asterisk users - Sep 2003 - Can you turn up the gain on sip calls?

Can you turn up the gain on sip calls?
I would like the calls to be louder but
the phones are already at max volume.

Hi,

Just got hold of Ericsson webswitch 100 G4 (4 FXO ports).
IT uses H323 as codec. The plan is to use it for incoming/outgoing calls on
two PSTN lines.
I have ATA 186 which is using SIP to use asterisk services.

I can not figure out:

1. where in asterisk do I edit conf files so it uses webswitch for
incoming/outgoing calls
2. how come there is no username+password on webswitch. (apparently just an
IP and port number)
3. is there any docs on gateway/gatekeeper for using H323 with asterisk.

If some could please explain this to me, I would be very grateful.

Thanks

Senad