sd Unable to find a codec translation path: (gsm) -> (vp8) Pages like this one: https://community.asterisk.org/t/unable-to-find-a-codec-translation-path-for-g729/71592
with similar issues you supposed to check the codes available: asterisk*CLI> core show codecs and asterisk*CLI> show translation And check if you have a gsm to vp8 translation in the translations and codes available. Thanks Seb Sebastian Wagner Director Arrakeen Solutions http://arrakeen-solutions.co.nz/ <https://www.youracclaim.com/badges/da4e8828-743d-4968-af6f-49033f10d60a/public_url> <https://www.youracclaim.com/badges/b7e709c6-aa87-4b02-9faf-099038475e36/public_url> On Thu, 29 Oct 2020 at 20:09, Maxim Solodovnik <solomax...@gmail.com> wrote: > 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 >
