On Tue, Mar 1, 2016 at 5:37 PM, Kevin Long <[email protected]> wrote:
> > > Interesting, thanks George. I pulled Asterisk 13 from git and the new > pjproject from the SVN and will test accordingly . > 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. > > > I have a few more questions about PJSIP in Asterisk 13: > > > 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. > > 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 . > 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. > > > Regards, > > Kevin Long > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
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