Hi Dan, as far as PPI and PAI Header, we use the channel Vars in order to do that. In Latest Asterisk you can set Channel vars within the create command in the Body. Just set the PJSIP(add,PAI) as you would do it in Dialplan. https://blogs.asterisk.org/2020/07/22/ari-create-channel-with-variables/ BR Jöran
On Fri, Aug 7, 2020 at 7:06 PM Dan Cropp <[email protected]> wrote: > An additional follow-up question, if I need to set the P-Asserted-Identity > on the create (originate), is there a way to do this with ARI? > > > > *From:* asterisk-users <[email protected]> *On > Behalf Of *Dan Cropp > *Sent:* Friday, August 7, 2020 11:51 AM > *To:* '[email protected]' <[email protected]> > *Subject:* [asterisk-users] With ARI, is it possible to create > (originate) a call and pass both the caller id name and number? > > > > I’m trying to transition from AMI to ARI. > > > > Running into a small hiccup when I try to create (originate a call) with > the caller id name and number > > > > I can pass the Name and Number if the name has no spaces in it and it > shows up in my PhonerLite application. > > > > curl -v -u asterisk:asterisk -X POST > http://asterisk:astersk@localhost:8088/ari/channels/mycallerid.1?endpoint=PJSIP/1003@1003&app=hello-world&extension=1000&context=mycontext&priority=1&channelId=mycallerid.1&formats=ulaw&timeout=30&callerId=Dan<291 > > > > > > However, when the caller id name has a space in it, I can’t figure out how > to pass the name and number successfully. The following only displays > asterisk for the number and Dan for the name > > > > curl -v -u asterisk:asterisk -X POST > http://asterisk:astersk@localhost:8088/ari/channels/mycallerid.1?endpoint=PJSIP/1003@1003&app=hello-world&extension=1000&context=mycontext&priority=1&channelId=mycallerid.1&formats=ulaw&timeout=30&callerId=Dan > Cropp<291> > > > > Here is an example of how we do this with AMI successfully. > > Action: Originate > > ActionID: S40 > > Channel: PJSIP/1003@1003 > > Exten: createcall > > Context: IS > > Priority: 1 > > Timeout: 60000 > > CallerID: Dan Cropp <291> > > Variable: > CALLERID(num-pres)=allowed_passed_screen,TrunkAllocateId=1,OriginateCallId=2 > > Async: true > > > > Dan > -- > _____________________________________________________________________ > -- 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] Telefon: +49 211-63 55 56-21 Telefax: +49 211-63 55 55-22 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
