On Mon, Jul 1, 2019, at 11:54 AM, Jason N wrote: > Unfortunately I am not allowed any changes to H's PBX / dialplan. > The restriction I have is that upon H's total disconnection from C, > that S continues the call with C. That's why I thought that if I could > get S to SIP JOIN the call from C, that once H disconnects S can > continue. I can extract the SIP call info on H and pass that to S (so > it can join the call). > > I'm just not sure if this concept is possible/practical.
There is no such thing as "joining" a call like that in Asterisk. It would be trying to do server side three way calling, which is not supported like that. -- 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
