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Keith keith at rhizomatica.orgOn 20/04/2020 11:02, Laurent Kza wrote: > Hi all, Hi Laurent, it's really a very very long time since I have used Asterisk, but I can possibly help you to work through this. > 1/ RTP configuration Just to be clear, you are using the full stack, osmo-bsc/msc/mgw ? or the osmo-nitb? What is your BTS? > I was wondering if there is any possibility to send the RTP flow to an address which is not localhost ? It is signaled in the SDP, if you are originating the call from asterisk, then it's an asterisk parameter somewhere. If it's a mobile to mobile call, the B-leg is still "originating" from Asterisk. When you say "localhost instead of your server address", can you clarify, how many "servers" (be they VMs or whatever) are involved here? Just the one? Maybe you mean your public (or private) address on a network card? To my mind, "localhost" is a "server address" certainly in the case of an osmo-mgw <--> asterisk stream on the same box, then it would be. > Found audio description format GSM for ID 3 > [2020-04-20 17:38:19] NOTICE[14918][C-00000015]: chan_sip.c:10957 process_sdp: No compatible codecs, not accepting this offer! > This looks to me like Asterisk is pretty clearly saying it does not support the GSM codec, or does not have this enabled in configuration somewhere. BTW, rather than looking at SIP log on the console, I highly recommend you use sngrep, it gives you a visual representation of SIP messages and media flows that makes every some much more clear! K.