<html><head><meta http-equiv="Content-Type" content="text/html; charset=utf-8"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; line-break: after-white-space;" class="">Hello Freeswitch Pros,<div class=""><br class=""></div><div class="">I just wanted to see if by chance anyone had any suggestions on the below info and issue. Thank you so much for your feedback and direction as it is greatly appreciated!</div><div class=""><br class=""></div><div class="">Thank you,</div><div class=""><br class=""></div><div class="">-Spence<br class=""><div><br class=""><blockquote type="cite" class=""><div class="">On Jan 19, 2020, at 3:54 PM, Spencer Angerbauer <<a href="mailto:spencer.angerbauer@gmail.com" class="">spencer.angerbauer@gmail.com</a>> wrote:</div><br class="Apple-interchange-newline"><div class=""><meta http-equiv="Content-Type" content="text/html; charset=utf-8" class=""><div style="word-wrap: break-word; -webkit-nbsp-mode: space; line-break: after-white-space;" class=""><div class="">I am trying to get all conferences to say a phrase before entering the pin #1 prior to entering (as we are connecting via outbound dialer from api). I have installed TTS to do a “say” function, and have updated the <span class="" style="caret-color: rgb(23, 43, 77); background-color: rgb(255, 255, 255);">conf/autoload_configs/c</span>onference.conf.xml by making the “pin” required and to require pin as “1”, but every time a caller connects, it just transfers them directly into the conference without announcing that they need to press “1” to enter, nor does it require a “pin” to connect to the conference.<br class=""><font color="#5856d6" class=""><span style="caret-color: rgb(88, 86, 214);" class=""><br class=""></span></font>Is there somewhere else I need to update the dialing plan to require all my conference calls require a “1” when using the web api outbound dialer to connect everyone? (I am using http://<<servername>>:8080/<font face="Helvetica Neue" class="">webapi/</font><a href="http://api.joonto.com:8080/webapi/originate?sofia/gateway/signalwire/18015737111%20&conference(1234)" style="font-family: "Helvetica Neue";" class="">originate?sofia/gateway/signalwire/<<phone number>>%20&conference(<<conference name>>)</a><span style="font-family: "Helvetica Neue";" class="">to connect multiple calls together).</span></div></div></div></blockquote><blockquote type="cite" class=""><div class=""><div style="word-wrap: break-word; -webkit-nbsp-mode: space; line-break: after-white-space;" class=""><div class=""><font color="#5856d6" class=""><span style="caret-color: rgb(88, 86, 214);" class=""><br class=""></span></font>Also, is there another location to add speak to text for the beginning of every conference from webapi or does it all need to be handled by the dial plan?<br class=""><font color="#5856d6" class=""><span style="caret-color: rgb(88, 86, 214);" class=""><br class=""></span></font>Thank you for your help on both these related issues.<br class=""><font color="#5856d6" class=""><span style="caret-color: rgb(88, 86, 214);" class=""><br class=""></span></font><div style="word-wrap: break-word; -webkit-nbsp-mode: space; line-break: after-white-space;" class=""><div class="">-Spence</div></div></div><br class=""></div></div></blockquote></div><br class=""></div></body></html>