Access certain functions

[satock1222]

How to access certain functions such as calling, mute, hold from this library and use it in our own custom dial pad component?

[prinze77]

Thank you for the interest to this project satock1222!

The current version does not gives us the opportunity to use it with custom dial pad.

I can try to make it more flexible if you explain me whats wrong with the current dial pad componets, or you can even show me yours dial pad componets so maybe i use them for this softphone

I tried to make it as much simple as possible. With only one "import" you adding a web softphone without the need of making custom diallpads and etc

[satock1222]

Could you explain how transfer and conference call works? I could get the normal call to work but haven't had any positive outcome with transfer and conference call. Also what are the channels 'Ch1', 'Ch2' and 'Ch3' for?

[prinze77]

The use case of channels is simply to put call on hold, switch to next channel and make a new call while the first call is on hold, then you can put on hold current channel switch back to first channel and unhold it to continue the conversation. Unfortunately right now you can not make conversation within the channels, but it sounds like a good feature to add.

To make a conversation call you need to make a call to first person then after he answers you should input the second participant number and press the attended transfer button, it will put on hold the first participant an start calling the second one after the second answer the phone you can press the new buttons tha will appear, make conversation with the first participant that was on hold, with the second participantand with you, also there will be a button to simply connect first and second participant without you or you can press the button to hung up the call with second participant and switch back to first participant

All this call manipulations currently work with asterisk server, because our web soft phone simply sends commands to voip server and voip server hundles the call transfers etc, for example to make a call forward with asterisk voip server you need to input the following ##numbertotranfer if your voip server needs a different command to make the same action then you will need to make some changes to make it work like asterisk work .

[satock1222]

For incoming calls you can't convert it into a conference call? Also I have asterisk configured to take normal calls, anything extra needed to be done on asterisk side to get transfer/conference call features to work? Also thank you for the explanation!

[prinze77]

The above scheme works for incoming calls aswell. By default asterisk can tranfer calls. More here https://wiki.asterisk.org/wiki/display/AST/Feature+Code+Call+Transfers.