[Freeswitch-users] newbie alert. Help with dialplan

[Julian Lyndon-Smith]

Thanks Michael. What the extension currently does is :

A) Add 1 to the current list of Queue calls (we have a blended system
where I need to dynamically allocate agents from outbound to inbound)
This counter is maintained in the phone system
B) call my application via a web call (CURL) to tell it that a new
Queue call has come in. This web call returns various bits of
information regarding the queue (message to play to the caller,
message to play to the agent, is the queue open etc)
C) if this fails (bad queue) then drop out
D) start recording, play the welcome message to the caller
E) Ask them if they want to participate in an automated questionnaire
at the end of the call.
F) If they agree, play a thank you message
G) If the call is out of hours, go to voicemail
H) Send a message to the application that a call is about to enter the queue
I) Join the queue
J) if the call is dropped out of the queue because no-one answered,
record that status
K) Stop recording

If they have answered yes to the questionnaire, then redirect the call
to an IVR when the agent hangs up

When the agent hangs up, or when the caller hangs up, then a jabber
message is sent to the agent, and a web service is called to indicate
the end of the call


2009/8/28 Michael Collins <msc at freeswitch.org>:
> Julian,
>
> It might be better if you gave us the run-down on what your current Asterisk
> extension does. Most likely there is an elegant way of doing it in
> FreeSWITCH. Part of the unlearing of the Asterisk way is coming to grips
> with the fact that the FS dialplan is not a programming language in and of
> itself. While it's convenient to have GotoIf's all over the place, in the
> long run that's inefficient.
>
> Let's start from the top-down instead of the bottom-up. Discussion your
> application in general terms, i.e., non-Asterisk-specific language. Tell us
> what it does, not how it does it, and then we can think it through using FS
> concepts.
>
> -MC
>
> On Thu, Aug 27, 2009 at 11:16 AM, Julian Lyndon-Smith <asterisk at dotr.com>
> wrote:
>>
>> I am a long-time asterisk user (2005), so I come from an unfortunate
>> position of having to unlearn an awful lot of stuff in order to make
>> freeswitch do the things I want ;(
>>
>> I *think* i've got my head around using mod_curl_xml (?) to read all
>> the config stuff from my webserver. I *think* I've got my head around
>> setting up the sip clients etc etc ...
>>
>> However, where I am really struggling is the dialplan. For the life of
>> me I simply cannot seem to grasp the fs way - that's no disrespect to
>> the fs way, but perhaps the failure of this old brain to change !
>>
>>  have pastebinned an example of show 1234546 at inboundq at
>> http://www.pastebin.ca/1544890. If possible, would someone be able to
>> show me how to convert this dialplan to the fs way ? If I could be
>> given a little foot-up I'm sure that I would be able to convert the
>> rest of the dialplan !
>>
>> Thanks in advance, and please go easy !
>>
>> Julian
>>
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