Skip to content
  • Categories
  • Recent
  • Tags
  • Popular
  • World
  • Users
  • Groups
Skins
  • Light
  • Brite
  • Cerulean
  • Cosmo
  • Flatly
  • Journal
  • Litera
  • Lumen
  • Lux
  • Materia
  • Minty
  • Morph
  • Pulse
  • Sandstone
  • Simplex
  • Sketchy
  • Spacelab
  • United
  • Yeti
  • Zephyr
  • Dark
  • Cyborg
  • Darkly
  • Quartz
  • Slate
  • Solar
  • Superhero
  • Vapor

  • Default (No Skin)
  • No Skin
Collapse
Brand Logo

FOP2 Community Forum

  1. Home
  2. Support
  3. One way audio from dial

One way audio from dial

Scheduled Pinned Locked Moved Support
3 Posts 2 Posters 142 Views
  • Oldest to Newest
  • Newest to Oldest
  • Most Votes
Reply
  • Reply as topic
Log in to reply
This topic has been deleted. Only users with topic management privileges can see it.
  • S
    S
    sollostech
    wrote on last edited by
    #1

    I set up a new FOP2 install this weekend on a Paetec SIP service and I am getting one way audio. The settings are the same as in my other installs. Dialing from the phone audio is fine, dialing from FOP2 the number called you cannot hear what they say. Paetec has a different setup than my other SIP providers so I assume that is the issue. Wondering where I can try to hunt for a solution. For now I just turned the phonebook off so no one tries to use it.

    BTW: Bought two licenses this weekend, one for this client and one for another. Love FOP2!

    1 Reply Last reply
    0
    • N
      N
      nicolas
      wrote on last edited by
      #2

      Hello,

      FOP2 does not do audio, it just commands asterisk to perform originations, redirections, etc.

      Setting a call up FROM asterisk (fop2) is not the same as calling directly from one endpoint out to a trunk or another endpoint, because when you originate a call from asterisk it first sends a call to your endpoint and from there to the other side. So the audio path might be different in the end.

      However, this is not something you can fix with a fop2 configuration. You will have to review your asterisk configuration. If your pbx is behind NAT, be sure to set externip and localnet correctly in sip.conf and having the nat parameter on the peers/trunk that require them. Troubleshooting one way audio is cumbersome and complicated, so I wish you good luck on fixing it!

      Best regards,

      1 Reply Last reply
      0
      • S
        S
        sollostech
        wrote on last edited by
        #3

        To fix I verified the external IP, which turned out to somehow be wrong, but it was adding a canreinvite=no that appears to have resolved the problem.

        1 Reply Last reply
        0

        Hello! It looks like you're interested in this conversation, but you don't have an account yet.

        Getting fed up of having to scroll through the same posts each visit? When you register for an account, you'll always come back to exactly where you were before, and choose to be notified of new replies (either via email, or push notification). You'll also be able to save bookmarks and upvote posts to show your appreciation to other community members.

        With your input, this post could be even better 💗

        Register Login
        Reply
        • Reply as topic
        Log in to reply
        • Oldest to Newest
        • Newest to Oldest
        • Most Votes


        • Login

        • Don't have an account? Register

        • Login or register to search.
        Powered by NodeBB Contributors
        • First post
          Last post
        0
        • Categories
        • Recent
        • Tags
        • Popular
        • World
        • Users
        • Groups