[Asterisk-Users] Dialing w/analog phone via FXS port.

[Ken D'Ambrosio]

Argh.  I can't figure out what I'm doing wrong.  I can dial with my SIP
phones just fine, but I want to set up an analog phone plugged into my FXS
port... and, while it gets dialtone, no matter what digit I press, I get
stuff like:

VERBOSE[21963]:     -- Starting simple switch on 'Zap/1-1'
DEBUG[21963]: DTMF digit: 9 on Zap/1-1
DEBUG[21963]: Hangup: channel: 1 index = 0, normal = 13, callwait = -1,
thirdcall = -1
DEBUG[21963]: Set option TDD MODE, value: OFF(0) on Zap/1-1
DEBUG[21963]: Updated conferencing on 1, with 0 conference usersApr  2
VERBOSE[21963]:     -- Hungup 'Zap/1-1'

I've tried to make it as similar to the SIP stuff in zapata.conf as
possible.  Any suggestions on what to read to get this right?  I've RTFM'd
no small amount, but, obviously, not the *right* stuff.  I'll gladly send
my config files to anyone who wants 'em, or will gladly look at
functioning config files anyone wants to send my way.

Thanks!

-Ken