[asterisk-users] volume control and setting volume on transfer

[Iain Mott]

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, which are 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();
     }
}


On 23/04/2020 16:42, Iain Mott wrote:
> Hello,
>
> 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 resolved by 
> putting the following in its extension 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 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.
>
> Thanks,
>
> Iain
>
>