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>
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
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 <jcolp at digium.com> 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 - 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/20190117/08be5a8a/attachment.html>