HI,<div>Instead of taking traces on asterisk console, try capturing traffic in pcap using tcpdump command and later analyze it with wireshark.</div><div><font face="'courier new', monospace">#tcpdump -i eth<i>N </i>-s 0 -w sip-capture.pcap -v</font></div>
<div><font face="arial, helvetica, sans-serif">Download this file to any machine with wireshark installed and apply "sip" as filter </font><span style="font-family:arial,helvetica,sans-serif">there</span><span style="font-family:arial,helvetica,sans-serif">.</span></div>
<div>Regards.</div><div>Sammy</div><div><br></div><div>On Fri, Mar 2, 2012 at 11:46 PM, Markus <span dir="ltr"><<a href="mailto:universe@truemetal.org">universe@truemetal.org</a>></span> wrote:<div class="gmail_quote">
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">For a few days now I'm getting<br>
<br>
"chan_sip.c: No compatible codecs, not accepting this offer!"<br>
<br>
on the CLI and in the messages log in irregular intervals. How can I find out what (who) is causing this? If I turn on "sip debug" I get flooded with SIP messages from peers/users (qualify=yes etc.) and it's hard to analyze.<br>
<br>
Thanks!<br>
<br>
<br>
--<br>
______________________________<u></u>______________________________<u></u>_________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
New to Asterisk? Join us for a live introductory webinar every Thurs:<br>
<a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br>
<br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/<u></u>mailman/listinfo/asterisk-<u></u>users</a><br>
</blockquote></div><br></div>