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

[Rich Adamson]

> 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.

Without seeing the appropriate sections of your config files, I'd have
to take a pure guess that you're not using contexts in the correct way.

Can you post just those sections that pertain to this (don't need the
entire file)?