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asterisk at dotr.com Guest
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Posted: Thu Aug 27, 2009 1:20 pm Post subject: [Freeswitch-users] newbie alert. Help with dialplan |
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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@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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msc at freeswitch.org Guest
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Posted: Fri Aug 28, 2009 12:34 pm Post subject: [Freeswitch-users] newbie alert. Help with dialplan |
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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@dotr.com (asterisk@dotr.com)> wrote:
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asterisk at dotr.com Guest
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Posted: Fri Aug 28, 2009 3:16 pm Post subject: [Freeswitch-users] newbie alert. Help with dialplan |
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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@freeswitch.org>:
Quote: | 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@dotr.com>
wrote:
Quote: |
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@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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FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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