Hey Peter,

I'd suggest to make LCR use SIP for communcation with Asterisk. See http://stuge.se/lcr.txt for a minimal example of LCR configuration.

Do you have some more example configuration files for the Asterisk end of the SIP trunk?

Since you're using bridging you might also want to try rtp-bridge between GSM and SIP, which could allow GSM phones and SIP phones to negotiate codec directly, avoiding any transcoding. (But maybe it only works with Abis over IP and not over ISDN? I'm not sure.)

I am almost sure that RTP bridge can only be used with IP based BTSes which I don't have. My units are connecting via E1 dahdi.

You can of course continue to debug the LCR-Asterisk module but I would suggest moving to SIP since I think working with SIP on both legs makes debugging a bit easier.

My problem is that I made test calls an analyzed the logs at both LCR and Asterisk end, and there is no difference between a good and a half sided call. First, I forced the GSM phones to use TCH/F FR only, then I forced LCR and Asterisk SIP clients to use Alaw only (SIP clients are not supporting any GSM codec). Transcoding obviously happens between TCH/F FR and Alaw, but how on earth is possible that the direction of the call can affect that this transcoding is going to be a success or not? If its a transcoding failure it shouldn't work in any direction. If its a call routing problem, then the call shouldn't make its way to the called party. But none of this is what happens.

So I really don't know where to look, or how to debug this problem.

BR, Csaba