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Okay, webrtc is working now. Now the only issue is with Video, through webrtc video is not working. If I call from Yealink t49G to Yealink t49G, video works perfect. Any Idea?

Hi Dario, Thanks for reply. SSL is working fine on the machine. When I logged in it says secure. When I see in registration I find registered on WSS but still I'm not able to call out. When I dial just video window pop up and nothing else happen. Even free-switch logs are silent.

I got it, in crontab just execute below command as root and it will sync the time as per time selected. fs_cli -x "fsctl sync_clock_when_idle"

Hi, I'm using fusionpbx and I noticed after couple of weeks free-switch time get differ about 10 minutes from the machine time. If I run a command under freeswitch console "strftime" it shows me about 10 minutes difference then the time I've on my debian machine, due to this it create problem in...

Anyone please

Hi, it's letsencrypt

Hi, I've installed fusionpbx on Debian 9 with 4.5.9 version. How can I enable WebRTC on it. I've installed valid SSL. When I click at option webrtc it takes me to "number" to dial, I dial but nothing happen. Even I do not see any log generated in sngrep and console. Please advise.

@mingus I tested on two different machines with freeswitch version 1.6.20 and has same issue on both. Also I tested with freeswitch version 1.8.5 it works well. So it seems issue is with freeswitch version. Is there a way to get this fixed on version 1.6.20?

Yes I'm at bare metal server provided by colocation america.

-- Part of FusionPBX -- Copyright (C) 2013 Mark J Crane <markjcrane@fusionpbx.com> -- All rights reserved. -- -- Redistribution and use in source and binary forms, with or without -- modification, are permitted provided that the following conditions are met: -- -- 1...

I did but no luck. It did not make any difference.

Please check this is how you want me to do it?

Also I observe one-thing, when greeting says record your message after the tone, after tone within a sec I hangup the call, but message says I've a voicemail with 3 second duration.

Yes after exactly 3 seconds. If I record the message and then press any key and listen, I'm able to listen full message, I save it and get the complete message. Now after recording message if I hangup the call it lost last 3 seconds. I recorded the message and then wait for 3 seconds and then...

It's via both way, either I received on email or listen from mailbox.

Hi Mingus, Thanks for reply back. I'm already on local block "/var/lib/freeswitch/storage/voicemail". Only last 3 seconds cutoff. It usually happen when calling from extension to extension. If someone calling from outside it works well.

Hello, anyone get this fixed? I'm having the same issue.

Hi James, can you please put any example so I can do it easily. Thanks

Hi James, yes I want to route call based on caller ID number or CNAM/caller ID name. Here is my query. Current I've a DID appointed on Fusionpbx 5162100768 and when someone from outside call on this number I've set it to ring on ext 213. Now I want if caller 6466535718 call on DID 5162100786 it...

Hi Guys, just want to know is there any possibility to route incoming calls on the bases of caller ID. Suppose I've set DID 123456789 for incoming calls. I want if New York <6466535718> call to this DID it should go to extension 214 mailbox, if anyone else is calling it should ring on ext 213...