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.
5 comments:
Hello Khan, have you tried asking around to users@lists.opensips.org ?
I did some opensips config only for "very basic" experiment, as I don't have the resources and time to do things like you did.
Method not supported - M=OPTIONS ERROR:core:forward_reply: no 2nd via found in reply
This would indicate something wrong in your configuration, or your request related to M=OPTIONS (though I don't know what it's)
About forward_reply, I'm also not familiar with what they mean as "2nd via", probably you did not put something like the 2nd forwarding address in the parameter.
thanks Jesse, I'm going to try the users@lists.opensips.org also posting this problem at some blogspots to get some help. I appreciate your input.
Hi, I hope your problem is solved. I don't have an idea about how NAT traversal works... I installed opensips for the first time, and I needed to create a demo. I know how to work with db_text alone. If you want opensips to work at full capacity, I believe you have to install radius.
Hello Khan,
Did u solved one way problem in you RTP Proxy config...? i've used me too and i really don't know how to make it fully functional. If you can pelase help me. (i don't know how to contact you by email )
thanks in advace
Sorry Stacy,
I have been busy in other projects, as soon as i have solution i will update this post. I'm sorry really i have been slacking on my hobbies :)
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