Hi, thanks for the hint. What we have done so far:
- get an incopming call - create a new channel - set stuff on outgoing channel - dial outgoing channel - get a Dial Evente State "PROGRESS" - push both channels into the bridge then nothing happens by default. we will try your suggested way! (putting both Channels into bridge before dialing the B channel) BR Jöran On Thu, Jan 17, 2019 at 4:49 PM Joshua C. Colp <[email protected]> wrote: > 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 -- Jöran Vinzens - [email protected] sipgate GmbH - Gladbacher Str. 74 - 40219 Düsseldorf HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois Steuernummer: 106/5724/7147, Umsatzsteuer-ID: DE219349391 www.sipgate.de - www.sipgate.co.uk
-- _____________________________________________________________________ -- 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
