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Peter Stuge peter at stuge.seSipos Csaba wrote: > I am setting up openbsc with LCR and Asterisk to provide external > connectivity for the GSM clients, and I encountered and interesting > problem. 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. 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.) > But when the SIP phones initiates a voice call towards the GSM phone, > only the SIP phone can hear the voice of the GSM phone, and not the > other way around (half sided call). 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. //Peter