After several hours of testing I was able to answer myself the previous mentioned questions.
It appears that # and the 0 option work _only_ if user C has answered the call OR voicemail system answers it. user A ---call---> user B----attended xfer---> user C At this point I have new question. In example user C does not have a voicemail and the call timeout is not an option to wait for. How can user B go back to the user A, who is listening to MOH? Could someone help me with an advice/tip? At the moment I have just one idea for accomplishing it: 1) try to use bind_meta_app in the extension with the att_xfer (not sure if it can be done). To have a key feature that takes the user A call leg id and bridging it with user B Thank you in advnace, Anatoliy Kounitskiy On Wed, Aug 26, 2009 at 5:51 PM, Anatoliy Kounitskiy<[email protected]> wrote: > 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/${attxfer_callth...@${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 > -- Anatoliy Kounitskiy ------------------------- E-mail: [email protected] Mobile: +359898913540 _______________________________________________ FreeSWITCH-users mailing list [email protected] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
