asterisk users - Jul 2009 - need help, service unavailable, registered but call does not get through

hi, i have a new install, 1.6 2, 2 extension, but the call doesnt get
thorugh: here is my sip debug outout: thx for ur help!!
<asterisk-users at lists.digium.com>

--- (13 headers 16 lines) ---
Sending to AA.BBB.CCC.DD : 28127 (NAT)
Using INVITE request as basis request -
Y2QxNTg4NjE3MTZjNGMzZGM5NzE3YWY4NjAyOTYzMjk.
Found user '701' for '701'
Found RTP audio format 107
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 192.168.0.73:40958
Found unknown media description format BV32 for ID 107
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10c
(ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc
(ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.0.73:40958
Looking for 702 in from-internal (domain ABC.dyndns.org)
list_route: hop: <sip:701 at 123.456.789.000:37587>
acerdebian*CLI>
<--- Transmitting (NAT) to 123.456.789.000:28127 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.73:15158
;branch=z9hG4bK-d8754z-0a540c5d3439c271-1---d8754z-;received=123.456.789.000;rport=28127
From: "me"<sip:701 at ABC.dyndns.org <sip%3A701 at
ABC.dyndns.org>>;tag=3c08d834
To: "702"<sip:702 at ABC.dyndns.org <sip%3A702 at
ABC.dyndns.org>>
Call-ID: Y2QxNTg4NjE3MTZjNGMzZGM5NzE3YWY4NjAyOTYzMjk.
CSeq: 2 INVITE
User-Agent: Asterisk PBX 1.6.0.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:702 at 192.168.1.4 <sip%3A702 at 192.168.1.4>>
Content-Length: 0


<------------>
    -- Executing [702 at from-internal:1] ResetCDR("SIP/701-0864f1b8",
"") in
new stack
    -- Executing [702 at from-internal:2] NoCDR("SIP/701-0864f1b8",
"") in new
stack
    -- Executing [702 at from-internal:3] Wait("SIP/701-0864f1b8",
"1") in new
stack
Retransmitting #1 (NAT) to 123.456.789.000:9855:
OPTIONS sip:701 at 123.456.789.000:37587;rinstance=9428b8620cd7a907 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.4:5060;branch=z9hG4bK782c5851;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown at 192.168.1.4 <sip%3AUnknown at
192.168.1.4>
>;tag=as43db5836
To: <sip:701 at 123.456.789.000:37587;rinstance=9428b8620cd7a907>
Contact: <sip:Unknown at 192.168.1.4 <sip%3AUnknown at 192.168.1.4>>
Call-ID: 564e1f392a3b289f3b655a7200a67378 at 192.168.1.4
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.0.6
Date: Thu, 02 Jul 2009 18:51:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0
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Hello
I have 2 queues (queue_1 and queue_2 ) in my Asterisk, and I want to 
send 2/3 of the calls to queue_1 and 1/3 of the calls to queue_2
How can I do that load balancing in extensions.conf?

I have something like this:
exten => 123,1,Ringing
exten => 123,2,Wait(1)
exten => 123,3,Answer

;  2 in 3 calls go to queue_1
exten => 123,x,Queue(queue_1)

; 1 in 3 calls go to queue_2
exten => 123,x,Queue(queue_2)

But how can I configure this call distribution?
Thanks
Regards
Joao Pereira

-- 
StarTel - A Rede Livre
Joao Gomes Pereira
www.startel.pt
+351 304500650
sip: gomespereira at startel.pt