[Freeswitch-users] auto dialing question ...

Fri Jan 23 15:55:06 PST 2009

Try it from the FS CLI first to see how to do it.

originate <dial string> <extension> <dialplan> <context>

so type this in the console replacing the sofia url with one of your choice:

originate sofia/default/100 at dom.com 9998 XML default

when you answer the call you should hear the tetris song.  so you can use
any ext at context+dialplan combo with those args.

You can do the same over xmlrpc or mod_event_socket

On Fri, Jan 23, 2009 at 5:39 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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>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
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>>
>> AIM: anthm
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>>
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>
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-- 
Anthony Minessale II

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