asterisk users - Sep 2003 - Grandstream and G729

Hello,

I want to buy Grundstream sip phones.
Could someone tell me if thay are working with Asterisk ?

I want to buy also codec G729. How many channels I should buy if I have 24 ext.
Let's say if I buy 20 and there is 20 calls the other calls with use another
codec then ?

Reagds,

-- Bart
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I am an * newbie and was hoping that someone would be able to help me out.
 
I am having trouble getting SJ Softphone registered with *.
 
My sip.conf file indicates:
 
[phone1]
type=friend
username=keith
secret=keith
host=dynamic
qualify=2000
disallow=g729
auth=md5
context=sip
mailbox=9999
callerid="keith@10.1.1.12" <1000>
 
 
But the log in SJPhone indicates that the registration is being rejected:
 
2003-09-25 18:55:34.776 UDP LOCAL->10.1.1.12:5060
REGISTER sip:10.1.1.12 SIP/2.0
Content-Length: 0
Contact: <sip:10.1.1.62>
Call-ID: FF0AF646-C45F-49CB-A8BE-EF3D3D7BF8BC@10.1.1.62
From: <sip:keith@10.1.1.12>;tag=12629129
CSeq: 25 REGISTER
To: <sip:keith@10.1.1.12>
Via: SIP/2.0/UDP 10.1.1.62:5060
 

2003-09-25 18:55:34.786 UDP 10.1.1.12:5060->LOCAL
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.1.1.62:5060
From: <sip:keith@10.1.1.12>;tag=12629129
To: <sip:keith@10.1.1.12>;tag=as1fc64603
Call-ID:  <mailto:FF0AF646-C45F-49CB-A8BE-EF3D3D7BF8BC@10.1.1.62>
FF0AF646-C45F-49CB-A8BE-EF3D3D7BF8BC@10.1.1.62
CSeq: 25 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:keith@10.1.1.12>
Content-Length: 0

 
In SJ Phone I have the proxy ip address set to the local ip address of the *
server (no NAT is involved here...everything is local on a lan).   I have
the callerid in SJ Phone set to sip:keith@10.1.1.12 along with the
authentication section filled out with the respective username and password.
 
Any pointers??
 
Thanks
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