previously i was using for codec<br>allow=all<br>after that i changed<br>disallow=all<br>allow=silk24<br><br>and i also change softph x-lite from jitsi(because of codec)<br>now voice was coming fine from both side.<br>But when i came to home from office not getting voice from both side.<br>
Threr is Airtel Broadband at my place. <br><br><br><div class="gmail_quote">On Thu, Jul 5, 2012 at 3:36 PM, SamyGo <span dir="ltr"><<a href="mailto:govoiper@gmail.com" target="_blank">govoiper@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Hi,<div><b><span style="color:rgb(34,34,34);font-size:13.333333969116211px;font-family:arial,sans-serif">CSeq: 245 OPTIONS</span> </b></div>
<div><b><br></b></div><div>This is just SIP keep-alive. It has nothing to do with any Call-media degradation. If you are not getting clear voice check the codecs, network latency/delay/loss/jitter parameters.</div>
<div><br></div><div>BR</div><div>Sammy</div><div><br><br><div class="gmail_quote"><div><div class="h5">On Thu, Jul 5, 2012 at 2:34 PM, alok srivastava <span dir="ltr"><<a href="mailto:alokkic@gmail.com" target="_blank">alokkic@gmail.com</a>></span> wrote:<br>
</div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div class="h5">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" target="_blank">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" target="_blank">sip:2002@122.160.154.189</a>>;tag=5a1cc54c<br>
To: "2002" <<a href="mailto:sip%3A2002@122.160.154.189" target="_blank">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>
<br></div></div><span class="HOEnZb"><font color="#888888">--<br>
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