On Sat, May 25, 2019, at 2:34 PM, Saint Michael wrote: > Joshua > Is there a way in PJSIP to send the audio between the parties always, > unless one of the parties is behind a NAT? > A session refresh would work. > That my only problem with PJSIP. This is routine in the old sip channel.
Any such functionality would be documented on the wiki[1]. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Configuration_res_pjsip -- Joshua C. Colp Digium - A Sangoma Company | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
