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@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@surroundapps.com <mailto:tanim@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@gmail.com <mailto:david.villasmil.work@gmail.com>> wrote: Hello,
This is a webrtc conference?
On Mon, 6 Jan 2020 at 07:07, Md,Mehedi Hasan Kabir(Tanim) <tanim@surroundapps.com <mailto:tanim@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@surroundapps.com <mailto:tanim@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@gmail.com <mailto:david.villasmil.work@gmail.com>> wrote: can you please share a SIP trace and all FS logs?
Regards,
David Villasmil email: david.villasmil.work@gmail.com <mailto:david.villasmil.work@gmail.com> phone: +34669448337
On Tue, Dec 31, 2019 at 2:59 PM Md,Mehedi Hasan Kabir(Tanim) <tanim@surroundapps.com <mailto:tanim@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@gmail.com <mailto:david.villasmil.work@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@surroundapps.com <mailto:tanim@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@gmail.com <mailto:s.safarov@gmail.com>> wrote: Is call disconnected after 34 second?
On Mon, Dec 30, 2019 at 4:37 PM Md,Mehedi Hasan Kabir(Tanim) <tanim@surroundapps.com <mailto:tanim@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