[asterisk-users] With ARI, is it possible to create (originate) a call and pass both the caller id name and number?

Mon Aug 10 08:57:31 CDT 2020

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 <dan at 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-bounces at 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 at 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 <dan at 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-bounces at lists.digium.com> *On
> Behalf Of *Dan Cropp
> *Sent:* Friday, August 7, 2020 11:51 AM
> *To:* 'asterisk-users at lists.digium.com' <asterisk-users at 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 at 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
>
> --
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>
>
>
> --
>
> Jöran Vinzens - vinzens at 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 - vinzens at 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
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