need help, service unavailable, registered but call does not get through

[tom <tomabroad@xxxxxxxxx>]

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!!


--- (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@xxxxxxxxxxxxxxx: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@xxxxxxxxxxxxxx>;tag=3c08d834
To: "702"<sip:702@xxxxxxxxxxxxxx>
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@xxxxxxxxxxx>
Content-Length: 0


<------------>
    -- Executing [702@from-internal:1] ResetCDR("SIP/701-0864f1b8", "") in new stack
    -- Executing [702@from-internal:2] NoCDR("SIP/701-0864f1b8", "") in new stack
    -- Executing [702@from-internal:3] Wait("SIP/701-0864f1b8", "1") in new stack
Retransmitting #1 (NAT) to 123.456.789.000:9855:
OPTIONS sip:701@xxxxxxxxxxxxxxx: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@xxxxxxxxxxx>;tag=as43db5836
To: <sip:701@xxxxxxxxxxxxxxx:37587;rinstance=9428b8620cd7a907>
Contact: <sip:Unknown@xxxxxxxxxxx>
Call-ID: 564e1f392a3b289f3b655a7200a67378@xxxxxxxxxxx
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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