asterisk users - Oct 2006 - SIP trunk from an Audiocodes mediant 1000

Hi,

I am configuring an audiocodes Medant1000 to talk to my asterisk box.
So far I have successfull in landing a single call from mediant to my
*box. my sip conf is as follows:

[general]
context=sip
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes

[3911700]
type=friend
host=dynamic
dtmfmode=info
secret=blah
context=sip

where  3911700 is my E1 telephone no. in my extensions.conf I have

exten => 3911700,1,Dial(SIP/100)

When I dial from outside to my E1 number calls are coming like the following:

INVITE sip:3911700@192.168.9.210;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.9.230;branch=z9hG4bKac806223297
Max-Forwards: 70
From: <sip:9387802673@192.168.9.230>;tag=1c806218385
To: <sip:3911700@192.168.9.210;user=phone>
Call-ID: 80621773621200024215@192.168.9.230
CSeq: 1 INVITE
Contact: <sip:9387802673@192.168.9.230>
Supported: em,100rel,timer,replaces,path
Allow:
REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Remote-Party-ID: <sip:3911700@192.168.9.210>;party=called;npi=1;ton=4
Remote-Party-ID:
<sip:9387802673@192.168.9.210>;party=calling;privacy=off;screen=yes;screen-ind=1;npi=1;ton=0
User-Agent: Audiocodes-Sip-Gateway-Mediant 1000/v.4.60A.016.003
Content-Type: application/sdp
Content-Length: 348

and the call get's connected to SIP/100 via the line in extensions.conf

But what I am expecting is that the calls to come to the context's
's'
extension. I am not sure if the changes are to be done in Asterisk or
to Mediant.

Any help in this will be much appreciated.

raj