On 12/11/08 17:14, Jeffrey Ollie wrote:
On Thu, Dec 11, 2008 at 3:51 AM, Daniel-Constantin Mierla miconda@gmail.com wrote:
On 12/11/08 00:49, Jeffrey Ollie wrote:
Hello, I'm a newbie with OpenSER/Kamailio so please bear with me. I'm trying to use OpenSER 1.3.4 and rtpproxy to connect two IP PBXs. The two PBXs (and the phones they serve) cannot talk directly to each other, they need to go through my OpenSER system. One system is a Cisco CallManager cluster (v5.1.3) which I administer and the other is a Mitel system run by another organization. I have a CentOS box running OpenSER and rtpproxy with two interfaces one on each network. Routing on the CentOS box appears correct as I can ping everything I expect to. I've gotten OpenSER configured so that SIP messages are sent between the PBXs, but the SDPs aren't getting rewritten properly and the RTP isn't getting passed through the rtpproxy. I've attached my config so far, can anyone point out where I've gone wrong or point me to some example configs that might help me out? All of the example configs that I've found so far seem to be oriented towards SIP endpoints.
is routing working between the two ip interfaces or you need rtpproxy in bridge mode? Check: http://kamailio.org/docs/modules/1.3.x/nathelper.html http://openser.svn.sourceforge.net/viewvc/openser/trunk/modules/nathelper/ex...
Well, I've modified my config to look like the above example, but something still isn't working... The SDPs never get re-written so the RTP isn't flowing through the rtpproxy. I've attached my current config and a packet capture of a call. I'm sure I'm missing something simple... Can anyone point out what I'm doing wrong?
FWIW, rtpproxy is started with:
/usr/bin/rtpproxy -F -s unix:/var/run/rtpproxy.sock -t 0xB8 -m 10000 -M 20000
in the second link, in the comments, it is mentioned that rtpproxy needs a special -l parameter format, specifying the interfaces to listen to.
Cheers, Daniel