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uk.telecom.voip (UK VOIP) (uk.telecom.voip) Discussion of topics relevant to packet based voice technologies including Voice over IP (VoIP), Fax over IP (FoIP), Voice over Frame Relay (VoFR), Voice over Broadband (VoB) and Voice on the Net (VoN) as well as service providers, hardware and software for use with these technologies. Advertising is not allowed.

noob guidance

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Old July 13th 06, 02:53 AM posted to uk.telecom.voip
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Default noob guidance

Hi everyone,

I've been doing a lot of research into all things VoIP lately:
asterisk, voicexml, the JAIN API, OpenVXI, etc., etc. Having gained a
basic appreciation of these various technologies, I was hoping someone
could offer some insight into which option would best suit my goal:

I need to build a relatively basic server architecture that can accept
SIP calls from a remote user, pass audio (from the user) to a
parser/recognizer/etc., and then returns pre-recorded audio and
language recognition "scores". The usual features such as barge-in and
silence-detection are preferred, but no sort of routing or PBX features
are needed.

The catch is, we are trying to do it with free tools, with a particular
emphasis on implementing the Sphinx recognizer. The description of what
I want to do is exactly what voicexml provides, but I have yet to find
a voicexml solution in which I can incorporate Sphinx (and it seems to
me that OpenVXI is a mess). I have looked into building a custom server
using the Galaxy architecture, but I am having trouble finding
information on how to establish a physical SIP connection. JAIN looks
great, but unless Im mistaken, more lower-level work must be done to
establish a connection (SIP stack?).

I would really appreciate it if someone with some experience would
offer a helping hand and give their impressions. Is it possible to
implement voicexml with Sphinx? Would working from the group up with an
SIP API be better? Could I strip down Asterisk to do the job?

Thanks a lot.


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