Low volume at endpoint using DISA

Inbound PSTN trunk. Outbound IAX trunk. Final destination on the PSTN.

Asterisk 1.4 on an IP04.

The called party can hardly hear me, while I can hear the called party just fine. Each part of the link works just fine during testing, but not so when strung together.

I see that there is a VOLUME command in 1.6. I suppose that’ll do what I want, but how would I fix this - or figure out what is wrong - in 1.4?


Connecting an inbound call on the subscriber end of a PSTN line to an outgoing call via another PSTN line never works very well. The reason is that the PSTN, when connecting any two end-points together, is designed to insert about 10 db of loss between them. (Normal analog telephones are designed for 10 db of ‘via net loss’ so they work well on the PSTN.) Connecting two PSTN connections together, however, results in 20 db of via net loss. This is why a multi-line analog phone with a conferencing function usually doesn’t work well. It either makes it hard for the two outside parties to hear each other, or it tries to overcome this limitation by introducing gain between the two lines, which usually produces distortion or feedback.

DISA works very will if you use VoIP inbound and outbound. You get a virtually lossless connection to the PSTN through the SIP (or IAX) origination and termination providers, and the PSTN is only inserting 5 db of loss on each of the two connections, thus resulting in a total of 10 db of total via net loss through your switch.

If Asterisk can introduce gain between two PSTN lines without introducing feedback, then give that a try. It may solve your problem. I routinely use DISA between SIP trunks, and the connection results in normal volume in both directions.

Thanks for the explanation Dave.

In this case, my phone plan allows me free calls between my mobile and my analog home phone. You can see where this is going.

So VOIP incoming is not a go. And since my recipients may be anywhere, VOIP on the third leg is a no go too. The only VOIP leg is the middle one.

If I can find a way to inject a bit of gain, then I’ll bear your explanation in mind.


My cell plan gives me a short list of ‘friends and family’. I have an inbound SIP number which is a ‘friend’ of my cell phone. I dial into my * system on this number, and then use a SIP termination provider for the outbound leg. Maybe you can do something like this.