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Laurent Kza mirpax2000 at yahoo.frHi all, I am new to the use of the Osmocom project, it is indeed a very nice job. I am currently trying to set up a configuration with a Asterisk PBX server and I have 2 questions: 1/ RTP configuration The SIP part (sip-connector vs Asterisk connection) works well so far, the communication starts but with no audio. I noticed that the RTP flux is sent to localhost instead of my server address (set as remote in sip-connector.cfg) and I was wondering if there is any possibility to send the RTP flow to an address which is not localhost ? sip local 0.0.0.0 5069 remote 127.0.0.1 5060 2/ codec issue In a configuration where all the Osmocom servers (MSC, MGW, BSC…) and Asterisk are on the same machine, it got a message from my asterisk server, saying that no codec can be found to start a communication. By default, the wiki/manuals states that gsm has to be used but perhaps I am missing something in the BSC configuration, especially in the codec choice. <--- SIP read from UDP:10.184.10.162:5069 ---> INVITE sip:899 at 127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 10.184.10.162:5069;rport;branch=z9hG4bKpe1UXZU1amUja Max-Forwards: 70 From: <sip:422 at 0.0.0.0:5069>;tag=vyQKX32r72ZyQ To: <sip:899 at 127.0.0.1:5060> Call-ID: cd65c5a0-fdbf-1238-51a9-000c29cfd753 CSeq: 949096397 INVITE Contact: <sip:10.184.10.162:5069> User-Agent: sofia-sip/1.12.11devel Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE Supported: timer, 100rel Content-Type: application/sdp Content-Length: 133 v=0 o=Osmocom 0 0 IN IP4 127.0.0.1 s=GSM Call c=IN IP4 127.0.0.1 t=0 0 m=audio 4016 RTP/AVP 3 a=rtpmap:3 GSM/8000 a=sendrecv <-------------> --- (13 headers 8 lines) --- Sending to 10.184.10.162:5069 (no NAT) Sending to 10.184.10.162:5069 (no NAT) Using INVITE request as basis request - cd65c5a0-fdbf-1238-51a9-000c29cfd753 No matching peer for '422' from '10.184.10.162:5069' == Using SIP RTP CoS mark 5 Got SDP version 0 and unique parts [Osmocom 0 IN IP4 127.0.0.1] Found RTP audio format 3 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! Thanks for your help Laurent -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.osmocom.org/pipermail/openbsc/attachments/20200420/db5048c4/attachment.htm>