Re: xAP Asterisk integration

[Gregg Liming <gregg@xxxxxxxxxxx>]

James wrote:

>Gregg Liming wrote:
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>>I know how to do this on the Asterisk side using a manager
interface
>>app.  The issue becomes getting the info from a sender.  You
mention
>>xAPSwitchBoard; does it send xAP messages that define the call to
be
>>placed (I'm guessing that this may be a question for James)?
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>For calling a number from Switchboard a couple of things need to be
>known. Most of the info comes form the cti. schema. These are messages
>sent by the line's handler, the meteor, asterisk, modem etc. They
detail
>the type of line, PSTN, vPSTN, SIP,Skype etc. whether that line can be
>used as a dialler and also the current status of that line, free or
>busy.
>
Ok--I'm looking at the ReadMe.rtf in xAPTel that depicts the schema.
Are the values for Network opaque (i.e., does switchboard change
behavior or perform validation against the values)? If so, then I could
use others as they make sense for asterisk--right?  In general, asterisk
tends to communicate to VoIP/PSTN gateways, direct PSTN lines and VoIP
networks.  The first two reasonably map into PSTN (but, often, the
outgoing and sometimes incoming "line" can be both
"busy" and
"available" at the same time since additional
"channels" can be added
"on the fly").  Perhaps this last comment addresses the dialer
aspect.
For direct PSTN connections, a "line" in asterisk might be
considered
outgoing and incoming.  As a general rule, all VoIP "lines" are
either
incoming or outgoing (but, not both).  So.... how does this all "map
out" (or does it not)?

If I just start feeding switchboard "test" cti.info and cti.event
messages, can I expect switchboard to react in some way that I can
better understand its behavior (I don't have normal "PSTN" or
Skype to
test)?

>Using these three bits of info Switchboard is able to list an
>accurate choice of lines to dial on, i.e. lines of the correct type and
>ones that are available.( btw vPSTN is a PSTN call that at some part
>travels over voip, SkypeIn/Out is an example.)
>
>
So, the user has no control over this?  If I'm at my desk, I would want
to specify that my deskphone (which could be some subaddress) is the
phone to connect to some VoIP "line" w/ some corresponding
destination
"number".  I get the feeling that switchboard only knows about
the
outgoing line and assumes that only one device can be connected to
it--is that accurate?

>When it comes to actual dialling there are two sequences of dial
>messages depending on type of line. This is all done to prevent
>accidental calls being made and not terminated so running up big bills.
>For regualr calls pstn,vpstn and probably sip
>Switchboard sends a dial message
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>
Can you tell me what schema is used for this?  All that I saw in the doc
is the meteor schema which only specifies the number to dial.

>Line is dialled
>In Switchboard the caller has an option to disconnect but it is
asssumed
>that the call will be closed by other means, such as puting the phone
down.
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What schema is used for the disconnect?

>For skype calls
>Switchboard sends a dial message
>Line is dialled
>Switchboard displays a countdown in seconds and a continue button
>If the continue button is not pressed then the skype gateway will drop
>the call
>If pressed the call can continue as normal
>In Switchboard the caller has an option to disconnect but it is
asssumed
>that the call will now be closed by other means, such as puting the
>phone down.
>
>hth
>
>James
>
>Schemas:
>http://www.mi4.biz/modules.php?name=Content&pa=showpage&pid=49
>http://www.mi4.biz/modules.php?name=Content&pa=showpage&pid=50
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>Yahoo! Groups Links
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