So basically to do not have a working environment yeah.... As mentioned: * You do not need both media proxy and rtp proxy. Which one you need depends on your network (i..e, bridging vs public network). * The original error is either due to a non running rtp media proxy, or misconfiguration of module variables in your script, or both....
Also explained to you is the learning curve of OpenSIPS, it took me upwards of 2 years to have a basic understanding of how a proxy as such can fit in our architecture. Even then, I run into uncharted territory. My suggestion is, take a week maybe two or three or four, and go through real material vs blogs written by people that know as much about OpenSIPS, Linux, how to solve world peace as you and I, and go through "Building Telephony Systems with OpenSIPS 1.6". Notice is say 1.6 and not .8 or .9, you will have to make the transition to the script yourself or ask on this list. I will be more than happy to help. The other benefit of going through the material is you will get enough of an understanding to be able to ask the same questions without being called out. Question like "no audio" "one way audio" etc.. do not help, we need sip traces, explanation of your network, "c=" and record route information. Kind Regards, Nick. _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
