12-07-2006 07:25 AM - edited 03-14-2019 07:07 PM
Hi, I'm using CallManager 5 and have set up a SIP trunk to a remote voice gateway which in turn is connected to the PSTN. I dial a number from the CCM SIP phone (a 7960), it connects but no conversation is heard. Debug on the voice gateway shows the call but reports "no codec". SIP trunk config is set to G.711a law as is the dial peer on the gateway - any ideas what I'm not doing right? thanks
12-11-2006 05:06 AM
Well, I expected a better response than nothing at all!!! Oh well, found out myself that it's a NATing issue. Looks like I need to enable NATing globally on the CCM 5 server - if anyone knows if such an option exists it'd be appreciated.
12-21-2006 03:50 AM
Pretty obvious in the end. Unlike Asterisk SIP proxy server which switches the RTP traffic, Callmanager doesn't - which is pretty obvious when I think about. The router we have setup performs NATing but no packet inspection. To translate the signalling port(s) is straightforward; to do so for the RTP range is nigh on impossible. A PIX or firewall would solve the problem by inspecting the packets, seeing the RTP traffic and then forwarding it. Easy really....
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