Fernando,
I'll leave #1 for somebody else to answer (AFIK, thats a
codec feature
and hence not under the control of sipcomm). As for #2, that
would be a
third party call control ( server ) feature. SipComm ought
to work
unchanged with a third party call controller (which runs on
the server ).
Ranga
Fernando Lujan wrote:
> I developed a pbx/client solution integrated to the CRM
of my company
> using asterisk and moziax.
>
> I need to know two things:
>
> 1)Does sip-communicator supports echo cancellation?
> 2)Can you develop the click to dial feature?
>
> We can pay some money for the feature.
>
> Thanks in advance.
>
> Fernando Lujan
>
>
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--
M. Ranganathan
Advanced Networking Technologies Division,
National Institute of Standards and Technology (NIST),
100 Bureau Drive, Stop 8920, Gaithersburg, MD 20899.
tel:301 975 3664 , fax:301 590 0932 http://w3.antd.nist.gov/
Advanced Networking Technologies For the People!
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