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WebRTC dialer customization


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Hi all,

we are currently experimenting with the webRTC feature. Everything works fine in Chrome, Firefox requests permission to use the microphone and then nothing happens and as for Internet Explorer, clicking the "Make Call" button does nothing.

Anyway, we have the following customer case: the customer wants to add the talk button to their company website and receive incoming calls from their visitors. However, because of the keypad below the "Make Call" button the dialer interface is confusing. Some visitors think that they have to call a number which they do not know and as a result they don't proceed with the call.

Question 1: How can we customize the dialer interface? We primarilly want to hide the keypad and leave only the "Make Call" button. And it would have been nice to change the logo to our own (well, this seems easy, I assume that we only have to change the url sources for the images in the code provided by the pbx).

Question 2: Where is the web server running?

Question 3: I noticed that when the call button is pressed there is a dynamic javascript object that gets instantiated and gets fetched from the server (/usr_callbutton.htm). Is this file located somewhere in the file system and is this the one we have to edit?

(I am aware that my questions may sound naive. I am not the web developer myself, I just want some initial information to pass to him).



PS. Dynamically altering the resulting web page (using javascript to hide elements) is not an option for us.

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Yes usr_callbutton.htm is that file but you can't access it directly. What you can try though is, create a file by that exact name in the html folder of the pbx working directory. And copy all the contents of that file from the browser into it. Once you are sure it works, from there on you can change it according to your needs.


I hope that helps.

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thank you for your reply. We tried your suggestion and it works. However, this is not a suitable solution for the hosted edition since the customized file can serve one customer only (the url for the sip-trunk is hardcoded). We won't be able to provide the same service to any more customers.


Any suggestions to that?

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The customized file will not have any hard-coded URL. The customized file is the one that provides the WebRTC functionality and dialer. This file is the one you can customize according to your needs for the dialer etc. The part that goes in customer webpage should be generated with the "GENERAL" link when you create a new trunk for that customer. That generated code will go in the customer webpage to access your customized usr_callbutton.htm to make calls to the pbx on that trunk. One trunk per customer, therefore one generated code per customer so that call comes into that trunk. And all of them are served your customized file for dial in.


I hope it's clear.

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