Problems with Asterisk - noob user

Greetings to all,
I need some help regarding the Asterisk installing procedure and configuration.
Before all data I should tell: this system was installed for traning purpose and learning
Asterisk server (good configuration)
Thomson ST2022 VOIP Phone
Laptop with software phone xlite 4
Switch - 4 ports
2. Instalation
Base system: CentOS 5.5
Asterisk 1.8 installed with svn.
3. Trouble: Registration from ‘sip:4321@;user=phone’ failed for ‘’ - No matching peer found — my Thomson phone is not connected. I cannot call from Laptop to the phone or reverse
I tryied almost everything without luck.
the sip.conf looks like this

cat /etc/asterisk/sip.conf
context=LocalSets ; default context for incoming calls
allowguest=yes ; disable unauthenticated calls
srvlookup=no ; enabled DNS SRV record lookup on outbound calls
udpbindaddr= ; listen for UDP requests on all interfaces
tcpenable=no ; disable TCP support

[ivan] ; create a template for our devices
secret=patrocle ; the channel driver will match on username first, IP second
context=LocalSets ; this is where calls from the device will enter the dialplan ; the device will register with asterisk
nat=yes ; assume device is behind NAT
; *** NAT stands for Network Address Translation, which allows
type=friend ; multiple internal devices to share an external IP address.
secret=patrocle ; a secure password for this device – DON’T USE THIS PASSWORD!
dtmfmode=auto ; accept touch-tones from the devices, negotiated automatically

;define a device name and use the office-phone template
00147fe1a092 ; Thomson ST2022 test phone

;define the softphone
001fe2123bb4 ; laptop with softphone

the extensions.conf looks something like this:
cat /etc/asterisk/extensions.conf
exten => 1234,1,Dial(SIP/test)
exten => 4321,1,Dial(SIP/ivan)

In peers I only see the IP of laptop.
How to register the phone and to be able to call from laptop???

Preferably: change 4321 to match the actual sip.conf section name.

Eaiser, but less secure: change the sip.conf section name to be 4321.

Hello all,
I used the trick from david55 (by the way: Thanks David55 for you info) and it work. I not very clear how the changing of the section name helped me but it works and now I’m able to continue.
For the moment I don’t have other info, but if some one want some info about I will provide it: phone reset codes, admin pages for the phone, asterisk configration.