[asterisk-users] Connecting Asterisk to a Metaswitch

[kharris]

I am having a difficult time connecting an Asterisk box to a 
Metaswitch.  I looked at the page at 
http://www.voip-info.org/wiki/view/Asterisk+How+to+connect+to+Metaswitch
but was not able to make much progress.  If someone could direct in what 
direction to start troubleshooting this problem I would be very 
appreciative.

When I try to dial out now.  I just get dead air (no ring, no dial tone, 
or recording).  I can dial between phones that are connected to the 
asterisk box with no problem.

Asterisk version 1.2.9.1

In the following files, I changed the phone numbers/extensions and ip 
address.

sip.conf
[metaswitch]
type=friend
context=Internal
host=1.2.3.4
fromdomain=1.2.3.4
qualify=900
username=1234567890
canreinvite=no
allow=ulaw
dtmfmode=inband
jb-enable=yes
jb-max-size=300
nat=yes

[1234567890]
type=friend
username=1234567890
secret=1234
context=Internal
callerid="cisco 1" 1234567890
host=dynamic
canreinvite=no
mailbox=3000


extensions.conf
[Internal]
exten => 1234567890,1,Dial(SIP/1234567890,20)
exten => 1234567890,2,Voicemail(u3000)
exten => 1234567890,102,Voicemail(b3000)

exten => 798,1,Set,CALLERID(num)=1234567890
exten => 798,2,Dial(SIP/311 at metaswitch|20)
exten => 798,3,Congestion

exten => _XXXXXXX,1,Dial(SIP/${EXTEN}@66.170.55.210);
exten => _XXXXXXX,3,Congestion

exten => _1XXXXXXXXXX,1,Dial(SIP/${EXTEN}@66.170.55.210);
exten => _1XXXXXXXXXX,3,Congestion

exten => _X11,1,Dial(SIP/${EXTEN}@66.170.55.210);