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 _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
