<html><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; "><span class="Apple-style-span" style="font-family: -webkit-sans-serif; font-size: 13px; line-height: 19px; ">"With FreeSWITCH not having any supported ASR at the time of writing (with the exception of PocketSphinx), we needed something to allow us to connect it to an MRCP server to test SoftIVR's ASR functionality. After a few false starts, we implemented a simple MRCP connector using the outbound socket interface, unicast and a bit of Perl."</span><div><font class="Apple-style-span" face="-webkit-sans-serif" size="3"><span class="Apple-style-span" style="font-size: 13px; line-height: 19px;"><br></span></font></div><div><font class="Apple-style-span" face="-webkit-sans-serif" size="3"><span class="Apple-style-span" style="font-size: 13px; line-height: 19px;">Was mod_openmrcp not enough :) We really need someone to fund the writing of mod_unimrcp.</span></font></div><div><font class="Apple-style-span" face="-webkit-sans-serif" size="3"><span class="Apple-style-span" style="font-size: 13px; line-height: 19px;"><br></span></font></div><div><font class="Apple-style-span" face="-webkit-sans-serif" size="3"><span class="Apple-style-span" style="font-size: 13px; line-height: 19px;">/b</span></font></div><div><font class="Apple-style-span" face="-webkit-sans-serif" size="3"><span class="Apple-style-span" style="font-size: 13px; line-height: 19px;"><br></span></font><div><div>On Jan 12, 2009, at 5:49 AM, David Knell wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite"><div>Hi all -<br><br>In case anyone's interested, I've documented how we interfaced FS with <br>Lumenvox via MRCP using FS' event socket and unicast interfaces and a <br>bit of Perl here: <br><a href="http://www.softivr.com/wiki/index.php/FreeSWITCH_MRCP_in_Perl">http://www.softivr.com/wiki/index.php/FreeSWITCH_MRCP_in_Perl</a><br><br>Three surprises: that it worked at all, that it works quite well and <br>that it was really quite easy to do.<br><br>One thing I'm looking for: has anyone written a module which attaches a <br>bug to an audio stream and forwards the audio as RTP to a specified <br>IP/port to just allow audio to be tapped off a call and sent somewhere <br>else to be listened to?<br><br>Cheers --<br><br>Dave<br><br>-- <br>David Knell, Director, 3C Limited<br>T: 020 8114 8901 F: 020 3002 7257 M: 001 415 630 3031<br>http://www.3c.co.uk <br><br><br>_______________________________________________<br>Freeswitch-users mailing list<br>Freeswitch-users@lists.freeswitch.org<br>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users<br>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<br>http://www.freeswitch.org<br></div></blockquote></div><br></div></body></html>