External Extension to Extension - No Audio

Discussion in '3CX Phone System - General' started by mark.s.lowe, Feb 20, 2013.

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  1. mark.s.lowe

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    I initially setup 2 external extensions using Grandstreams which are located at different sites. Different firewalls. I'm able to call each other and answer but no audio. I then tested it with the 3CX Windows Softphone. Same issue. I tried different locations given it maybe a firewall issue still no audio.

    If I use the 3CX Tunnel it works great. But the Grandstreams can't use it so not a solution. If I turn on "PBX Delivers Audio" for the extensions it works as well but this is less than idea. STUN is setup on the 3CX Softphones so this should work?

    The server is setup 3CX v11 with latest updates. It has internal and external NIC's (different IP's). The firewall test completes successfully.

    I've read many WiKi's on configuring the server and clients like adding the server as the outbound proxy. No joy. Any help would be appreciated.

    ---Mark
     
  2. dimitripapa

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    open the 3cx ports to the router that you have the 3cx. find ports form the help....
    if you have static ip for 3cx use this ip with the port 5060 for stun
    if you dont have a static ip write first the dynamic ip adresss in <network settings-public ip- static public ip adresss
    config the phones , dont forget to change the stun port with 5060 and for stun the static ip or dynamic ip
    for all the other setting i think that is easy..
     
  3. jpillow

    jpillow Well-Known Member

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    For the external extensions using Grandstream handsets have you tried using the SIP Proxy Manager? Using SPM really eliminates alot of issues with routing etc.. when using external extensions, its essentially the same as using 3cx tunnel for the 3cx phone. I always check pbx delivers audio for external extensions.....
     
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  4. 3CXfoxhallsolutions

    3CXfoxhallsolutions New Member

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    Hi Mark
    I'm wondering why you say that using 'PBX delivers audio' is less than ideal ... If it makes it work, then why not?

    It might be useful to have a look through this presentation ... http://www.3cx.com/blog/advanced-training/configuring-remote-extensions/

    best regards ...
     
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  5. GManNAtl

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    I have to agree with the last post. I come from a strong PBX background and like you I felt the PBX Deliver's Audio seemed "less than ideal". In some ways I still feel that way. However, I have to say that I have looked at the loads on both the system resources and network traffic on my servers and it is negligible. It does introduce a little bit of latency, so if the connections aren't up to snuff it could be a problem, but as FoxHall said if it works it works. Just consider it a gift of a feature from 3CX. There are few (if any) systems that really make it this easy!

    Also, as jpillow said the SIP Proxy Manager is AWESOME as well.

    With that said, if you insist on making the connection P2P here is an overview of what you need to do. I don't work with Grandstream phones, so it is just a rough explanation. You will need to configure the phones to be "locked down" on the ports it negotiates for RTP during session description. Then forward the appropriate ports to the phones (in both sites) including 5060 obviously. If the ports are hard coded the RTP should succeed and navigate the firewall. In some of the FAQ's on configuring remote extensions they describe this in more detail. I believe this will break the provisioning options in 3CX, but should get you to where you need to go.
     
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