<html><head><meta http-equiv="Content-Type" content="text/html charset=us-ascii"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" class=""><div style="margin: 0px; font-size: 11px; font-family: Menlo;" class="">NAT endpoint calling local endpount - switching to native_rtp then no audio, both of them have direct_media=no, Verbose log:</div><div style="margin: 0px; font-size: 11px; font-family: Menlo;" class=""><br class=""></div><div style="margin: 0px; font-size: 11px; font-family: Menlo;" class=""> -- Executing [99@dialmap:1] <span style="font-variant-ligatures: no-common-ligatures; color: #34bbc7" class=""><b class="">AGI</b></span>("<span style="font-variant-ligatures: no-common-ligatures; color: #d53bd3" class=""><b class="">PJSIP/304-00000022</b></span>", "<span style="font-variant-ligatures: no-common-ligatures; color: #d53bd3" class=""><b class="">/pbx/agi.php</b></span>") in new stack</div><div style="margin: 0px; font-size: 11px; font-family: Menlo;" class=""> -- Launched AGI Script /pbx/agi.php</div><div style="margin: 0px; font-size: 11px; font-family: Menlo;" class=""> -- AGI Script Executing Application: (Dial) Options: (PJSIP/99/sip:<a href="mailto:99@192.168.1.73" class="">99@192.168.1.73</a>:5060,20)</div><div style="margin: 0px; font-size: 11px; font-family: Menlo;" class=""> -- Called PJSIP/99/sip:<a href="mailto:99@192.168.1.73" class="">99@192.168.1.73</a>:5060</div><div style="margin: 0px; font-size: 11px; font-family: Menlo;" class=""> -- PJSIP/99-00000023 is ringing</div><div style="margin: 0px; font-size: 11px; font-family: Menlo;" class=""> -- PJSIP/99-00000023 answered PJSIP/304-00000022</div><div style="margin: 0px; font-size: 11px; font-family: Menlo;" class=""> -- Channel PJSIP/304-00000022 joined 'simple_bridge' basic-bridge <da8840bc-9b71-4ca6-b1d8-9565bf8e5e28></div><div style="margin: 0px; font-size: 11px; font-family: Menlo;" class=""> -- Channel PJSIP/99-00000023 joined 'simple_bridge' basic-bridge <da8840bc-9b71-4ca6-b1d8-9565bf8e5e28></div><div style="margin: 0px; font-size: 11px; font-family: Menlo;" class=""> > Bridge da8840bc-9b71-4ca6-b1d8-9565bf8e5e28: switching from simple_bridge technology to native_rtp</div><div style="margin: 0px; font-size: 11px; font-family: Menlo;" class=""> > Locally RTP bridged 'PJSIP/99-00000023' and 'PJSIP/304-00000022' in stack</div><div style="margin: 0px; font-size: 11px; font-family: Menlo;" class=""> > Locally RTP bridged 'PJSIP/99-00000023' and 'PJSIP/304-00000022' in stack</div><div style="margin: 0px; font-size: 11px; font-family: Menlo;" class=""> > 0x7f4b50145420 -- Probation passed - setting RTP source address to 194.204.157.200:8972</div><div style="margin: 0px; font-size: 11px; font-family: Menlo;" class=""> > 0x7f4b5014f140 -- Probation passed - setting RTP source address to 192.168.1.73:5004</div><div style="margin: 0px; font-size: 11px; font-family: Menlo;" class=""> -- Channel PJSIP/304-00000022 left 'native_rtp' basic-bridge <da8840bc-9b71-4ca6-b1d8-9565bf8e5e28></div><div style="margin: 0px; font-size: 11px; font-family: Menlo;" class=""> -- Channel PJSIP/99-00000023 left 'native_rtp' basic-bridge <da8840bc-9b71-4ca6-b1d8-9565bf8e5e28></div><div style="margin: 0px; font-size: 11px; font-family: Menlo;" class=""> -- <PJSIP/304-00000022>AGI Script /pbx/agi.php completed, returning 4</div><div style="margin: 0px; font-size: 11px; font-family: Menlo;" class=""><br class=""></div><div style="margin: 0px; font-size: 11px; font-family: Menlo;" class=""><br class=""></div><div><blockquote type="cite" class=""><div class="">On 18 Mar 2015, at 18:26, Matthew Jordan <<a href="mailto:mjordan@digium.com" class="">mjordan@digium.com</a>> wrote:</div><br class="Apple-interchange-newline"><div class="">On Wed, Mar 18, 2015 at 9:53 AM, Nick Awesome <<a href="mailto:jleed@me.com" class="">jleed@me.com</a>> wrote:<br class=""><blockquote type="cite" class="">Well, it breaks audio for all NAT endpoints, how can I fix this?<br class=""><br class=""></blockquote><br class="">Local (packet to packet) bridging should not do that. Remote (direct<br class="">media) can do that.<br class=""><br class="">Can you confirm - by looking at a verbose level 4 log - how Asterisk<br class="">is bridging the two channels?<br class=""><br class="">-- <br class="">Matthew Jordan<br class="">Digium, Inc. | Director of Technology<br class="">445 Jan Davis Drive NW - Huntsville, AL 35806 - USA<br class="">Check us out at: <a href="http://digium.com" class="">http://digium.com</a> & <a href="http://asterisk.org" class="">http://asterisk.org</a><br class=""><br class="">-- <br class="">_____________________________________________________________________<br class="">-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" class="">http://www.api-digital.com</a> --<br class="">New to Asterisk? Join us for a live introductory webinar every Thurs:<br class=""> <a href="http://www.asterisk.org/hello" class="">http://www.asterisk.org/hello</a><br class=""><br class="">asterisk-users mailing list<br class="">To UNSUBSCRIBE or update options visit:<br class=""> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" class="">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br class=""></div></blockquote></div><br class=""></body></html>