<div> </div>
<div>This works for us....</div>
<div> </div>
<div> </div>
<div> </div>
<div>exten => h,1,Set(CDR(userfield)=${RTPAUDIOQOS})<br>exten => h,2,Hangup()<br></div>
<div>results in....</div>
<div>Set("SIP/rpx2399a-b61fc5e0", "CDR(userfield)=ssrc=213416392;themssrc=0;lp=0;rxjitter=0.000000;rxcount=0;txjitter=0.000000;txcount=0;rlp=0;rtt=0.000000")</div>
<div> </div>
<div> </div>
<div> </div>
<div>
<p class="MsoNormal" style="MARGIN: 0in 0in 0pt"><span style="FONT-SIZE: 11pt; COLOR: #1f497d; FONT-FAMILY: 'Calibri','sans-serif'; mso-ascii-theme-font: minor-latin; mso-hansi-theme-font: minor-latin; mso-bidi-font-family: 'Times New Roman'; mso-bidi-theme-font: minor-bidi; mso-themecolor: dark2"> </span></p>
<p class="MsoNormal" style="MARGIN: 0in 0in 0pt"><span style="FONT-SIZE: 11pt; COLOR: #1f497d; FONT-FAMILY: 'Calibri','sans-serif'; mso-ascii-theme-font: minor-latin; mso-hansi-theme-font: minor-latin; mso-bidi-font-family: 'Times New Roman'; mso-bidi-theme-font: minor-bidi; mso-themecolor: dark2"> </span></p>
<p class="MsoNormal" style="MARGIN: 0in 0in 0pt"><b><span style="FONT-SIZE: 10pt; FONT-FAMILY: 'Tahoma','sans-serif'; mso-fareast-font-family: 'Times New Roman'">From:</span></b><span style="FONT-SIZE: 10pt; FONT-FAMILY: 'Tahoma','sans-serif'; mso-fareast-font-family: 'Times New Roman'"> <a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a> [mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>] <b>On Behalf Of </b>Max Alex<br>
<b>Sent:</b> Friday, November 14, 2008 5:25 AM<br><b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion; <a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>; <a href="mailto:asterisk-users-request@lists.digium.com">asterisk-users-request@lists.digium.com</a><br>
<b>Subject:</b> [asterisk-users] RTP LOG</span></p>
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<p class="MsoNormal" style="MARGIN: 0in 0in 12pt"><font size="3"><font face="Times New Roman">Hi All,<br>I am using asterisk 1.4.22 in my local system<br>I want to know how can we set ability to log and report RTP and jitter statistics per call.<br>
Is there any configuration in logger or configuration in rtp?<br>Please provide some guide lines for this.<br>Thanks in advance!<br><br clear="all">Thanks,<br>Max Alex<br>Voip Developer</font></font></p></div>