Hello All, I really hope we have Asterisk experts on this list, who willing to help :))
My current configuration is described here https://github.com/apache/openmeetings/blob/master/openmeetings-server/src/site/markdown/AsteriskIntegration.md (the only exception im using `ws` instead of `wss`) I'm trying to send video stream from OM room to Asterisk room -- Registered SIP 'omsip_user' at 192.168.1.211:39117 == Using SIP VIDEO CoS mark 6 == Using SIP RTP CoS mark 5 > 0x7fc3d0028f50 -- Strict RTP learning after remote address set to: 0.0.0.0:9 -- Executing [4005@rooms-omsip:1] GotoIf("SIP/omsip_user-00000000", "1?ok:notavail") in new stack -- Goto (rooms-omsip,4005,2) -- Executing [4005@rooms-omsip:2] ConfBridge("SIP/omsip_user-00000000", "4005,default_bridge,omsip_user") in new stack == Manager 'openmeetings' logged on from 192.168.1.211 == Manager 'openmeetings' logged off from 192.168.1.211 [Oct 29 13:56:00] WARNING[27219]: res_http_websocket.c:559 ws_safe_read: Web socket closed abruptly -- Channel CBAnn/4005-00000000;2 joined 'softmix' base-bridge <33b25509-2939-4d7e-b057-81b0ea795ca4> [Oct 29 13:56:00] WARNING[27220][C-00000002]: channel.c:5686 set_format: Unable to find a codec translation path: (gsm) -> (vp8) [Oct 29 13:56:00] WARNING[27220][C-00000002]: file.c:1262 ast_streamfile: Unable to open conf-onlyperson (format (vp8)): No such file or directory -- Channel SIP/omsip_user-00000000 joined 'softmix' base-bridge <33b25509-2939-4d7e-b057-81b0ea795ca4> [Oct 29 13:56:00] WARNING[27220][C-00000002]: translate.c:488 ast_translator_build_path: No translator path: (starting codec is not valid) [Oct 29 13:56:00] WARNING[27220][C-00000002]: channel.c:5686 set_format: Unable to find a codec translation path: (slin) -> (vp8) -- Channel SIP/omsip_user-00000000 left 'softmix' base-bridge <33b25509-2939-4d7e-b057-81b0ea795ca4> It looks like it is not working :( what can be wrong? -- Best regards, Maxim
