<div>Hi,</div>
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<div>I have already set canreinvite=no in the sip.conf and also used the NAT=yes. But the funny thing that was in one case the user call and it wasn't working (one way audio as described) using an online dialer and then tried again using X-lite it was working. Then hanged up and tried X-lite again, it was not working. The second call was only a few seconds apart. Moving back to the online dialer, it wasn't working either. So it is just very strange to me how this happened and i was think maybe it was the RTP negiotation. Do you have any ideas?
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<div>Regards,</div>
<div>Kengie<br><br><br>>Few people, or no one, will take the time to see all the debug.<br><br>>The key here is that the RTP port and IP negotiated in the SDP message<br>>sent by asterisk to each party, should be "visible" for the party. A
<br>>common error is Asterisk sending in SDP a private IP address to a<br>>public UA, so the public UA will attempt to send RTP audio to a<br>>private IP, never reaching the Asterisk Server. Check <a onclick="return top.js.OpenExtLink(window,event,this)" href="http://voip-info.org/" target="_blank">
voip-info.org</a><br>>about RTP issues with NAT, check the option canreinvite in sip.conf,<br>>put canreinvite=no , may be that will help. If you have one of the UA<br>>behind a NAT, use nat=yes<br><br>>regards
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