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. -- 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
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