Hi Spence --

In your example URL, I would replace the application portion from:

&%20playback(/vr/migstory.wav)

to

%26conference(CONFERENCE_NAME)

where the parameter is set to something suitable for that particular instance.

Please also note the escaped ampersand sign (%26), be sure to URL-escape all characters which would mislead the browser.

On Tue, 5 Nov 2019 at 11:54, Spencer Angerbauer <spencer.angerbauer@gmail.com> wrote:
Hello,

I am semi new to FreeSwitch but have worked in Asterisk for many years.  I am trying to create a simple originate api call from the webapi module.   I have been able to successful create an internal user extension 1001 and connect to 1002 in a default conference room.  (See below call format which was successful.

http://XX.XX.XX.XX:8080/webapi/originate?user/1001%201002%20XML%20default


My question is, How do I take this same command structure to connect multiple external numbers (not internal extensions) into a specific or unique conference room via my SIP trunk line?  

I have tried multiple variations but am having difficulty finding anything online that shows some good examples of JSON restful webapi’s for originating an outbound call to 2 or more numbers and connecting them into a conference room...

I was able to successfully execute the webapi call below for a simple originate api call to an external number and connect but am unsuccessful and bridging these two api commands into what we need.

http://XX.XX.XX.XX:8080/webapi/originate?{origination_caller_id_number=18015555555}sofia/carriers/111#180155555551@XX.XX.XX.XX%20&%20playback(/vr/migstory.wav)

Please let me know if you have any resources or API references to simply connect 2+ external calls into a unique conference via the webapi calls.

Thank you so much in advance!

-Spence


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--
Best Regards,
Ciprian Dosoftei

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