Urgently required hardware spec for 1000 concurrent SIP cals

Please help me to get the specification for the following:

  1. What asterisk version we have to use (whether we can use the community
    version or we have to go for business edition) ?
  2. How many concurrent SIP peers an asterisk server can handle considering the
    following features?
    2.1 codec trans coding (g711, g729, gsm, etc.)
    2.2 Conferencing
  3. What is the hardware configuration required (shall we handle 1000
    concurrent SIP peers in a single asterisk server)?
    3.1 Please specify the number of servers.
    3.2 Other equipments we have to use.
    3.3 We may use Grandstream / Tenor for terminating the calls.

PLEASE dont double post. See response in support forum.


Use openSER+Asterisk

they both will work for you. Asterisk alone can’t handle 1000 cocurrent calls. they both will aslo do load balancing. You can use any for incoming and outbound like Quintum Tenor or Audiocode.


Please note that OpenSER no longer exists. It was forked into two projects, Kamailio and OpenSIPS. The two projects are currently very similar, given that they are built on the same code base, however the two projects have differing philosophies and goals (hence the fork).