I was able to get a full debug report (packet dump and asterisk debug) for one of these dropped calls and it does seem to be the provider that is at fault. I can see that they stop sending RTP packets to Asterisk when this happens and after a while they send a BYE. I will keep investigating though.
<br><br><div><span class="gmail_quote">On 12/22/05, <b class="gmail_sendername">David C. Nicosia</b> <<a href="mailto:david@alchetec.com">david@alchetec.com</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
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<p><font face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial;">In addition to having this with my SIP phones, I have also
experienced it with SCCP.</span></font></p>
<p><font face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial;"> </span></font></p>
<p><font face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial;">It started when I updated to the 1.2 release of asterisk. At
the time I updated I also switched VoIP providers and thought it was them.</span></font></p>
<p><font face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial;"> </span></font></p>
<p><font face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial;">Did you file this as a bug or find a solution to it? Thanks!</span></font></p>
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