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.