maybe we can make an origination_cancel_key=# you could set on the dial string to be able to cancel that originate with dtmf
On Wed, Aug 26, 2009 at 1:27 PM, Anatoliy Kounitskiy < [email protected]> wrote: > 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 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:[email protected] <msn%[email protected]> GTALK/JABBER/PAYPAL:[email protected]<paypal%[email protected]> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[email protected] <sip%[email protected]> iax:[email protected]/888 googletalk:[email protected]<googletalk%3aconf%[email protected]> pstn:213-799-1400
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