-- Accepting AUTHENTICATED call from 173.248.231.82:
       > requested format = ilbc,
       > requested prefs = (),
       > actual format = alaw,
       > host prefs = (alaw|ulaw|g729),
       > priority = mine

    -- Executing [3429121747@numberplan-custom-2:1] NoOp("IAX2/809-619", "") in new stack

    -- Executing [3429121747@numberplan-custom-2:2] Macro("IAX2/809-619", "trunkdial-failover|SIP/trunk_3/429121747|") in new stack

    -- Executing [s@macro-trunkdial-failover:1] Dial("IAX2/809-619", "SIP/trunk_3/429121747||r") in new stack

Audio is at 192.168.1.100 port 10046

Adding codec 0x8 (alaw) to SDP

Adding codec 0x4 (ulaw) to SDP

Adding non-codec 0x1 (telephone-event) to SDP

Reliably Transmitting (NAT) to 203.161.160.69:5060:
INVITE sip:429121747@byo.engin.com.au SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK052b56c3;rport
From: "" <sip:0742221141@byo.engin.com.au>;tag=as26ac20d0
To: <sip:429121747@byo.engin.com.au>
Contact: <sip:0742221141@192.168.1.100>
Call-ID: 208d160a2d69f225345890402be024dd@byo.engin.com.au
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "" <sip:@byo.engin.com.au>
Date: Tue, 17 May 2011 01:43:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 266

v=0
o=root 27830 27830 IN IP4 192.168.1.100
s=session
c=IN IP4 192.168.1.100
t=0 0
m=audio 10046 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

    -- Called trunk_3/429121747

IPPBX*CLI>
<--- SIP read from 203.161.160.69:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.100:5060;received=124.150.102.25;branch=z9hG4bK052b56c3;rport=5060
From: "" <sip:0742221141@byo.engin.com.au>;tag=as26ac20d0
To: <sip:429121747@byo.engin.com.au>
Call-ID: 208d160a2d69f225345890402be024dd@byo.engin.com.au
CSeq: 102 INVITE


<------------->
--- (6 headers 0 lines) ---

<--- SIP read from 203.161.160.69:5060 --->
SIP/2.0 604 Does not exist anywhere
Via: SIP/2.0/UDP 192.168.1.100:5060;received=124.150.102.25;branch=z9hG4bK052b56c3;rport=5060
From: "" <sip:0742221141@byo.engin.com.au>;tag=as26ac20d0
To: <sip:429121747@byo.engin.com.au>;tag=SD0d7ff99-66111699-1305596615948
Call-ID: 208d160a2d69f225345890402be024dd@byo.engin.com.au
CSeq: 102 INVITE
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---
    -- Got SIP response 604 "Does not exist anywhere" back from 203.161.160.69
Transmitting (NAT) to 203.161.160.69:5060:
ACK sip:429121747@byo.engin.com.au SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK052b56c3;rport
From: "" <sip:0742221141@byo.engin.com.au>;tag=as26ac20d0
To: <sip:429121747@byo.engin.com.au>;tag=SD0d7ff99-66111699-1305596615948
Contact: <sip:0742221141@192.168.1.100>
Call-ID: 208d160a2d69f225345890402be024dd@byo.engin.com.au
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "" <sip:@byo.engin.com.au>
Content-Length: 0


---

  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [s@macro-trunkdial-failover:2] GotoIf("IAX2/809-619", "0>0?s-CHANUNAVAIL|1:s-out|1") in new stack
    -- Goto (macro-trunkdial-failover,s-out,1)
    -- Executing [s-out@macro-trunkdial-failover:1] Hangup("IAX2/809-619", "") in new stack
  == Spawn extension (macro-trunkdial-failover, s-out, 1) exited non-zero on 'IAX2/809-619'
    -- Hungup 'IAX2/809-619'
Really destroying SIP dialog '208d160a2d69f225345890402be024dd@byo.engin.com.au' Method: INVITE

Really destroying SIP dialog '2dbd07086e6cad6e7ea3cd147798169d@127.0.0.1' Method: REGISTER

IPPBX*CLI>