Want to Get information on the Phone Creation through the API

[varun.srinivasan]

For the following issue when creating a WebRTC phone using the Genesys API:
• How to identify and get line base setting and lines for the webRTC user?
• There are multiple lines available under same linebasesetting, which one we have to tag to webrtc user.

We are getting issues relating to the Lienbase settings as required. But we are using WEBRTC
Wanted to know iof anyone has experienced similar issue.

[Priti]

Below APIs we are using for the above query:
POST : https://api.usw2.pure.cloud/api/v2/telephony/providers/edges/phones
GET : /api/v2/telephony/providers/edges/lines
GET : /api/v2/telephony/providers/edges/linebasesettings

[crespino]

For questions like this I usually recommend to launch Genesys Cloud in a browser, open the browser's developer tools, and use the Network tab to watch the traffic as you perform the actions in Genesys Cloud. In this case you should monitor the network traffic while create a Base Settings and you'll see the API requests that Genesys Cloud is making to perform that action. You can then repeat those same API requests in your code.

A quick test in my development org shows the following:

GET /api/v2/telephony/providers/edges/phonebasesettings/availablemetabases?pageSize=25&pageNumber=1 to get the list of the available "Phone Make and Model" lists.

GET /api/v2/telephony/providers/edges/phonebasesettings/template?phoneMetabaseId=inin_webrtc_softphone.json to get the WebRTC Base Template

POST /api/v2/telephony/providers/edges/phonebasesettings to save the new Base Setting.

Take a look at the payloads that Genesys Cloud is filling out when the UI is used and then you'll know what to fill out when making the API request from your application.

[Priti]

Hi Crespino,
Thanks for the update.
https://api.usw2.pure.cloud/api/v2/telephony/providers/edges/lines?lineBaseSettingsId=fff88565-ea27-43bd-a697-799fff810ab4 this API is giving 50 lines in response. Which one should get used? can we use any line?

[tim.smith]

If you're unsure which of your lines to use, check with your telephony administrator that set up the lines. They should be able to guide you on which of your configs you should use. Another option would be do a test like Jim said. Create a WebRTC phone via the UI and see what the UI is doing and use those settings.