audio

  1. 1

    No Audio on Calls to Single Number (sort of)

    Hi all, I'm having a strange problem on one of my FusionPBX installations. There's an external number that when called from any extension, appears to have no audio on the call. Either to the person calling from the Fusion system, or (apparently) the person they're calling. If I then go to the...
  2. I

    Call Recordings

    Hey all, For some odd reason, the start (30 seconds or so) of most call recordings is a bunch of robotic, extremely slow, junk that you can't understand. You can't make out anything said. All of a sudden, the call comes into "focus" and there's nothing wrong. Has anyone experienced this or have...
  3. A

    One Way Audio Via Dual NIC.

    Hello, I am facing one way audio on the calls. I have 2 Networks configured. One is Public IP and one is Private Network (From Voice Provider) Public Network Config IP:192.168.0.10 GW : 192.168.0.1 Private Network Config IP : 10.0.0.5 GW : 10.0.0.4 Termination Config Signalling IP : 10.0.1.10...
  4. N

    Poor gateway audio quality

    I set up a few local and external sip endpoints and they work fine. Then i tried to configure PSTN gateway (i'm using openvox, it's in the same network) and audio quality for remote side is very bad, voice sounds more metal, but local extension sound is good. endpoints codec - 722,alaw,ulaw...
  5. E

    Remote Extensions behind NAT - Registering but no audio

    Hi All, I am looking to achieve the following setup: Phone (local net 192.168.1.0) <-> NAT Router A <-> Internet <-> NAT Router B <-> Fusion PBX (local net 192.168.0.0) Using dynamic DNS for NAT Router B and opening SIP+RTP ports in routers+firewalls I am managing to register the phone to...
  6. J

    No audio on Fusion/Freeswitch 1.6.20, Debian 8, behind firewall - hosted Google cloud

    Hi all Let me say in advance, Thank you, as I have struggled all day trying to get this working. I have deployed the latest release Fusion/Freeswitch - 1.6.20 - to a Google cloud Debian 8 instance. I've created a new SIP profile for my domain [pbx01.mydomain.com], I've set the ext-rtp-ip and...