Hi Dan, i did it wrong, sorry:
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"} }' there was a bracket missing after the function of PJSIP_HEADER BR On Mon, Aug 10, 2020 at 3:57 PM Jöran Vinzens <vinz...@sipgate.de> wrote: > 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 > > -- 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