sip trace of talkshoe connection

In extensions.conf:

exten => 731,1,Set(CALLERID(num)=123)
exten => 731,n,Dial(SIP/123@66.212.134.192,60,)

Asterisk 1.2.18
Same happens in 1.4.4


   -- Starting simple switch on 'Zap/3-1'
    -- Executing Set("Zap/3-1", "CALLERID(num)=123") in new stack
    -- Executing Dial("Zap/3-1", "SIP/123@66.212.134.192|60|D(22622#1112223334#)") in new stack
We're at 192.168.67.18 port 11210
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 12 lines
Reliably Transmitting (no NAT) to 66.212.134.192:5060:
INVITE sip:123@66.212.134.192 SIP/2.0
Via: SIP/2.0/UDP 192.168.67.18:5060;branch=z9hG4bK1a7d2f9c;rport
From: "Channel 3" <sip:123@192.168.67.18>;tag=as7b39bbfa
To: <sip:123@66.212.134.192>
Contact: <sip:123@192.168.67.18>
Call-ID: 39962e1c067c58876c5af1240a19b112@192.168.67.18
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 22 Jun 2007 17:53:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 14976 14976 IN IP4 192.168.67.18
s=session
c=IN IP4 192.168.67.18
t=0 0
m=audio 11210 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
    -- Called 123@66.212.134.192
pungenday*CLI> 
<-- SIP read from 66.212.134.192:5060: 
SIP/2.0 180 Ringing
From: "Channel 3"<sip:123@192.168.67.18>;tag=as7b39bbfa
To: <sip:123@66.212.134.192>;tag=a000217-13c4-467bc56e-13c8fe99-587d9c53
Call-ID: 39962e1c067c58876c5af1240a19b112@192.168.67.18
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.67.18:5060;received=212.179.75.202;rport=5060;branch=z9hG4bK1a7d2f9c
Contact: <sip:123@66.212.134.192>
Content-Length: 0


--- (8 headers 0 lines) ---
    -- SIP/66.212.134.192-081b5458 is ringing
pungenday*CLI> 
<-- SIP read from 66.212.134.192:5060: 
SIP/2.0 403 Forbidden
From: "Channel 3"<sip:123@192.168.67.18>;tag=as7b39bbfa
To: <sip:123@66.212.134.192>;tag=a000217-13c4-467bc56e-13c8fe99-587d9c53
Call-ID: 39962e1c067c58876c5af1240a19b112@192.168.67.18
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.67.18:5060;received=212.179.75.202;rport=5060;branch=z9hG4bK1a7d2f9c
Content-Length: 0


--- (7 headers 0 lines) ---
Transmitting (no NAT) to 66.212.134.192:5060:
ACK sip:123@66.212.134.192 SIP/2.0
Via: SIP/2.0/UDP 192.168.67.18:5060;branch=z9hG4bK1a7d2f9c;rport
From: "Channel 3" <sip:123@192.168.67.18>;tag=as7b39bbfa
To: <sip:123@66.212.134.192>;tag=a000217-13c4-467bc56e-13c8fe99-587d9c53
Contact: <sip:123@192.168.67.18>
Call-ID: 39962e1c067c58876c5af1240a19b112@192.168.67.18
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Jun 22 20:53:19 WARNING[14990]: chan_sip.c:9847 handle_response_invite: Forbidden - wrong password on authentication for INVITE to '"Channel 3" <sip:123@192.168.67.18>;tag=as7b39bbfa'
    -- SIP/66.212.134.192-081b5458 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
  == Auto fallthrough, channel 'Zap/3-1' status is 'CONGESTION'
Destroying call '39962e1c067c58876c5af1240a19b112@192.168.67.18'
pungenday*CLI> sip no debug 
SIP Debugging Disabled
    -- Hungup 'Zap/3-1'