[Freeswitch-users] No audio for conference call
Mike Jerris
mike at freeswitch.org
Thu Jan 23 18:24:20 UTC 2020
Of note: 1.8.x is not the latest version. You should be looking at the latest 1.10.x releases.
> On Jan 6, 2020, at 8:11 AM, SamyGo <govoiper at gmail.com> wrote:
>
> Hi,
> We had a FS 1.8.6 dev setup and got exactly the same problem, no audio from participants in a conference. We upgraded to latest version 1.8.7 and the conference started working.
> All connected parties were able to record their joining name, and could hear conference announcing the joining participants recorded msg but there was no audio being mixed. So I tried undead, unmute ALL but in vain. Only version upgrading helped.
>
> PS. We have a mix of WebRTC + regular devices on the conference and now everything works fine. WebRTC users are fronted by Kamailio+RTPEngine.
>
> Regards,
> Sammy
>
>
>
>
> On Mon, Jan 6, 2020 at 8:52 AM Md,Mehedi Hasan Kabir(Tanim) <tanim at surroundapps.com <mailto:tanim at surroundapps.com>> wrote:
> Hi David Villasmil
>
> No. I just dialled 3000 from two zoiper android client to test freeswitch default conference dialplan at conf/dialplan/default.xml. My freeswitch version is 1.8.6.Direct audio call between two extension is fine.
>
> Regards
> Tanim
>
> On Mon, Jan 6, 2020 at 4:21 PM David Villasmil <david.villasmil.work at gmail.com <mailto:david.villasmil.work at gmail.com>> wrote:
> Hello,
>
> This is a webrtc conference?
>
> On Mon, 6 Jan 2020 at 07:07, Md,Mehedi Hasan Kabir(Tanim) <tanim at surroundapps.com <mailto:tanim at surroundapps.com>> wrote:
> Hi David Villasmil
>
> Can you guess anything from the attached log?
>
> Regards
> Tanim
>
> On Wed, Jan 1, 2020 at 3:55 PM Md,Mehedi Hasan Kabir(Tanim) <tanim at surroundapps.com <mailto:tanim at surroundapps.com>> wrote:
> Hi David
>
> Please find the attached log.
>
> in cli, conference 3000-dialengine.sensor.buzz json_list command output is as follows
> [{
> "conference_name": "3000-dialengine.sensor.buzz",
> "member_count": 2,
> "ghost_count": 0,
> "rate": 8000,
> "run_time": 235,
> "conference_uuid": "af76568b-80e1-4def-8c0a-f2e2b3076925",
> "canvas_count": 0,
> "max_bw_in": 0,
> "force_bw_in": 0,
> "video_floor_packets": 0,
> "locked": false,
> "destruct": false,
> "wait_mod": false,
> "audio_always": false,
> "running": true,
> "answered": true,
> "enforce_min": true,
> "bridge_to": false,
> "dynamic": true,
> "exit_sound": true,
> "enter_sound": true,
> "recording": false,
> "video_bridge": false,
> "video_floor_only": false,
> "video_rfc4579": false,
> "variables": {
> },
> "members": [{
> "type": "caller",
> "id": 20,
> "flags": {
> "can_hear": true,
> "can_see": true,
> "can_speak": true,
> "hold": false,
> "mute_detect": false,
> "talking": false,
> "has_video": false,
> "video_bridge": false,
> "has_floor": false,
> "is_moderator": false,
> "end_conference": false
> },
> "uuid": "584d5363-6243-4051-bb3e-6b2365ea4ed6",
> "caller_id_name": "2112",
> "caller_id_number": "2112",
> "join_time": 21,
> "last_talking": 0,
> "energy": 100,
> "volume_in": 0,
> "volume_out": 0,
> "output-volume": 0,
> "input-volume": 0
> }, {
> "type": "caller",
> "id": 19,
> "flags": {
> "can_hear": true,
> "can_see": true,
> "can_speak": true,
> "hold": false,
> "mute_detect": false,
> "talking": false,
> "has_video": false,
> "video_bridge": false,
> "has_floor": true,
> "is_moderator": false,
> "end_conference": false
> },
> "uuid": "3eb72d21-f767-4c49-a83f-f4d6ddc78eb4",
> "caller_id_name": "2111",
> "caller_id_number": "2111",
> "join_time": 235,
> "last_talking": 0,
> "energy": 100,
> "volume_in": 0,
> "volume_out": 0,
> "output-volume": 0,
> "input-volume": 0
> }]
> }]
>
> Regards
> Tanim
>
> On Tue, Dec 31, 2019 at 8:33 PM David Villasmil <david.villasmil.work at gmail.com <mailto:david.villasmil.work at gmail.com>> wrote:
> can you please share a SIP trace and all FS logs?
>
> Regards,
>
> David Villasmil
> email: david.villasmil.work at gmail.com <mailto:david.villasmil.work at gmail.com>
> phone: +34669448337
>
>
> On Tue, Dec 31, 2019 at 2:59 PM Md,Mehedi Hasan Kabir(Tanim) <tanim at surroundapps.com <mailto:tanim at surroundapps.com>> wrote:
> Hi David
>
> Yes,ext-sip-ip and ext-rtp-ip is set.Audio is ok for call between two zoiper client. problem occurs only for conference call.
>
> Regards
> Tanim
>
>
> On Tue, Dec 31, 2019, 4:36 PM David Villasmil <david.villasmil.work at gmail.com <mailto:david.villasmil.work at gmail.com>> wrote:
> Check the IP addresses offered from Zoiper and freeswitch. Make sure none is private.
>
> Is fs behind NAT? If so, did you set the ext-sip-ip and ext-rtp-ip ?
>
> On Tue, 31 Dec 2019 at 05:24, Md,Mehedi Hasan Kabir(Tanim) <tanim at surroundapps.com <mailto:tanim at surroundapps.com>> wrote:
> Hi Sergey
>
> No, call is not disconnected. Only some noise can be heard after enabling speaker.
>
> Regards
> Tanim
>
> On Mon, Dec 30, 2019 at 7:51 PM Sergey Safarov <s.safarov at gmail.com <mailto:s.safarov at gmail.com>> wrote:
> Is call disconnected after 34 second?
>
> On Mon, Dec 30, 2019 at 4:37 PM Md,Mehedi Hasan Kabir(Tanim) <tanim at surroundapps.com <mailto:tanim at surroundapps.com>> wrote:
> Hi
>
> I have tried to test FreeSWITCH conference by dialing 3000 from two zoiper clients. When I call the conference number, I can hear hold music. when the second call dial the conference number hold music stops, but no voice between legs. what would be the reason for this?
>
> Audio is fine when I call other users directly.
>
> Regards
> Tanim
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