<p>Try removing nat=yes<br>
</p>
<div class="gmail_quote">On Aug 28, 2013 1:19 PM, "王鹏" <<a href="mailto:voipwangpeng@126.com">voipwangpeng@126.com</a>> wrote:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div style="line-height:1.7;font-size:14px;font-family:arial"><br>HI,all<div><p style="margin:0px;padding:0px;font-family:Arial,FreeSans,Helvetica,sans-serif;font-size:13px;line-height:17px">my asterisk11 have a domain name with dynamic ipaddress.<br>
i want to use asterisk and linphone make p2p call, i open icesupport in sip.conf and rtp.conf, can call through but no voice.</p><p style="margin:16px 0px 0px;padding:0px;font-family:Arial,FreeSans,Helvetica,sans-serif;font-size:13px;line-height:17px">
here <a href="https://wiki.asterisk.org/wiki/display/~jcolp/ICE,+STUN,+and+TURN+Support" title="Follow link" style="color:rgb(119,51,0);text-decoration:initial" target="_blank">https://wiki.asterisk.org/wiki/display/~jcolp/ICE,+STUN,+and+TURN+Support</a><br>
i see ICE support is only used for communication between a remote endpoint and Asterisk.</p><p style="margin:16px 0px 0px;padding:0px;font-family:Arial,FreeSans,Helvetica,s ans-serif;font-size:13px;line-height:17px">so i close icesupport in sip.conf and rtp.conf, but invite from caller have ice paramter,but asterisk modify invite message, delete ice paramter, then pass to callee, so phenomenon is also can call through but no voice.</p>
<p style="margin:16px 0px 0px;padding:0px;font-family:Arial,FreeSans,Helvetica,sans-serif;font-size:13px;line-height:17px">how to make asterisk not modify sdp of invite, or have other method to use asterisk and linphone make p2p call.</p>
<br>thanks!!</div><div><br></div><div><br></div></div><br><br><span title="neteasefooter"><span></span></span><br>--<br>
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