Objectives:
I am trying to achieve the followings:
- I have 3 UAC within network and another 20 agents are registered to the OpenSIPS (X-Lite users) outside network.
- Most of the users are behind NAT.
- When these X-Lite users dials with a SIP ID from their soft phone / ATA / UAC (they should use user ID as a prefix), it should go to a SIP server, supplied by us (SIP invite). Later we would like it to add PSTN GW too so provision has to be in the configuration but commented out for the timebeing.
- Calls made through UA/UAC/ATA configured with DID goes to SIP Server, should go to user behind NAT configured to DID, and then it should receive/send calls to a PSTN gw (registered user is able to make calls to DIDs or registered SIP user, no unauthorized SIP forwarding).
- Asterisk used to monitor, voice mail, agent's caller id should be changed to DID number. If anyone calls, we should be able to see their DID/User ID as a caller's number/name. It should be changed via OpenSIPS, not via Asterisk.
- I would like to authenticate all users and load balancing via FreeRADIUS .
Currently I can make successful calls within the network and UAC registers outside the network but getting one way sound...
Problem:
I am having trouble configuring NAT traversal part, FreeRADIUS is not installed or included in configuration, and also the Asterisk part is not configured or checked. Since, I'm stuck an NAT Traversal thus above mentioned software is not added in configuration but needed to configure.
