Newb: Getting dial tone with IP Phone and sip.conf

As I understand it, I should be able to get a dial tone if I configure my IP phone to connect to a computer running asterisk and the user names/authentication are setup properly on both the phone and asterisk (via sip.conf). However, I’ve been unable to get a dial tone with my phone connected to a laptop running asterisk.

The sip.conf has the following:

[mytest] username=mytest secret=mytest type=friend host= context=default

I also have the following lines in the extensions.conf:

[default] exten => 1001,1,Dial(SIP/mytest,20) exten => 1001,n,Hangup()

Following the directions, I found here I set the following configuration on my phone:

[code]DHCP: Disabled Current IP:
Host Name: Domain:
Current Netmask: Current Gateway:
Primary DNS:

SIP Settings
SIP Port: 5060

Proxy and Registration
Proxy: Register: no
Make Call Without Reg: no Register Expires:
Ans Call Without Reg: no

Subscriber Information
Display Name: mytest User ID: mytest
Password: mytest Use Auth ID: no
Auth ID: mytest[/code]

This is what I saw in the /var/log/asterisk/messages file:

ERROR[2896] chan_sip.c: Peer 'mytest' is trying to register, but not configured as host=dynamic NOTICE[2896] chan_sip.c: Registration from '"mytest" <sip:mytest@>' failed for '' - Peer is not supposed to register
But changing the Register field to ‘no’ on the phone doesn’t seem to make any difference. Neither does changing the ‘host’ line to ‘dynamic’ in sip.conf.

My environment:
RedHat Linux 5.1, kernel 2.6.18
Phone: Linksys SPA921
For the network connection I’m using a USB ethernet adapter. Linux has a compatibility issue with laptops ethernet card. I can ping the phone from my laptop so the two should be able to reach other.

any ideas what might be going on?

After some more fiddling, I finally got a dial tone. So never mind. If an admin wants to delete or close this topic, that’s fine.