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

Jöran Vinzens vinzens at sipgate.de
Mon Aug 10 08:59:23 CDT 2020


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 <vinzens at 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 <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
>
>

-- 

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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