On Thu, Jan 17, 2019, at 11:40 AM, Jöran Vinzens wrote: > Hi all, > > we are working on a A to B basic Call scenario with early media. > On that scenario we get a call from a PJSIP endpoint and we place a new > call using ARI. On the created channel we receive a 183 Session > progress where we have an announcement regarding e.g. the cost of the > call (it's important for us to have this announcement to inform our > customers about the costs). > Using asterisk Dialplan this is done by App Dial automatically. > On ARI we receive a Dial Event "PROGRESS" where we thought we put both > channels into a bridge and the asterisk will then forward the RTP > towards the "A" Client using a 183 (since the channel is not answered, > yet). Unfortunately nothing happens. > > We searched the documentation and we have not figured it out. There is > no "/ari/channel/progress" command we can use and there is no > "early_media=true" in pjsip.conf which would enable the desired > behaviour. > > We would love to get a hint in the right direction and we very much > appreciate any help.
There's a blog post which shows how it is supposed to work[1]. It expects the channel to be created, then both put into the bridge, and then dialed. This also requires Asterisk 14 or above to operate. What version are you using? [1] https://blogs.asterisk.org/2016/08/24/asterisk-14-ari-create-bridge-dial/ -- 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
