Trouble transfering calls

Version 1
    This document was generated from CDN thread

    Created by: Aaron Birk on 12-08-2009 02:53:57 PM
    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. 

    Subject: RE: Trouble transfering calls
    Replied by: VijayPrasad Neelamegam on 13-08-2009 11:43:05 AM
    Hi Aaron,
     
    Rediect mode just transfers the leg id to the destination number.You have mentioned ls_056 occured while transfering,are you trying to do a consult tranfer or a blind transfer,and what protocol you are using H323 or SIP.Have you captured transfer staus in the script?
     
    Can you pass me the script ?
     
    Thanks
    Vijay

    Subject: RE: Trouble transfering calls
    Replied by: Aaron Birk on 13-08-2009 03:57:23 PM
    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

    Subject: RE: Trouble transfering calls
    Replied by: VijayPrasad Neelamegam on 14-08-2009 12:30:19 PM
    Thanks for the script and information,
     
    I understand that you are trying to make a hairpin call.I request you to try the following steps
     
    set callInfo(mode)        ROTARY
    set callInfo(rerouteMode) REDIRECT_ROTARY
     
    and knidly let me know the results.
     
    Thanks
    Vijay

    Subject: RE: Trouble transfering calls
    Replied by: Aaron Birk on 08-09-2009 08:56:43 PM
    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.