asterisk users - Dec 2003 - Cisco and Asterisk 2621

Ok here is a question that has gotten me stumped.  I have an Asterisk system up
and running.  I need to connect it via the Internet to a Sip Cisco system.  This
is what they have.  I have there IP address's and login. X-lite is able to
connect to them and make a call! So I have the name right!

CISCO router model: 2621 
VoIP module: NM-HDA-4FXS

I have done Google lookup and at the Wiki about this.  What I did get is the
following from them.  Following in the SIP.CONF file.

register => name@217.XXX.XXX.XXX:5060

This does not seem to work.

I have also tried at the extensions.conf a setting of.

exten => 380,1,Dial(SIP/name@217.XXX.XXX.XXX)

I feel I have missed something some place or I just don't understand what to
do!
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I have a 2621 working with asterisk. See below:

sip.conf
=====[general]
port = 5060                     ; Port to bind to
bindaddr = 0.0.0.0              ; Address to bind to
context = default               ; Default for incoming calls

[cisco] ; Cisco 2621 Router
type=friend
canreinvite=no
insecure=yes
host=192.168.62.1			; address of the cisco router
dtmfmode=inband
context=default

extensions.conf
==============
; My asterisk numbers are 600-699 (omitted from example)

; Send all calls prefixed with 9 to the cisco
exten => _9.,1,Dial,sip/BYEXTENSION@cisco

relevant part of cisco configuration
===================================[c2600-is-mz.122-13.T.bin]
!
dial-peer voice 6 pots
 description Incoming Call from PSTN to number 6xx
 application session
 incoming called-number 6..
 destination-pattern 6..
 no digit-strip
 direct-inward-dial
 port 1/0:15
!
dial-peer voice 600 voip
 description Outgoing call to Asterisk Server for numbers 6xx
 application session
 destination-pattern 6..
 session protocol sipv2
 session target ipv4:192.168.62.60
 session transport udp
 dtmf-relay rtp-nte
 codec g711ulaw
!         
dial-peer voice 9 voip
 description Incoming Call from Asterisk Server to number beginning with 9
 application session
 incoming called-number 9T
 dtmf-relay rtp-nte
 codec g711ulaw
!
dial-peer voice 900 pots
 description Outgoing call to PSTN for numbers beginning with 9
 application session
 destination-pattern 9T
 no digit-strip
 port 1/0:15
!         
-----Original Message-----
From: Ariel Batista [mailto:abatista@avionica.com]
Sent: 03 December 2003 17:06
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Cisco and Asterisk 2621


Ok here is a question that has gotten me stumped.  I have an Asterisk system
up and running.  I need to connect it via the Internet to a Sip Cisco
system.  This is what they have.  I have there IP address's and login.
X-lite is able to connect to them and make a call! So I have the name right!

CISCO router model: 2621 
VoIP module: NM-HDA-4FXS

I have done Google lookup and at the Wiki about this.  What I did get is the
following from them.  Following in the SIP.CONF file. 

register => name@217.XXX.XXX.XXX:5060

This does not seem to work.

I have also tried at the extensions.conf a setting of.

exten => 380,1,Dial(SIP/name@217.XXX.XXX.XXX)

I feel I have missed something some place or I just don't understand what to
do!