3 sites unable to communicate

Discussion in '3CX Phone System - General' started by locutus2k, Mar 26, 2009.

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  1. locutus2k

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    We are using 3 sites. One site hosts the 3CX server, and one IP Phone. The second site hosts an ATA, and one IP Phone. The third site hosts 2 IP phones. After some resets, we were able to get the two phones at site 3 to register with the server, but when we call the voicemail system, we get no audio. We are also unable to dial between extentions. After a period of time one of the phones will nolonger be registered with the server.

    At the site with the 3CX server, is a Verizon FiOS router. Ports 5060 and 9000-9049 are open and pointed to the 3CX server. At site 3 (the ones with 2 phones) we have a Linksys wrt54g2 firewall. The phones are Linksys SPA941 running firmware 5.1.8 as documented.

    I think this is a firewall issue, but putting the server in a DMZ does not seem to help the issue. Any help would be greatly appreciated. Hopefully I have articulated the issue well enough.
     
  2. leejor

    leejor Well-Known Member

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    What about the site where the phones are located, are ports forwarded through a router or are you using STUN (recommended). Did they work in the past then just stop? Not hearing the Voicemail at one of the "remote" phones suggests that a port to that phone isn't open. What is the registration interval of the sets?
     
  3. nb

    nb Support Team
    Staff Member 3CX Support

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    Remove the PBX from DMZ
    Make sure that you have ports forwarded correctly.

    Also how are you connecting the remote offices together? using the bridge or the Tunnel?

    If you still have a problem with ports why dont you use tunnel functionality? If 5090 is open and the tunnels register you will ALWAYS have audio. On each and every call you make.
     
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  4. austintheen

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    Success! We now have site 3 working behind NAT with full audio and everything appears to work.

    In addition to the STUN, NAT and SIP settings below, we had to enable PBX delivers audio option in the Extension options in 3cx.

    Found the answer at voipuser.org

    STUN Support
    SIP > NAT Support Parameters
    Handle VIA received: yes
    Handle VIA rport: yes
    Insert VIA received: yes
    Insert VIA rport: yes
    Substitute VIA Addr: yes
    Send Resp To Src Port: yes
    STUN Enable: yes
    STUN Test Enable: yes
    STUN Server: any public STUN server
    NAT Keep Alive Intvl: 15 (can be increased after you are confident everything is configured properly)

    Line 1 > NAT Settings
    NAT Mapping Enable: yes
    NAT Keep Alive Enable: yes
    NAT Keep Alive Msg: $PING
    NAT Keep Alive Dest: $PROXY
     
  5. nb

    nb Support Team
    Staff Member 3CX Support

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    perfect - Well done!!
     
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