[Freeswitch-users] Questions about att_xfer (freeswitch version 1.0 trunk 14633M)

[Anatoliy Kounitskiy]

Hello everybody!
I have few questions about the att_xfer application. First, what i want 
to accomplish is: user A calls user B, after that user B makes attended 
transfer to user C.
In the dialplan i have:

<context name="vpbx">
  <extension name="local_number">
...
      <action application="bind_meta_app" data="1 b s 
execute_extension::dx XML features"/>
      <action application="bind_meta_app" data="2 b s 
record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
      <action application="bind_meta_app" data="3 b s 
execute_extension::cf XML features"/>
      <action application="bind_meta_app" data="4 b s 
execute_extension::attented_xfer XML features"/>
 ....
    </condition>
  </extension>

So when user B answers the call, he sends *4 and the extensions for the 
attended transfer is started - the usual - plays message and read the 
input dtmf:

features.xml
...
    <extension name="attented_xfer">
      <condition field="${toll_allow}" expression="local"/>
      <condition field="destination_number" expression="^attented_xfer$">
        <action application="info"/>
        <action application="read" data="3 4 ivr/ivr-enter_ext.wav 
attxfer_callthis 30000 #"/>
        <action application="set" data="call_timeout=15"/>
        <action application="att_xfer" 
data="user/${attxfer_callthis}@${domain_name}"/>
      </condition>
    </extension>
...

To this problems everything is perfect. But here comes the questions, so 
if you can give some tips would be great.

1) when user B enters the extension number of C - the C's phone starts 
ringing in the tcpdump i can see that the phone is sending 180 ringing, 
BUT user B does not hear the ringing.
2) as mentioned in the 
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer
quote: "If the other leg is a voicemail or doesn't answered you can 
hangup that leg by pressing dtmf # (fixed in r14438) "
It doesn't work. The option 0 is working even before C answering the 
phone - after he answers it's a threeway conference :) - i like this 
feature.

I'm using FreeSWITCH Version 1.0.trunk (14633M)

Also I tried to set call timeout to see if I can go back the user A, who 
is listening to MOH - no luck here.

Probably I'm missing something. Tried to look in the source of att_xfer 
to understand why the feature i want is not working - but it seems my 
C/C++ skills are not so good, as i want :( .

Thank you in advance,
Anatoliy Kounitskiy