Hi Michael, Is your Opus codec loaded ? Do you see these in FS log ? 2015-09-16 11:27:00.235630 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 8000hz 40ms (VBR) 2015-09-16 11:27:00.235644 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 8000hz 40ms (VBR) 2015-09-16 11:27:00.235651 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 8000hz 120ms (VBR) 2015-09-16 11:27:00.235658 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 8000hz 100ms (VBR) 2015-09-16 11:27:00.235663 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 8000hz 80ms (VBR) 2015-09-16 11:27:00.235669 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 8000hz 60ms (VBR) 2015-09-16 11:27:00.235675 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 8000hz 20ms (VBR) 2015-09-16 11:27:00.235680 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 8000hz 20ms (VBR) 2015-09-16 11:27:00.235685 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 8000hz 10ms (VBR) 2015-09-16 11:27:00.235690 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 8000hz 10ms (VBR) 2015-09-16 11:27:00.235696 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 48000hz 40ms (VBR) 2015-09-16 11:27:00.235701 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 48000hz 20ms (VBR) 2015-09-16 11:27:00.235707 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 48000hz 20ms (VBR) 2015-09-16 11:27:00.235712 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 48000hz 10ms (VBR) 2015-09-16 11:27:00.235717 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 48000hz 10ms (VBR) Any errors in your log ? Are u sure you bridge the call to the g.711 trunk ? I think your client-to-client calls work because is just doing media bypass or pass through . Regards, Dragos On 2015-09-16 13:31, Michael Nielsen wrote:
My setup is based upon https://github.com/voxserv/freeswitch_conf_minimal [1].
On Wed, Sep 16, 2015 at 1:26 PM, Michael Nielsen <mic.niel84@gmail.com> wrote:
I've got a client with OPUS and G.711 codecs enabled and FreeSWITCH with OPUS and G.711 codecs enabled. I've got a SIP gateway connected to FreeSWITCH with only support for G.711.
I want to have clients only use OPUS codec - as this seems best for mobile data connections.
As of now client <-> FS <-> client use OPUS, but whenever the SIP gateway is involved G.711 is chosen.
If I disable G.711 codec on the client side, to force OPUS, I'm not allowed to dial out via my SIP gateway. So transcoding to G.711 does not occur.
My fs_cli reports this: Hangup sofia/external/+4413265874 [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]
Again, reenabling G.711 in the client and everything works.
I have not set disable-transcoding=true.
What to do?
Links: ------ [1] https://github.com/voxserv/freeswitch_conf_minimal
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