<div dir="ltr"><div class="gmail_default" style="font-family:arial narrow,sans-serif"><br></div><div class="gmail_extra"><br><div class="gmail_quote">On Tue, Mar 1, 2016 at 5:37 PM, Kevin Long <span dir="ltr"><<a href="mailto:kevin.long@haloprivacy.com" target="_blank">kevin.long@haloprivacy.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><br>
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Interesting, thanks George. I pulled Asterisk 13 from git and the new pjproject from the SVN and will test accordingly .<br></blockquote><div><br></div><div><div class="gmail_default" style="font-family:'arial narrow',sans-serif;display:inline">Yeah, actually you do need Asterisk 13 from git because pjproject deprecated an api in trunk and we only handle that in the current git 13 branch.</div> </div><div><br></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
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I have a few more questions about PJSIP in Asterisk 13:<br>
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1. Is there any way to list current ongoing calls and see what codecs are being used in the RTP streams? With chan_sip, “sip show channels” did this.<br>
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2. Also with a PJSIP initiated call, is there a way to see how man RTP packets have been sent and received for the call , I am debugging some intermittent 1-way and no-way audio on calls , and I am having trouble figuring out fi it is the client, firewall, or Asterisk/pjsip that is the culprit .<br></blockquote><div><br></div><div><div class="gmail_default" style="font-family:'arial narrow',sans-serif">Unfortunately, no to both (at least that I'm aware of). I remember looking at the channel stats a long while back and for some reason didn't go any further. I can re-look.</div></div><div><br></div><div> </div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
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Regards,<br>
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Kevin Long<br>--<br>
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