ould also experience this when dialing using SIP.
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Eric Wieling * Asteria Solutions Group * Huntsville, AL
Call centers * IVRs * Enterprise PBXs * Conferencing applications
256-705-0277 * http://www.asteriasgi.com/ * sa...@asteriasgi.com
___
> cards in the same computers, but not on different ones.
You can install Asterisk on the PCs with the cards in them and then use
SIP or IAX2 to transport those calls to your main Asterisk server.
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Eric Wieling * Asteria Solutions Group * Huntsville, AL
Call centers * IVRs * Enterprise PBXs * Co
ire a terminal of some kind?
It requires an ADSI analog phone. It's not nearly as cool as it sounds
like. IIRC, ADSI uses 2400bps FSK bursts to do what it does. It does
not help that most ADSI phones seem to be locked by the telco or the vendor.
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Eric Wieling * Asteria Solutions Gro
would new phones cost? If they cost more than $10 each then
you will spend more money by buying new phones rather than buying a G729
license. Remember G729 is licensed on a per simultaneous channel basis.
i.e. if you buy 10 licenses you can have up to 10 G729 channels in use
at any one ti
arate parameters for
> application, functions and macros?
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Eric Wieling * Asteria Solutions Group * Huntsville, AL
Call centers * IVRs * Enterprise PBXs * Conferencing applications
256-705-0277 * http://www.asteriasgi.com/ * sa...@asteriasgi.com
_
ill welcome..
>
>
>
>
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ion about why this behaviour is so
>> bad/inconsistent/something that it should be changed. Simply labeling it
>> a bug is just a conclusion. Why is it a bug???
>
> It's a bug, because the "i" extension has a very limited intended usage, and
> any additional case
DERR only goes to the Asterisk console if you are running 1.4 or later
and enable agi debug in the CLI.
I seem to recall something about AGIs not working correctly (streamfile
or DTMF read) if your AGI script does not process the input Asterisk
sends it on STDIN when Asterisk starts the
use my rxgain is too low on that port.
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Eric Wieling * Asteria Solutions Group * Huntsville, AL
Call centers * IVRs * Enterprise PBXs * Conferencing applications
256-705-0277 * http://www.asteriasgi.com/ * sa...@asteriasgi.com
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Robert Broyles wrote:
> Okay. I'm using this all over SIP Trunking with Vitelity.
> Any other suggestions?
Sorry for wasting your time.
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Eric Wieling * Asteria Solutions Group * Huntsville, AL
Call centers * IVRs * Enterprise PBXs * Conferencing applications
256-705-
>> Is there a convenient way to limit the number of call files (outgoing
>> directory) that are processed concurrently?
>
> On Fri, 27 Feb 2009, Danny Nicholas top posted:
>
>
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Eric Wieling * Asteria Solutions Group * Huntsville, AL
Call centers * IVRs * Enterprise P
irectory in your Asterisk source.
--
Eric Wieling * Asteria Solutions Group * Huntsville, AL
Call centers * IVRs * Enterprise PBXs * Conferencing applications
256-705-0277 * http://www.asteriasgi.com/ * sa...@asteriasgi.com
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