On Fri, May 24, 2019, at 9:47 AM, Dan Cropp wrote: > > We are working with an Avaya switch. > > > We send them a REFER. If the transfer is successful, everything is > great. If it fails (busy), they send an INVITE in-dialog with a media > attribute of inactive. After that, they send a 486 busy. > > The problem is Avaya basically put the call on hold so audio is not active. > > The Avaya rep is indicating we need to send in dialog invite to get the > call audio back? They are essentially saying they put the call on hold > because we told them to transfer and it’s our responsibility to take > the call off hold. > > > Is there a way to do this?
I don't think there is. We provide the ability in PJSIP to do a session refresh[1] but there's no ability to set the stream state like that, so I'm not sure what we would specify in that scenario automatically. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Function_PJSIP_SEND_SESSION_REFRESH -- 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
