Volume issue with analogue phone

Asterisk Asterisk Support
1

Hello,

I have an analogue phone connected to Asterisk via a Linksys SPA2102 VoIP/SIP interface. The gain on the speaker’s voice on this phone is low but I can increase it successfully with the following in the extension configuration:

Set(VOLUME(TX)=3);

A problem arises however when the caller to this extension makes a transfer to another number, the “transfer” audio message becomes blisteringly loud and distorted (the distortion starts at VOLUME(TX)=2).

Is there some way I can lower the gain on the message for this particular extension only?

I’m running Asterisk 13.18.3

Thank you,

2

I suppose the trick would be to set the volume back to a lower value before the transfer code is sent by the softfone. It is actually a script that is sending the DTMF codes to a Linphone daemon, so the script could potentially send a message to Asterisk to change the volume. I’m still learning the basics of Asterisk and this messaging, if possible, I don’t know how to do.
Thanks

3

Post the dial plan it is easier to help you with that, but if you get the variable holding that device you can set the volume lower using ExecIf

4

Thanks, I’ll look at ExecIf. Pasted below are the relevant entries in extensions.ael. The sequence is that extension 8001 (a softphone on the Asterisk server itself) calls ext. 601 and then makes a transfer to extension 602. The caller on ext 8001 that hears the distorted signal.

	601 => {     // green phone
        System(oscsend localhost 11222 /micthrough sii on 0 1); 
	Set(CHANNEL(hangup_handler_push)=hangerupper,voicecall,1(args));
	Set(VOLUME(TX)=4);     // increase gain on green phone speaker's voice
   	    Dial(SIP/sip-spa1,20,tT);
    	    Hangup();
	    }

	602 => {   // white phone
        System(oscsend localhost 11222 /micthrough sii on 0 1); 
	Set(CHANNEL(hangup_handler_push)=hangerupper,voicecall,1(args));
   	    Dial(SIP/sip-spa2,20,tT);
    	    Hangup();
	    }
6

Still haven’t got it working, could you please elaborate a little?

7

There is audiohok on the volume function, so the channel with the increased volume need to be adjusted after the transfer, I dont use ael but i think you can simple use this same syntax with the correct volume

Set(VOLUME(TX)=x);
8

Thanks but how can that be done? In features.conf I am using the default “*2” to set “atxfer”. Is there a way of manually performing the attended transfer with an applicationmap and resetting the volume in the process?

9

My calling phone is SIP/switchboard1 and the receiving phone with the low gain on its microphone is SIP/sip-spa1. In CLI, if instead of adjusting the volume for the SIP/switchboard1 channel with, for example:

dialplan set chanvar SIP/switchboard1-0000002d VOLUME(TX) 4

I instead set for the following for the receiving channel:

dialplan set chanvar SIP/sip-spa1-0000002b VOLUME(RX) 4

The volume of the phone SIP/sip-spa1 is corrected and there is no distortion on the voice prompt when the caller on SIP/switchboard1 makes the transfer since the voice prompt pertains to its own channel and not that of SIP/sip-spa1.

Is it possible to configure this volume adjustment on the receiver’s channel via the dialplan and if so, how?

10

Found a solution in pre-dial handlers. The handler allows the gain to be set on the receiving phone’s channel before the dial:

here’s are the relevant bits in 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();
	}	 
}
1 Like
11

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