<html><head><meta http-equiv="Content-Type" content="text/html charset=utf-8"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" class="">Hey guys, <div class=""><br class=""></div><div class="">have issues with reinvite, no matter what endpoint is calling asterisk always tries switch simple_bridge to native_rtp<div class=""><br class=""></div><div class=""><div class="" style="margin: 0px; font-size: 11px; font-family: Menlo;"> Bridge 0422bfa0-9d22-4bba-9108-a3f14d7d1cab: switching from simple_bridge technology to native_rtp</div></div><div class=""><br class=""></div><div class=""><font face="Menlo" class=""><span class="" style="font-size: 11px;">in endpoints table “direct_media” sets to “no” on all endpoints but it doesn’t help.</span></font></div><div class=""><font face="Menlo" class=""><span class="" style="font-size: 11px;"><br class=""></span></font></div><div class=""><font face="Menlo" class=""><span class="" style="font-size: 11px;">if native_rtp not work for some reason I have oneway audio. how can I fix this? if I add mix_monitor it works, but it’s not a right way to fix this issues.</span></font></div><div class=""><font face="Menlo" class=""><span class="" style="font-size: 11px;"><br class=""></span></font></div><div class=""><div style="margin: 0px; font-size: 11px; font-family: Menlo;" class="">Asterisk 13.2.0</div></div></div></body></html>