On Trixbox 1.1, get busy tone AS SOON As call is picked up

Hello,

Have setup Mutualphone as a Trunk on my Trixbox machine. Setup as an outbound route. The settings are taken as is from the mutualphone website. There is plenty of balance in the account. However, if I make a call, the call is setup properly... get the following in the AsteriskCLI:

[quote] -- Executing Macro("SIP/111-683a", "dialout-trunk|1|011234567||") in new stack
-- Executing GotoIf("SIP/111-683a", "1?3:2") in new stack
-- Goto (macro-dialout-trunk,s,3)
-- Executing Macro("SIP/111-683a", "user-callerid") in new stack
-- Executing GotoIf("SIP/111-683a", "0?report") in new stack
-- Executing GotoIf("SIP/111-683a", "0?start") in new stack
-- Executing Set("SIP/111-683a", "REALCALLERIDNUM=111") in new stack
-- Executing NoOp("SIP/111-683a", "REALCALLERIDNUM is 111") in new stack
-- Executing Set("SIP/111-683a", "AMPUSER=111") in new stack
-- Executing Set("SIP/111-683a", "AMPUSERCIDNAME=AAA-BBB") in new stack
-- Executing GotoIf("SIP/111-683a", "0?report") in new stack
-- Executing Set("SIP/111-683a", "CALLERID(all)=AAA-BBB <111>") in new stack
-- Executing NoOp("SIP/111-683a", "Using CallerID "AAA-BBB" <111>") in new stack
-- Executing Macro("SIP/111-683a", "record-enable|111|OUT") in new stack
-- Executing GotoIf("SIP/111-683a", "0 > 0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing AGI("SIP/111-683a", "recordingcheck|20060808-213718|1155062237.150") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20060808-213718|1155062237.150: Outbound recording not enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp("SIP/111-683a", "No recording needed") in new stack
-- Executing Macro("SIP/111-683a", "outbound-callerid|1") in new stack
-- Executing GotoIf("SIP/111-683a", "1?start") in new stack
-- Goto (macro-outbound-callerid,s,3)
-- Executing NoOp("SIP/111-683a", "REALCALLERIDNUM is 111") in new stack
-- Executing Set("SIP/111-683a", "USEROUTCID=") in new stack
-- Executing Set("SIP/111-683a", "EMERGENCYCID=") in new stack
-- Executing Set("SIP/111-683a", "TRUNKOUTCID=12125551234") in new stack
-- Executing GotoIf("SIP/111-683a", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,11)
-- Executing GotoIf("SIP/111-683a", "0?usercid") in new stack
-- Executing Set("SIP/111-683a", "CALLERID(all)=12125551234") in new stack
-- Executing GotoIf("SIP/111-683a", "1?report") in new stack
-- Goto (macro-outbound-callerid,s,15)
-- Executing NoOp("SIP/111-683a", "CallerID set to "" <12125551234>") in new stack
-- Executing Set("SIP/111-683a", "GROUP()=OUT_1") in new stack
-- Executing GotoIf("SIP/111-683a", "0?108") in new stack
-- Executing Set("SIP/111-683a", "DIAL_NUMBER=011234567") in new stack
-- Executing Set("SIP/111-683a", "DIAL_TRUNK=1") in new stack
-- Executing AGI("SIP/111-683a", "fixlocalprefix") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
fixlocalprefix: Could not parse /etc/asterisk/localprefixes.conf
-- AGI Script fixlocalprefix completed, returning 0
-- Executing Set("SIP/111-683a", "OUTNUM=011234567") in new stack
-- Executing Set("SIP/111-683a", "custom=SIP/Mutualphone") in new stack
-- Executing GotoIf("SIP/111-683a", "0?16") in new stack
-- Executing Dial("SIP/111-683a", "SIP/Mutualphone/011234567|120|Ttr") in new stack
-- Called Mutualphone/011234567
asterisk1*CLI> sip show channels
Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message
82.165.235.209 011234567 24178a02173 00103/00000 g729 No Tx: INVITE
192.168.1.100 111 c3ce8871-b9 00101/00102 ulaw No Rx: INVITE
2 active SIP channels
-- SIP/Mutualphone-2aff is ringing
-- SIP/Mutualphone-2aff is making progress passing it to SIP/111-683a
-- SIP/Mutualphone-2aff answered SIP/111-683a
asterisk1*CLI> sip show channels
Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message
82.165.235.209 011234567 24178a02173 00104/00000 unkn No (d) Tx: ACK
1 active SIP channel[/quote]

The called party's phone starts ringing... however, the MOMENT the phone is picked up, instantly, I hear a busy tone and the call gets dropped. This is happening systematically.

Also, I've got 2 x g729 liceses registered on the system, so its likely not a codec issue.

Thanks for any help.
Rizwan Sher