[Freeswitch-users] Questions about att_xfer

Anatoliy Kounitskiy anatoliy at kounitskiy.com
Wed Aug 26 13:10:59 PDT 2009 

Thank you for the fast patch :)

I'm going to test it right away and report you if there are problems or not.


On Wed, Aug 26, 2009 at 10:59 PM, Anthony
Minessale<anthony.minessale at gmail.com> wrote:
> i added a patch to attempt to do this so try adding
>
> {origination_cancel_key=#} before the dial string
> or
> <action application="set" data="origination_cancel_key=#"/>
> before you bridge.
>
> On Wed, Aug 26, 2009 at 1:39 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> 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
>> <anatoliy at kounitskiy.com> 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<anatoliy at kounitskiy.com> 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/${attxfer_callthis}@${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
>>> >
>>>
>>>
>>>
>>> --
>>> Anatoliy Kounitskiy
>>> -------------------------
>>> E-mail: anatoliy at kounitskiy.com
>>> Mobile: +359898913540
>>>
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>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
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>>
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>>
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>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
>
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>



-- 
Anatoliy Kounitskiy
-------------------------
E-mail: anatoliy at kounitskiy.com
Mobile: +359898913540

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