[asterisk-users] call screening feature

Marco Mouta marco.mouta at gmail.com
Tue Mar 18 08:42:00 CDT 2008


Your solution is Asterisk Manager Interface

http://www.voip-info.org/wiki-Asterisk+manager+API

On Tue, Mar 18, 2008 at 6:24 AM, Janu Mukherjee <janu.mukhi at gmail.com>
wrote:

> Hi,
>
> I have our software with SIP running on it.I configured asterisk server as
> proxy. How do I implement the call screening features(incoming and outgoing)
> using asterisk server.Please suggest me how to proceed on this.
>
> Thanks & Regards,
> Jahnavi.
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
Esta mensagem (incluindo quaisquer anexos) pode conter informação
confidencial para uso exclusivo do destinatário. Se não for o destinatário
pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu
esta mensagem por engano, por favor informe o emissor e elimine-a
imediatamente. Obrigado.

This e-mail message is intended only for individual(s) to whom it is
addressed and may contain information that is privileged, confidential,
proprietary, or otherwise exempt from disclosure under applicable law. If
you believe you have received this message in error, please advise the
sender by return e-mail and delete it from your mailbox. Thank you.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080318/b0908783/attachment.htm 


More information about the asterisk-users mailing list