Control of SIP phone volume on attended transfer in Asterisk

Question

I've been trying to resolve a volume issue. I have an analogue SIP phone that has low gain on its microphone. This can be remedied by putting the following in its extension.ael config:

Set(VOLUME(TX)=4);

The problem is that the caller to this extension will be making attended transfers and the change in channel volume distorts the voice prompt "transfer" and the subsequent dial tone.

Is there a way that I can redefine "atxfer" in features.conf, or use a featuremap to make the transfer, such that the volume of the channel is set back to 1 before the transfer is made? I would like to do the opposite after that, ie. return the volume of the extension to 4 when the transfer is finalised with "atxferthreeway".

If anyone can help with this or has other suggestions, please let me know.

Answer 1

You can do manual version of transfer via feature codes(features.conf)

No, there are no option like change volume while transfer.

Answer 2

I found a solution, the problem is that the gain increase should be set on the receiver's channel and not on that of the caller. This way the voice prompts on the caller's channel don't get distorted. The solution uses a pre-dial handler. Here are the relevant parts from extensions.ael:

context outgoing {

    greenphone_pre_dial_handler => {
    Set(VOLUME(RX)=5);
    Return();
    }

    601 => {     // green phone
    Set(VOLUME(TX)=1);     
    Dial(SIP/sip-spa1,20,Tb(outgoing^greenphone_pre_dial_handler^1));
    Hangup();
    }    
}