I am trying to transfer a call using the excerpt of code below.  I am getting the  leg setup status ls_056, which is transfer failed, service unavailable or unsupported.  Why would my gateway not support transfer?  Is there something I have to do in order to be able to transfer a call on a gateway?  If you need to see more of my code just let me know.  Thanks for all of your help.
 
set callInfo(mode) "REDIRECT"
leg setup $destination callInfo leg_incoming
 
$destination is a 7 digit number that has a corresponding dial-peer on the gateway. 

Thanks for your help.
 
Let me start with what I am trying to do.  I am new to TCL so I started with app_debitcard.2.0.2.8.tcl as my base program.  Then I removed a few procedures to do what I need.  We use CUCM 6 and a user dials a specific number that goes to my gateway and hits my incoming dial-peer.  The program starts and asks the user to enter a 7 digit number starting with '72'.  The program then puts a '#' in front of the 7 digits and sends it back to CUCM.  From there it tries to connect to a meetme conference.  If the conference has not started, it plays a message then tries again in 30 seconds.  Once the caller is connected with the conference, I dont need the program anymore.
 
Right now, my program will connect and everything is good if our caller dials a different number that takes them out to PSTN then it passes back into the gateway.  The gateway then connects to the conference and everything works.  I am not sure what the program does after I connect.  However, with this scenario, we still pay for a call because we are using the PSTN.  We want to be able to dial straight to the gateway, hit the program, then hairpin back to CUCM.  The problem right now is that when the call connects using the hairpin, we do not get any audio.  I was just trying the transfer to see if that would make any difference.  Let me know if you need anything else.  Again, thanks for all of your help.
 
Here are the show call active voice brief
IP hairpin call-
1DA8 : 2167 4666625650ms.1 +10 pid:220077 Answer 40000 active
 dur 00:01:48 tx:235/37600 rx:374/59840
 IP 172.24.30.44:19128 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: off
 media inactive detected:n media contrl rcvd:n/a timestamp:n/a
 long duration call detected:n long duration call duration:n/a timestamp:n/a
 
1DA8 : 2170 4666633240ms.1 +60 pid:2200777 Originate #7240000 active
 dur 00:01:40 tx:0/0 rx:5031/804960
 IP 172.24.255.29:20028 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: off
 media inactive detected:n media contrl rcvd:n/a timestamp:n/a
 long duration call detected:n long duration call duration:n/a timestamp:n/a
 
 
PSTN call-
12A1 : 2171 4666812070ms.1 +10 pid:100 Answer 3094940000 active
 dur 00:00:14 tx:575/96600 rx:101/15842
 Tele 2/2:23 (2171) [2/5.3] tx:4950/2005/0ms g711ulaw noise:-51 acom:7  i/0:-45/-42 dBm
 
12A1 : 2174 4666821360ms.1 +60 pid:2200777 Originate #7240000 active
 dur 00:00:05 tx:101/15842 rx:266/42560
 IP 172.24.255.29:18984 SRTP: off rtt:0ms pl:3955/0ms lost:0/0/0 delay:65/65/65ms g711ulaw TextRelay: off
 media inactive detected:n media contrl rcvd:n/a timestamp:n/a
 long duration call detected:n long duration call duration:n/a timestamp:n/a

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I apologize for taking so long to reply.  I have been sidetracked doing other things.  I actually got the script working now.  I was looking through the other posts and after reading http://developer.cisco.com/web/vgapi/forums/-/message_boards/message/1426834 I decided to try a handoff.  I was thinking maybe the gateway was not fully letting go of the outbound call.  After I put in the command ¿handoff appl leg_all default¿, audio was coming across just fine.  Since then, I have been working hard documenting everything and adding small things to the program.  Thanks for everyone¿s input and help.