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Jöran Vinzens

2020-Aug-10 13:57 UTC

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

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 at
localhost:8088/ari/channels/mycallerid.1?endpoint=PJSIP/1003 at
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 at
localhost:8088/ari/channels/mycallerid.1?endpoint=PJSIP/1003 at
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
>
> --
> _____________________________________________________________________
> -- 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
>
> --
> _____________________________________________________________________
> -- 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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Jöran Vinzens

2020-Aug-10 13:59 UTC

head link

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

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 at
localhost:8088/ari/channels/mycallerid.1?endpoint=PJSIP/1003 at
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 at
localhost:8088/ari/channels/mycallerid.1?endpoint=PJSIP/1003 at
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
>>
>> --
>> _____________________________________________________________________
>> -- 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
>>
>> --
>> _____________________________________________________________________
>> -- 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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Dan Cropp

2020-Aug-10 14:34 UTC

head link

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

Thank you Jöran

That did the trick.
I had been trying to figure out how to do this without the json content and
couldn’t figure out how to do it.

Dan

From: asterisk-users <asterisk-users-bounces at lists.digium.com> On
Behalf Of Jöran Vinzens
Sent: Monday, August 10, 2020 8:59 AM
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,

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<mailto: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<mailto: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<mailto: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<mailto: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<mailto: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<mailto: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<mailto:asterisk-users at
lists.digium.com>' <asterisk-users at
lists.digium.com<mailto: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 at
localhost:8088/ari/channels/mycallerid.1?endpoint=PJSIP/1003 at
1003&app=hello-world&extension=1000&context=mycontext&priority=1&channelId=mycallerid.1&formats=ulaw&timeout=30&callerId=Dan<291<http://asterisk:astersk
at localhost:8088/ari/channels/mycallerid.1?endpoint=PJSIP/1003 at
1003&app=hello-world&extension=1000&context=mycontext&priority=1&channelId=mycallerid.1&formats=ulaw&timeout=30&callerId=Dan%3c291>>

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 at
localhost:8088/ari/channels/mycallerid.1?endpoint=PJSIP/1003 at
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
--
_____________________________________________________________________
-- 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<mailto: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<http://www.sipgate.de> -
www.sipgate.co.uk<http://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<mailto: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<http://www.sipgate.de> -
www.sipgate.co.uk<http://www.sipgate.co.uk>


--

Jöran Vinzens - vinzens at sipgate.de<mailto: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<http://www.sipgate.de> -
www.sipgate.co.uk<http://www.sipgate.co.uk>
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