asterisk users - Apr 2005 - Dialplan not showing up.

I recently updated my sip.conf and extensions.conf files and after 
shutting down asterisk and restarting it (asterisk -cvvv)
it shows and empty dialplan (show dialplan)
*CLI> show dialplan
-= 0 extensions (0 priorities) in 0 contexts. =-
 
What could cause somthing like this
 
below is a copy of my extensions.conf file located in /etc/asterisk/
[root@Asterisk-NY17S asterisk]# more extensions.conf 
[general]
static=yes
writeprotect=yes
 
[bogons]
exten => _.,1,Congestion
 
[from-sip]
exten => 1001,1,Dial(SIP/1001,20)
exten => 1001,2,Voicemail(u1001)
exten => 1001,102,Voicemail(b1001)
exten => 1001,103,Hangup
 
exten => 7301,1,Dial(SIP/7301,20)
exten => 7301,2,Voicemail(u7301)
exten => 7301,102,Voicemail(b7301)
exten => 7301,103,Hangup
 
exten => 7302,1,Dial(SIP/7302,10)
exten => 7302,2,Voicemail(u7302)
exten => 7302,102,Voicemail(b7302)
exten => 7302,103,Hangup
 
exten => 1999,1,VoicemailMain(${CALLERIDNUM})
 
[root@Asterisk-NY17S asterisk]# 
 
 
And here is my sip.conf file:
 
[root@Asterisk-NY17S asterisk]# more sip.conf 
[general]
port=5060
bindaddr=0.0.0.0
allow=all
context=bogons
 
[1001]
type=friend
context=from-sip
username=1001
callerid="1001"<1001>
mailbox=1001@telx.com
host-dynamic
dtmfmode=rfc2833
 
[7301]
type=friend
context=from-sip
username=7301
callerid="mdm" <7301>
mailbox=7301@telx.com
host=dynamic
dtmfmode=rfc2833
 
[7302]
type=friend
context=from-sip
username=7302
callerid="Moses" <7302>
mailbox=7302@telx.com
host=dynamic
dtmfmode=rfc2833
[root@Asterisk-NY17S asterisk]# 

Regards, 
Michael DiMartino 
Director of MIS 
The telx Group, Inc. 
17 State St, 33rd Floor 
New York, NY 10004 
T: 212.480.3300 X2022 
C: 646.207.6603 
  

 
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Hi,


I am trying to figure out how to setup call waiting on a asterisk@home
box. We get the call waiting signal from the telco and would like to be
able to switch calls.
Our setup right now is as following:

[PSTN] -> [asterisk@home] -> [sip to Cisco ATA 188] -> Siemens 8825
(Analog)

When we had the siemens plugged into the PSTN directly, we could switch by
just pressing the line button again. Now this doesn't seem to work going
through the sip channel. Does anyone know how to enable this such that the
switch line signal is propagate back the the PSTN and be able to switch
the call to the call waiting one?


Please let me know

Thanks
Sascha