Calls to 911 imediately drop

Hello,
Just found out that calls to 911 drop immediately. We are set up as a PRI with AT&T.
Below is what I see from the CLI when dialing 911 from my desk.
Can someone please help?

== Using SIP RTP CoS mark 5
– Executing [911@users:1] Playback(“SIP/x228-0004969f”, “emergency”) in new stack
– <SIP/x228-0004969f> Playing ‘emergency.gsm’ (language ‘en’)
– Executing [911@users:2] SayDigits(“SIP/x228-0004969f”, “911”) in new stack
– <SIP/x228-0004969f> Playing ‘digits/9.gsm’ (language ‘en’)
– <SIP/x228-0004969f> Playing ‘digits/1.gsm’ (language ‘en’)
– <SIP/x228-0004969f> Playing ‘digits/1.gsm’ (language ‘en’)
– Executing [911@users:3] Playback(“SIP/x228-0004969f”, “connecting”) in new stack
– <SIP/x228-0004969f> Playing ‘connecting.gsm’ (language ‘en’)
– Executing [911@users:4] Set(“SIP/x228-0004969f”, “CALLERID(num)=9897524275”) in new stack
– Executing [911@users:5] Set(“SIP/x228-0004969f”, “CALLERID(name)=Glastender”) in new stack
– Executing [911@users:6] Dial(“SIP/x228-0004969f”, “DAHDI/g1/911”) in new stack
[Apr 7 08:35:36] WARNING[16275]: channel.c:5754 ast_request: No channel type registered for ‘DAHDI’
[Apr 7 08:35:36] WARNING[16275]: app_dial.c:2345 dial_exec_full: Unable to create channel of type ‘DAHDI’ (cause 66 - Channel not implemented)
== Everyone is busy/congested at this time (1:0/0/1)
– Executing [911@users:7] Hangup(“SIP/x228-0004969f”, “”) in new stack
== Spawn extension (users, 911, 7) exited non-zero on 'SIP/x228-0004969f’
trumpetvine*CLI>
Disconnected from Asterisk server
Executing last minute cleanups

[quote]Apr 7 08:35:36] WARNING[16275]: channel.c:5754 ast_request: No channel type registered for ‘DAHDI’
[Apr 7 08:35:36] WARNING[16275]: app_dial.c:2345 dial_exec_full: Unable to create channel of type ‘DAHDI’ (cause 66 - Channel not implemented)[/quote]

You need to load the DAHDI driver in order to use it.