Might I recommend a project that was written for that exact purpose? Have you heard of OpenSIPS/OpenSER or what is that other name... I can't spell it nor remember it... but it combined with FreeSWITCH would solve your problem better I suspect. Use the proxy for registrations and FreeSWITCH for media servers. /b On Dec 30, 2009, at 12:55 AM, Sharad wrote:
Thanks /b for your kind reply...
I want to integrate the freeswitch only for Auto Attendant & voicemail with a SIP server. All the users (approx 5000) shall be registered with SIP server. Whenever auto attendant or voicemail will be required, SIP server will establish the SIP trunking with Freeswitch.
So I am assuming that freeswitch may get 50-60 calls at a time either for auto attendant or voicemail. Whenever there is a call with freeswitch for auto attendant, caller will punch the desired user no. & than freeswitch will throw the call back to SIP server. in this case, RTP will not be flowing through freeswitch server.
So just want to know what should be the PC hardware specifications for this call traffic.
I am willing to buy a good server hardware for this. So plz advice...
Regards Sharad