[asterisk-users] SwitchVox and Asterisk

Luca Pradovera luca.pradovera at gmail.com
Mon May 8 11:07:09 CDT 2017

sorry for not being clear, the application part of this (the voice
directory) is already built, mostly working and I have no problem with
that. It is based on LumenVox if anyone would like to know, with just a
plain XML grammar.

I do need to get SwitchVox to send a call to Asterisk/FreePBX, which will
in turn call one of SW's extensions.


On Mon, May 8, 2017 at 9:00 AM, Antony Stone <
Antony.Stone at asterisk.open.source.it> wrote:

> On Monday 08 May 2017 at 16:43:24, Luca Pradovera wrote:
> > Hello,
> > I need to have an extension on a SwitchVox server dial out to one on an
> > Asterisk (FreePBX actually) box which will host a voice directory.
> What's a voice directory?
> > The Asterisk server will then need to dial one of the SwitchVox
> extensions
> > if it gets a voice match.
> You mean, listen to the caller speaking and identify who they are?
> Sounds "non-trivial" to me...
> > Anyone has done that, and could let me know how? So far it looks like IAX
> > peering (what SW calls "SwitchVox peering") could work?
> IAX will connect two Asterisk servers and allow them to communicate (it
> stands
> for Inter Asterisk eXchange) - think of it in the same way as a SIP trunk -
> you can have multiple calls to/from multiple numbers going over the link.
> However, are you saying that you've already got the "voice directory" and
> "voice match" parts working in Asterisk, and you just need to know how to
> dial
> between that and SwitchVox?
> Or is the "voice" part of the challenge also something you're looking for
> help
> with?
> Antony.
> --
> Numerous psychological studies over the years have demonstrated that the
> majority of people genuinely believe they are not like the majority of
> people.
>                                                    Please reply to the
> list;
>                                                          please *don't* CC
> me.
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
> New to Asterisk? Start here:
>       https://wiki.asterisk.org/wiki/display/AST/Getting+Started
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20170508/545e73e9/attachment.html>

More information about the asterisk-users mailing list