Hi Dan, i would do something like this (it is not a copy of what we are doing but an example of how i would do it) Important here is the "--data" and "-H" Option as well as the "variables" section within the Body. I added the callerid function here as well as it is the sample in the asterisk wiki.
curl -v -H "Content-Type: application/json" -u asterisk:asterisk -X POST " http://localhost:8088/ari/channels/newChannelId"; <http://localhost:8088/ari/channels/1400609726.3/play?media=sound:hello-world> --data '{ "endpoint": "SIP/Alice", "variables": { "CALLERID(name)": "Alice" , "PJSIP_HEADER(add,P-Asserted-Identity":"foobar"} }' BR Jöran On Mon, Aug 10, 2020 at 3:43 PM Dan Cropp <d...@amtelco.com> wrote: > Hi Jöran, > > > > Would it be possible to see an example using curl of how you are passing > the PAI Header through ARI create? > > > > Dan > > > > *From:* asterisk-users <asterisk-users-boun...@lists.digium.com> *On > Behalf Of *Jöran Vinzens > *Sent:* Friday, August 7, 2020 12:10 PM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion < > asterisk-users@lists.digium.com> > *Subject:* Re: [asterisk-users] With ARI, is it possible to create > (originate) a call and pass both the caller id name and number? > > > > 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 <d...@amtelco.com> 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 <asterisk-users-boun...@lists.digium.com> *On > Behalf Of *Dan Cropp > *Sent:* Friday, August 7, 2020 11:51 AM > *To:* 'asterisk-users@lists.digium.com' <asterisk-users@lists.digium.com> > *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 - vinz...@sipgate.de > 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 -- Jöran Vinzens - vinz...@sipgate.de 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
