New subject: OpenBSC + LCR + Asterisk = half sided calls in one

28 Oct 2013

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

Re: OpenBSC + LCR + Asterisk = half sided calls in one