[Freeswitch-users] auto dialing question ...

Shelby Ramsey sicfslist at gmail.com
Fri Jan 23 16:20:21 PST 2009 

Thanks Michael.
Here are some really simple examples (we limit what people do through the
web ... nothing really fancy ... just some good old fashion robo calls).
ultimately today it looks like this extensions_table (context, exten,
priority, app, appdata):

campaign,100,1,ANSWER() --> answers call
campaign,100,2,WAIT(1) --> this is a pause
campaign,100,3,PLAYBACK($INTROFILE) --> plays intro file
campaign,100,4,BACKGROUND($MESSAGE) --> plays message
campaign,100,5,WAITEXTEN(5) --> WAIT for DTMF
campaign,100,6,HANGUP()
Here are DTMF options
campaign,1,SYSTEM($SCRIPT campaign $EXTEN)
campaign,2,SYSTEM($SCRIPT campaign $EXTEN)
campaign,3,DIAL,zap/g1/$ANI

anothercampaign,112,1,ANSWER()
anothercampaign,112,2,BACKGROUND($SOMEFILE)
anothercampaign,112,3,WAITEXTEN(5)
anothercampaign,112,4,HANGUP()
Options:
anothercampaign,1,DIAL(sip/$ANI@$IP)
anothercampaign,2,PLAYBACK($GOODBYE)

And they pretty much all look like that ... it's easy to return the stuff
for extension 100 via XML ... but the challenge is the DTMF options
(relating to the same context) ... or maybe I'm just missing something
(which is a definite possibility).  We don't ever do anything complicated in
the IVRs (TTS or ASR) but there is just a lot of them that all get
controlled, manipulated via a web interface.

I can originate a call ... do all of that ... just trying to figure out a
"simple" way to return the above via xml.

Thanks for your help Anthony and Michael as always!

Shelby





On Fri, Jan 23, 2009 at 5:55 PM, Michael Collins <msc at freeswitch.org> wrote:

