date:20200810

Re: [asterisk-users] ARI Stop Playback

2020-08-10 Thread Dan Cropp

Thank you Joshua.  That matches what I experienced last week.

I will build the string to play the number prompts using individual sound 
prompts instead of using number.

Dan


From: asterisk-users  On Behalf Of 
Joshua C. Colp
Sent: Monday, August 10, 2020 8:49 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Subject: Re: [asterisk-users] ARI Stop Playback

On Thu, Aug 6, 2020 at 7:28 PM Dan Cropp 
mailto:d...@amtelco.com>> wrote:
Should the ARI DELETE /playback/{playbackId} be able to stop a playback when a 
number is being played?

Here is a test I am running.  I am playing multiple portions (sounds and 
numbers).

curl -v -u asterisk:asterisk -X POST 
http://localhost:8088/ari/channels/1596750578.46/play?media=sound:hello-world,sound:tt-monkeys,number:553,sound:demo-instruct,number:123,number:456,number:789

If I attempt to stop while any of the numbers portions is being played, it does 
not stop the playback.  I hear a skip or hiccup, but it continues playing.
When I send the same exact stop playback a few seconds later (after the 553 
portion playing the sound portion), it does stop the playback.  I have run this 
test several times and it’s failing.

curl -v -u asterisk:asterisk -X DELETE 
http://localhost:8088/ari/playbacks/3b38730d-7954-4c50-9cdc-e4643ffc8c62

I am using 16.12.0

Is there some reason a DELETE playback doesn’t work during a number portion or 
is this a bug?

They are two separate paths and implementations, and I don't think the number 
part has implemented the handling of stop and such.

--
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
Check us out at www.sangoma.com and 
www.asterisk.org
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Re: [asterisk-users] With ARI, is it possible to create (originate) a call and pass both the caller id name and number?

2020-08-10 Thread Dan Cropp

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  On Behalf Of 
Jöran Vinzens
Sent: Monday, August 10, 2020 8:59 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion 

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 
mailto: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";
 --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 
mailto: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 
mailto: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 
mailto: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 
mailto: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 
mailto: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' 
mailto: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: 6
CallerID: Dan Cropp <291>
Variable: 
CALLERID(num-pres)=allowed_passed_screen,TrunkAllocateId=1,OriginateCallId=2
Async: true

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

2020-08-10 Thread Jöran Vinzens

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  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";
> 
>  --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  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  *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  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  *On
>> Behalf Of *Dan Cropp
>> *Sent:* Friday, August 7, 2020 11:51 AM
>> *To:* '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: 6
>>
>> 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-di

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

2020-08-10 Thread Jöran Vinzens

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  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  *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  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  *On
> Behalf Of *Dan Cropp
> *Sent:* Friday, August 7, 2020 11:51 AM
> *To:* '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: 6
>
> 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
-- 
___

Re: [asterisk-users] ARI Stop Playback

2020-08-10 Thread Joshua C. Colp

On Thu, Aug 6, 2020 at 7:28 PM Dan Cropp  wrote:

> Should the ARI DELETE /playback/{playbackId} be able to stop a playback
> when a number is being played?
>
>
>
> Here is a test I am running.  I am playing multiple portions (sounds and
> numbers).
>
>
>
> curl -v -u asterisk:asterisk -X POST
> http://localhost:8088/ari/channels/1596750578.46/play?media=sound:hello-world,sound:tt-monkeys,number:553,sound:demo-instruct,number:123,number:456,number:789
>
>
>
> If I attempt to stop while any of the numbers portions is being played, it
> does not stop the playback.  I hear a skip or hiccup, but it continues
> playing.
>
> When I send the same exact stop playback a few seconds later (after the
> 553 portion playing the sound portion), it does stop the playback.  I have
> run this test several times and it’s failing.
>
>
>
> curl -v -u asterisk:asterisk -X DELETE
> http://localhost:8088/ari/playbacks/3b38730d-7954-4c50-9cdc-e4643ffc8c62
>
>
>
> I am using 16.12.0
>
>
>
> Is there some reason a DELETE playback doesn’t work during a number
> portion or is this a bug?
>

They are two separate paths and implementations, and I don't think the
number part has implemented the handling of stop and such.

-- 
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org
-- 
_
-- 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

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

2020-08-10 Thread Dan Cropp

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  On Behalf Of 
Jöran Vinzens
Sent: Friday, August 7, 2020 12:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion 

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 
mailto: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 
mailto: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' 
mailto: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: 6
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

Re: [asterisk-users] ARI Stop Playback

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

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

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

Re: [asterisk-users] ARI Stop Playback

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

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