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I wonder whether any Expert can develop a call switching platform(CSP thereafter) with second dialing features by Voice Guide for us. Basically we are implementing a VoIP call switching to PSTN. The System will accept VoIP via a H.323 compatible gateway, possible CISCO 5300/QuintumD2400/IPLink and terminate the call via an T1(24 channels) PSTN interface. The CSP then hold the line(playing connection message) and dial to our provider. Our provider is using an IVRS to accept our call and authentificaiton would be done by DTMF tone exchange. Upon authentification process success, our provider will dial to the final destination and return with ring tone or busy tone. The incoming call is therefore connected to the destination.

 

Anyone who can develop such a system pls prepare H/W, S/W required, estimated delivery time contact voipguru@mail.com. AOL messenger ID: voipguru2003. Good money!

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