> I see your dilemma. To keep things dynamic you definitely want to use
> your XML_CURL stuff. Like you said, nothing static.
>
> Can you post a sample call-flow in plain English? I'm curious about
> something. I don't want to say anything else. Just post a simple call
> flow:
> Initiate call
> Wait for answer/busy/timeout/invalid
> On busy/timeout/invalid: update db and move on
> On answer play greeting, accept digit, route based on digit
>
> Something like that would help me conceptualize what you are trying to do.
>
> Thanks
> MC
>
> P.S. - I've done a little bit of outbound IVR calling so hopefully I
> can assist you.
>
> On Fri, Jan 23, 2009 at 3:39 PM, Shelby Ramsey <sicfslist at gmail.com>
> wrote:
> > Anthony / Michael,
> > Thanks for the quick responses.  What I don't want to do is "drive the
> call"
> > (by that listen on a socket ... do this on this event ... or anything
> else
> > that my very limited FS foo would break) ... Just want to start it and
> then
> > give it instructions on where to go.
> > So I guess a better question would be ... how do I give directions to FS
> for
> > this (and I get the 1st part ... that's obvious ... really lost on the
> DTMF
> > digit part) ... and please keep in mind we're talking hundreds of
> extensions
> > / IVR's and distributed machines so I can't have any dependancy on static
> > conf files other than maybe something like what Michael mentioned where I
> > point every call to something:
> > [campaign]
> > exten => 100,1,ANSWER()
> > exten => 100,n,PLAYBACK(somefile)
> > exten => 100,n,BACKGROUND(somefile)
> > exten => 100,n,WAITEXTEN(4)
> > exten => 100,n,HANGUP()
> > but in that same context is someone triggers DTMF:
> > exten => 1,1,DOSOMETHING
> > exten => 2,1,DOSOMETHING
> > I was imaging issuing originate via XML_RPC ... something like originate
> > sofia/$ANI@$IP $SOMEEXTEN then on answer when FS tries to connect to
> > $SOMEEXTEN it will ask me what to do via xml_curl ... where I would
> normally
> > respond with something like this:
> > <?xml version="1.0" encoding="UTF-8" standalone="no"?> <document
> > type="freeswitch/xml"> <section name="dialplan" description="FS
> RESPONSE">
> > <context name="public"> <extension name="$EXTEN"> <condition
> > field="destination_number" expression=""> <action application="set"
> > data="hangup_after_bridge=true"/> <action application="set"
> > data="continue_on_fail=true"/> <action application="set"
> > data="call_timeout=180"/> <action application="set"
> > data="proxy_media=true"/> <action application="set"
> > data="pass_rfc2833=true"/> <action application="set"
> > data="accountcode=$CUSTOMER" /> <action application="set"
> > data="origination_caller_id_name=NULL" /> <action application="set"
> > data="origination_caller_id_number=$CIDNUM" /> <action application="set"
> > data="effective_caller_id_name=NULL" /> <action application="set"
> > data="effective_caller_id_number=$CIDNUM" /> <action application="set"
> > data="userfield=$BUNCHOFCRAPFORMYCDR" /> <action application="bridge"
> > data="sofia/external/$ANI@$PROVIDERIP" /> </condition> </extension>
> > </context>
> > </section> </document>
> > The challenge I've got is I have no idea how to do stuff like the IVR
> > mentioned above (the playback part is easy) ... but I can't grasp
> > conceptually how to get the "context" with "multiple extensions" part
> back
> > to FS via this method (is it possible?)...
> > Sorry for what is probably a very simple answer and any AST references
> (but
> > I've been using it in heavy production environments for about 5 years).
> Just
> > trying to "port" what I do today without making my brain melt out of my
> ears
> > (and it doesn't take much for that to happen).
> > Shelby
> > PS ... Really enjoy the list. I usually fall out of my chair laughing
> once a
> > day from your remarks Anthony. Keep it coming!
> >
> > On Fri, Jan 23, 2009 at 4:55 PM, Anthony Minessale
> > <anthony.minessale at gmail.com> wrote:
> >>
> >> Does AST mean Asterisk Open Source PBX ?
> >>
> >> If so, then yes I am familiar with it's archetechure as I am a former
> >> developer from that project.
> >>
> >> You have 3 choices with FreeSWITCH
> >>
> >> 1) You can open a dedicated connection to mod_event_socket or XMLRPC per
> >> call and issue the originate command from there:
> >>     This will block until you know for sure the outcome of the attempt.
> >> If it's success it will give you the uuid if not it gives you the cause
> >> code.
> >>
> >> 2) You can use a single mod_event_socket or XMLRPC connection to send
> all
> >> calls but use the bgapi mechanism which will do the same as above
> >>     only asynchronously, The command will return immediately and the
> >> result will be fired as an event that you can pick up on the same or
> >> different event_socket connection or
> >>     other event consumer such as a custom C,perl,lua etc module.
> >>
> >> 3) You can use mod_xml_cdr to generate detailed 1-file-per-leg call
> files
> >> that will tell you when where and why the calls failed or did not fail.
> >>
> >>
> >> On Fri, Jan 23, 2009 at 4:39 PM, Michael Collins <msc at freeswitch.org>
> >> wrote:
> >>>
> >>> On Fri, Jan 23, 2009 at 2:15 PM, Shelby Ramsey <sicfslist at gmail.com>
> >>> wrote:
> >>> > Sorry for the double post ... actually hit send too early ...
> >>> > OK ... Here goes another I'm doing this with AST  ... but I want to
> >>> > move it
> >>> > to FS.  Searched via google site:lists.freeswitch.org auto dialer
> and
> >>> > others
> >>> > ... nothing useful.
> >>> > Today I have a platform for auto dialing with AST (centrally managed
> >>> > ...
> >>> > about 10 machines) and we do this:
> >>> >   -- Remote machines query central DB for numbers to call based on
> >>> > certain
> >>> > configs
> >>> >   -- Use AMI to generate the call
> >>> >   -- If call gets answered, extension info queried via rta (central
> db
> >>> > again)
> >>> > The nice thing about all of this is it's relatively easy to manage
> >>> > (through
> >>> > one central web interface we built) and it works ... the bad part is
> >>> > reporting ...
> >>> > So ... conceptually I'm trying to accomplish the same thing ...
> >>> > Today we use FS a lot for termination of VoIP traffic ... all done
> via
> >>> > XML_CURL ... which is awesome  (not to xml cdr ... and the "proxying"
> >>> > of
> >>> > media) ...
> >>> > Would like to do something like:
> >>> >   -- originate request (looks simple enough)
> >>> >   -- on answer XML_CURL posts info
> >>>
> >>> Several choices, depending upon how much you want it handled inside
> >>> the dialplan vs. handled in the scripting language. For the sake of
> >>> testing you could do something like this:
> >>> <extension name="ivr-start">
> >>>  <condition field="destination_number" expression="ivr_whatever">
> >>>    <action application="set" data="execute_on_answer=transfer
> >>> IVR_ANSWER XML default"/>
> >>>    <!-- rest of dialplan -->
> >>>  </condition>
> >>> </extension>
> >>>
> >>> Then have:
> >>> <extension name="ivr-answer">
> >>>  <condition field="destination_number" expression="IVR_ANSWER">
> >>>    <action application="lua" data="post-info.lua
> >>> ${some_important_value}"/>
> >>>  </condition>
> >>> </extension>
> >>>
> >>> This would have any answered call go to the "ivr-answer" extension
> >>> while unanswered calls could stay in the ivr-start extension to get
> >>> properly handled. (Busy, no answer, invalid/SIT, etc.)
> >>>
> >>> You could then have the "ivr-answer" extension do whatever is
> >>> appropriate, like listen for digits, play announcement, beg for money,
> >>> etc. :)
> >>>
> >>> -MC
> >>>
> >>> > But for the life of me I can't figure out how to translate this into
> >>> > the xml
> >>> > response ...
> >>> > [campaign]
> >>> > exten => 100,1,ANSWER()
> >>> > exten => 100,n,WAIT(2)
> >>> > exten => 100,n,BACKGROUND(${SOUND_DIR}/somefile)
> >>> > exten => 100,n,WAITEXTEN(10)
> >>> > exten => 100,n,HANGUP()
> >>> > exten => 1,1,PLAYBACK(goodbye)
> >>> > .... and so on ...
> >>> > I've looked at the ivr.conf stuff but it's all static and all of this
> >>> > has to
> >>> > be manageable via a web interface .... meaning dumping into a DB and
> >>> > returning an XML response seems reasonable ... but trying to stick or
> >>> > modify
> >>> > static text files from the web interface is too much text parsing and
> >>> > bad
> >>> > things will happen ...
> >>> > Any thoughts or pointing me in the right direction would be
> >>> > appreciated.
> >>> > Shelby
> >>> >
> >>> >
> >>> >
> >>> > _______________________________________________
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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/
> >>
> >> AIM: anthm
> >> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> IRC: irc.freenode.net #freeswitch
> >>
> >> FreeSWITCH Developer Conference
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> >> pstn:213-799-1400
> >>
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