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dan at amtelco.com Guest
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dan at amtelco.com Guest
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Posted: Fri Aug 07, 2020 12:05 pm Post subject: [asterisk-users] With ARI, is it possible to create (origina |
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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@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
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vinzens at sipgate.de Guest
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Posted: Fri Aug 07, 2020 12:10 pm Post subject: [asterisk-users] With ARI, is it possible to create (origina |
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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@amtelco.com (dan@amtelco.com)> wrote:
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dan at amtelco.com Guest
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Posted: Fri Aug 07, 2020 2:10 pm Post subject: [asterisk-users] With ARI, is it possible to create (origina |
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Thank you Jöran
I also figured out my problem with the caller id name/number. In case anyone else encounters the caller id name issue, replace the spaces in the name with control sequence for a space %20
From: asterisk-users <asterisk-users-bounces@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 <dan@amtelco.com (dan@amtelco.com)> wrote:
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dan at amtelco.com Guest
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Posted: Mon Aug 10, 2020 8:42 am Post subject: [asterisk-users] With ARI, is it possible to create (origina |
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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@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 <dan@amtelco.com (dan@amtelco.com)> wrote:
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vinzens at sipgate.de Guest
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Posted: Mon Aug 10, 2020 8:58 am Post subject: [asterisk-users] With ARI, is it possible to create (origina |
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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" --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@amtelco.com (dan@amtelco.com)> wrote:
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vinzens at sipgate.de Guest
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Posted: Mon Aug 10, 2020 9:00 am Post subject: [asterisk-users] With ARI, is it possible to create (origina |
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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" --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@sipgate.de (vinzens@sipgate.de)> wrote:
Quote: | 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" --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@amtelco.com (dan@amtelco.com)> wrote:
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dan at amtelco.com Guest
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Posted: Mon Aug 10, 2020 9:35 am Post subject: [asterisk-users] With ARI, is it possible to create (origina |
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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@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@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" --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@sipgate.de (vinzens@sipgate.de)> wrote:
Quote: |
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" --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@amtelco.com (dan@amtelco.com)> wrote:
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