Re: xAP Asterisk integration

[James <james@xxxxxxx>]

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>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)?
>
It logs the network type on incoming calls so when dialling those
contacts it knows what networks to route the call back to.

> If so, then I could
>use others as they make sense for asterisk--right?
>
Correct, but it's good to keep them precise. VOIP would be bad as it is
an umbrella term, msm,sip,skype would be fine.

> In general, asterisk
>tends to communicate to VoIP/PSTN gateways, direct PSTN lines and VoIP
>networks.
>
The each gateway can report as a vpstn device using one set of sub
addresses, pstn as pstn and voip as sip all using different sub addresses

> 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)?
>
>
All lines report using the same schema. The only difference between
incoming / both ways is the inclusion of a dialler=yes line and
different uids and sub addresses

>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)?
>
>
>
There are a few ways of seeing this. firstly the line status screen.
(little telephone under the big telephone on the homepage). Assuming
dialling is turned on in the config screen then clicking on a  dial
button will list all available and suitable lines.

>>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?
>
>
Currently yes. It can send out going calls to any number of gateways but
it assumes that when dialling the user has access to that line to pick
it up.
As switchboard is web based i would have to track the ip address of the
user and have a preference for the local phone for that ip. The ip for
the local phone could then be added to the cti. messages.

>
>
>>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
>>
>>
>>
>>
>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.
>
>
cti.cmd
http://www.mi4.biz/modules.php?name=Content&pa=showpage&pid=50

>
>
>>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.
>>
>>
>>
>>
>>
>What schema is used for the disconnect?
>
>
cti.cmd again


Switchboard was originally designed around pstn and was extended to
cover the voip features of skype. Given the needs required for full voip
support i will get it added. Figuring out voip/sip/* is high on my todo
list anyway as I've got to roll some it out at work! So don't worry if
Switchboard lacks some needed features, I'll get them added.

James

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<br>
<blockquote cite="mid42E0128F.7090208@xxxxxxx"
type="cite">
<pre wrap=""><!---->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)?</pre>
</blockquote>
It logs the network type on incoming calls so when dialling those
contacts it knows what networks to route the call back to.<br>
<blockquote cite="mid42E0128F.7090208@xxxxxxx"
type="cite">
<pre wrap=""> If so, then I could
use others as they make sense for asterisk--right? </pre>
</blockquote>
Correct, but it's good to keep them precise. VOIP would be bad as it is
an umbrella term, msm,sip,skype would be fine.<br>
<blockquote cite="mid42E0128F.7090208@xxxxxxx"
type="cite">
<pre wrap=""> In general, asterisk
tends to communicate to VoIP/PSTN gateways, direct PSTN lines and VoIP
networks. </pre>
</blockquote>
The each gateway can report as a vpstn device using one set of sub
addresses, pstn as pstn and voip as sip all using different sub
addresses<br>
<blockquote cite="mid42E0128F.7090208@xxxxxxx"
type="cite">
<pre wrap=""> 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)?
</pre>
</blockquote>
All lines report using the same schema. The only difference between
incoming / both ways is the inclusion of a dialler=yes line and
different uids and sub addresses<br>
<br>
<blockquote cite="mid42E0128F.7090208@xxxxxxx"
type="cite">
<pre wrap="">
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)?

</pre>
</blockquote>
There are a few ways of seeing this. firstly the line status screen.
(little telephone under the big telephone on the homepage). Assuming
dialling is turned on in the config screen then clicking on a  dial
button will list all available and suitable lines.<br>
<br>
<blockquote cite="mid42E0128F.7090208@xxxxxxx"
type="cite">
<pre wrap=""></pre>
<blockquote type="cite">
<pre wrap="">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.)


</pre>
</blockquote>
<pre wrap=""><!---->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?
</pre>
</blockquote>
Currently yes. It can send out going calls to any number of gateways
but it assumes that when dialling the user has access to that line to
pick it up.<br>
As switchboard is web based i would have to track the ip address of the
user and have a preference for the local phone for that ip. The ip for
the local phone could then be added to the cti. messages.<br>
<blockquote cite="mid42E0128F.7090208@xxxxxxx"
type="cite">
<pre wrap="">
</pre>
<blockquote type="cite">
<pre wrap="">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


</pre>
</blockquote>
<pre wrap=""><!---->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.
</pre>
</blockquote>
cti.cmd<br>
<a class="moz-txt-link-freetext" href="http://www.mi4.biz/modules.php?name=Content&pa=showpage&pid=50";>http://www.mi4.biz/modules.php?name=Content&amp;pa=showpage&amp;pid=50</a><br>
<blockquote cite="mid42E0128F.7090208@xxxxxxx"
type="cite">
<pre wrap="">
</pre>
<blockquote type="cite">
<pre wrap="">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.



</pre>
</blockquote>
<pre wrap=""><!---->What schema is used for the
disconnect?
</pre>
</blockquote>
cti.cmd again<br>
<br>
<br>
Switchboard was originally designed around pstn and was extended to
cover the voip features of skype. Given the needs required for full
voip support i will get it added. Figuring out voip/sip/* is high on my
todo list anyway as I've got to roll some it out at work! So don't
worry if Switchboard lacks some needed features, I'll get them
added.<br>
<br>
James<br>




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