dear<br><br><br>please Help. I am continously getting this message after "sip set debug on". and not getting clear voice from both side.<br><br><br><--- Transmitting (NAT) to <a href="http://122.163.193.94:1893">122.163.193.94:1893</a> ---><br>
SIP/2.0 404 Not Found<br>Via: SIP/2.0/UDP 192.168.1.106:5060;branch=z9hG4bK-323331-c72a5bb7c840f6a183902fdcca79241b;received=122.163.193.94;rport=1893<br>From: "2002" <<a href="mailto:sip%3A2002@122.160.154.189">sip:2002@122.160.154.189</a>>;tag=5a1cc54c<br>
To: "2002" <<a href="mailto:sip%3A2002@122.160.154.189">sip:2002@122.160.154.189</a>>;tag=as64f1f102<br>Call-ID: 8c18bd84585128a3f0885f54dfa966ba@0:0:0:0:0:0:0:0<br>CSeq: 245 OPTIONS<br>Server: Asterisk PBX 10.0.0<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH<br>Supported: replaces, timer<br>Accept: application/sdp<br>Content-Length: 0<br><br><br><------------><br>Scheduling destruction of SIP dialog '8c18bd84585128a3f0885f54dfa966ba@0:0:0:0:0:0:0:0' in 32000 ms (Method: OPTIONS)<br>
Really destroying SIP dialog '6636e2ee56cce3b0e2287ebe51962c84@0:0:0:0:0:0:0:0' Method: OPTIONS<br>Really destroying SIP dialog '4ec2032c2b9a58e1cf2d1afa70c1970b@0:0:0:0:0:0:0:0' Method: OPTIONS<br><br>