[asterisk-users] Early media using ARI

Jöran Vinzens vinzens at sipgate.de
Thu Jan 17 09:40:10 CST 2019

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 - vinzens at sipgate.de

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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20190117/77aebbd3/attachment.html>

More information about the asterisk-users mailing list