9ey7kd26882</a><br>sip client ip 192.168.1.202<br>freeswitch ip 192.168.1.35<br>
mrcp connector/speech server ip 192.168.1.32<br>i have attached 3 logs - freeswitch log file, wireshark on freeswitch<br>server, wireshark on mrcp connector/speech server.<br>my script simply plays back audio from wave [basic freeswitch function],<br>
then plays back audio from TTS [which cannot be heard]</p><p>Thank you,<br>Michel.</p><p>---------- Forwarded message ----------<br>> From: Christopher Rienzo <cmrienzo at <a href="http://gmail.com" target="_blank">gmail.com</a>><br>
> To: FreeSWITCH Users Help <freeswitch-users at <a href="http://lists.freeswitch.org" target="_blank">lists.freeswitch.org</a>><br>> Date: Wed, 9 Mar 2011 22:25:34 -0500<br>> Subject: Re: [Freeswitch-users] ASR from Freeswitch to MS Speech Server<br>
> [using MRCP Connector] - Audio Problem<br>> I still would like to see the logs for your call.<br>><br>><br>> On Tue, Mar 8, 2011 at 7:18 AM, Michel Habib <michelhabib at <a href="http://gmail.com" target="_blank">gmail.com</a>>wrote:<br>
><br>>> Yes, I get the Audio from FS in regular calls - I already disabled all<br>>> possible firewalls - all 3 machines [softphone, freeswitch, Speech Server<br>>> (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<br>
>> 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><br>>><br>>><br>>><br>>> ---------- Forwarded message ----------<br>>><br>>>> From: Christopher Rienzo <cmrienzo at <a href="http://gmail.com" target="_blank">gmail.com</a>><br>
>>> To: FreeSWITCH Users Help <freeswitch-users at <a href="http://lists.freeswitch.org" target="_blank">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<br>
>>> [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<br>
>>> the call.<br>>>><br>>>><br>>>><br>>>> On Mon, Mar 7, 2011 at 4:55 AM, Michel Habib <michelhabib at <a href="http://gmail.com" target="_blank">gmail.com</a>>wrote:<br>
>>><br>
>>>> Hello All,<br>>>>> I have MS OCS Speech Server 2007 [working correctly, as i can make SIP<br>>>>> calls and use its ASR and TTS Services successfully]<br>>>>> I am also using MRCP Connector from AumTech - which allows me to use ASR<br>
>>>> 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<br>
>>>> voice is transferred succesfully from Speech Server [through MRCP Connector]<br>>>>> 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,<br>>>>> but the audio RTP sent to speech server [through MRCP Connector] is silent<br>>>>> [empty].<br>
>>>><br>>>>> I am suspecting something is wrong with the RTP Configuration - can you<br>>>>> help me?<br>>>>><br>>>>> Let me now if you need any specific logs/scripts/configuration?<br>
>>>><br>>>>> Thank you,<br>>>>> Michel.<br>>>>><br>>>><br></p>
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