<html xmlns:v="urn:schemas-microsoft-com:vml" xmlns:o="urn:schemas-microsoft-com:office:office" xmlns:w="urn:schemas-microsoft-com:office:word" xmlns:x="urn:schemas-microsoft-com:office:excel" xmlns:m="http://schemas.microsoft.com/office/2004/12/omml" xmlns="http://www.w3.org/TR/REC-html40"><head><META HTTP-EQUIV="Content-Type" CONTENT="text/html; charset=us-ascii"><meta name=Generator content="Microsoft Word 14 (filtered medium)"><style><!--
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</o:shapelayout></xml><![endif]--></head><body bgcolor=white lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Do the call both ways again and check(post) the CLI output.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><div><div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'><p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif";color:windowtext'>From:</span></b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif";color:windowtext'> asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] <b>On Behalf Of </b>Bart Coninckx<br><b>Sent:</b> Tuesday, September 25, 2012 4:23 PM<br><b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br><b>Subject:</b> [asterisk-users] no audio while call forwarding, yes audio with followme<o:p></o:p></span></p></div></div><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal style='margin-bottom:12.0pt'>Hi all,<br><br>the subject says it all.<br>Technical details: <br>- Asterisk 1.8.7.1 <br>- Behind NAT <br>- Using external SIP provider<br><br>The call forwarding is tested both with this functionality on the phone and with configuration in the dialplan. In the latter case a database variable is set to the external number, if set a Dial command calls this number. So really nothing fancy (actually I followed the example on <a href="http://www.voip-info.org/wiki/view/Asterisk+call+forwarding">http://www.voip-info.org/wiki/view/Asterisk+call+forwarding</a> ).<br><br>sip.conf has nat=yes, externip= ... and I tried every setting of directmedia in the providers configuration part. <br><br>Followme works flawlessly, so I'm really wondering if this is a NAT issue. <br><br><br>Can anyone point me into a certain direction?<br><br><br>Thx!!!!<br><br><br>BC<o:p></o:p></p></div></body></html>