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Re: xAP Asterisk integration
Hi Paul,
Sorry about the delay in responding. I needed to give your list a bit
of thought before responding. Comments are embedded below.
Gregg
Paul Gale wrote:
> Hi Greg,
>
> First of all, I don't consider myself an Asterisk expert - I do have
> it running with a Sipura SPA-3000 though, which was a feat in itself!
> I'm using the Asterisk@Home package.
>
> To be honest, I've not really thought in great detail what I want to
> achieve but these are some of the areas:
>
> · Call logging: I currently use James' xAPSwitchBoard which is
> really great. I currently have both PSTN and Skype calls logged using
> this and a couple of Meteor's but it would be nice to allow some
> integration with Asterisk calls, especially other incoming/outgoing
> VOIP calls.
>
A very minimal capability shouldn't be too difficult; Ove's AGI is a
good start for CID. After James's explanation of the mapping of lines
(virtual or real) to subaddresses, the AGI that I use (and, I'm guessing
Ove's as well) needs to be modified. Likewise, for outbound call
logging. I'll revisit this depending upon how far Ove pushes his script
and/or your need. I personally use this the most and therefore will
mature an implementation before most anything else.
> · Calling from xAPSwitchBoard: There is an option to call out
> from the contacts and log screen - would be nice to maybe integrate
> this with Asterisk?
>
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)? If so, can
it or does it send the device that should place the call as the target
subaddress? This would be required to initiate the call from a specific
phone that is managed by asterisk. From a user perspective, you'd
initiate the call from an app like xAPSwitchboard. The call would then
be initiated and would "ring" both the phone that you want to
place the
call from as well as the call destination phone. Assuming that the
details can be worked through on the xAPSwitchBoard side as to how the
info can be passed, then I can create a manager app to do this. I'm
rather motivated because my VoIP phone doesn't have speed dial lists and
I'm too lazy to want to manually dial numbers.
> · IVR menus: One of my original goals with Asterisk was to build
> some interactive voice response menus to speak the status of various
> systems at home such as alarm status, history etc as a direct
> replacement for my Comfort remote menus. It would also allow control
> of certain HA functions - heating, curtains etc. I'm not sure whether
> this needs the xAP Asterisk support though or is possible already?
>
I'm thinking a "first pass" would be a BSC command sender--which
is
one-way only. Subsequent iterations might perform BSC status queries
and "feed back the results" to you (either via dynamic TTS or
mapping to
pre-recorded audio). The first pass is quite easy and the second--I
don't see as too difficult. However, I'm not as motivated until I
complete the misterhouse BSC support--which would then let me use this
capability to more easily control misterhouse devices. So, this is a
bit down the priority list for me.
> · Better voices: I currently use NeoSpeech voices at home which
> are pretty good.
>
Yep--I have Kate and Paul as well.
> Is there some way to generate TTS on a remote PC/Server and then feed
> it back to Asterisk - again, maybe not a xAP Asterisk function? I'd be
> interested in what you currently do here.
>
Let me first address what I currently do--which is not the voices that
you are asking about. I use the "Frank" voice from cepstral.com.
It's
engine can run on Linux and therefore can be used to more easily to
create the wav/gsm file that can be streamed via AGI. The misterhouse
distro has this agi already and I can send you a copy if you decide to
pursue it.
The question that you are asking, however, is a considerably more
difficult. I haven't personally worked w/ the SAPI side of .Net yet;
so, I don't know how complex it is to intercept or grab a stream from
the NeoVoice engine and then restream to an asterisk "listener".
The
asterisk listener would have to buffer the inbound stream and then
"restream" it via an AGI. Overall, I'd say this is a bit complex
and
I'm not overly motivated (especially since the cepstral solution works
now). For purely "experimental purposes", I used the cepstral
demo site
to create wav files of interest and converted them to "gsm"
format using
sox so that they could be used as prerecorded voice prompts (and,
therefore, not require a TTS engine). That's a fairly trivial
experiment to "trial" a voice that you are considering for
eventual TTS
use in asterisk.
> · Control of HA devices depending on where a call is answered -
> i.e. an incoming call is answered in my office (certain asterisk
> extension). Via a xAP message, I can then mute or pause playback of
> audio playing through the whole house audio system or just mute that
room.
>
I see this AGI as no different than the one above addressing IVR menus.
The exception is that you really don't want to force too much
"smarts"
into an AGI--just have it send "event" info. That might mean that
some
xAP "middle-ware" might need to exist to perform "HA
logic" and then do
the retransmit. From what I can tell, xAP Switchboard does this (of
sorts) by mapping notification xAP messages based on various received
telephony events. Personally, I use misterhouse for implementing the
majority of my "middle-ware"; if you wanted something else, then
you'd
be on your own.
> I'm sure others will come up with a better thought out list - these
> were just off the top of my head.
>
> Paul.
>
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