<html xmlns:v="urn:schemas-microsoft-com:vml" xmlns:o="urn:schemas-microsoft-com:office:office" xmlns:w="urn:schemas-microsoft-com:office:word" xmlns:m="http://schemas.microsoft.com/office/2004/12/omml" xmlns="http://www.w3.org/TR/REC-html40"><head><META HTTP-EQUIV="Content-Type" CONTENT="text/html; charset=us-ascii"><meta name=Generator content="Microsoft Word 12 (filtered medium)"><style><!--
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</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>ADTRAN has some interesting Voice Quality Monitoring built into their switches, routers, etc: <a href="http://adtran.com/web/url/vqm">http://adtran.com/web/url/vqm</a><o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'><p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] <b>On Behalf Of </b>Stefan at WPF<br><b>Sent:</b> Wednesday, June 20, 2012 2:05 PM<br><b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br><b>Subject:</b> [asterisk-users] Proactive problem monitoring on SIP on Asterisk<o:p></o:p></span></p></div><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>Hello,<br><br>1) I am wondering what is the best practice to monitor if there are or were problems with SIP calls on my Asterisk box. E.g. how about a software that extracts all calls from the /var/log/asterisk/full (I have permanently enabled verbose 10 and sip debug) log and tells me on which of them were problems? Checking the logs manually is very hard, but as SIP is a standardized protocoll, there should be tools doing that for you? As an example, a person calling me recently got a 488 Not acceptable error as reply from my Asterisk box. Nothing came through to my SIP phone, so I didn't know anything about the call or the problems (which were on his phone btw). I would like to be informed about such cases, know that there was a call to my Asterisk box that made problems.<br><br>2) How about monitoring speech quality? E.g. sometimes it seems like a packet is missing (I then have a short pause during the call), how to monitor such things and create statistics out of this data?<br><br>So basically I want to monitor my Asterisk installation proactively for reliability/problems and (speech) quality.<br><br>Thanks for any hints!<br><br>Best regards<br>Stefan<o:p></o:p></p></div></body></html>