Interesting, thanks George. I pulled Asterisk 13 from git and the new pjproject from the SVN and will test accordingly .
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 . Regards, Kevin Long
smime.p7s
Description: S/MIME cryptographic signature
-- _____________________________________________________________________ -- 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
