External SIP Phone

Discussion in '3CX Phone System - General' started by Anonymous, Jun 3, 2007.

  1. Anonymous

    Anonymous Guest

    Here is the scenario

    --3cx server behind basic firewal configured with SIP port 5050 and static IP
    --3 extensions - 2 internal and one external extension setup as external and bound to media server.
    --3cx firewall forwarding 9000-9003 & 5050 to 3cx server
    --External extension is a Linksys SPA962 behind NAT firewall with STUN server configured and no port forwarding configured.

    The Linksys phone registers with 3CX and I can dial extensions yet I hear no audio or ring back.?

    Calls show as connected in 3CX server log.

    How do I get audio?

    Brian
    http://www.jaydien.com
     
  2. archie

    archie Well-Known Member
    3CX Staff

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    Bind it to media server
     
  3. Anonymous

    Anonymous Guest

    Already have that option set. Anything else?..

    Brian
     
  4. Anonymous

    Anonymous Guest

    All traffic going through a firewall requires the RTP ports open.

    It is a big range but you can narrow it down as you go, I recommend ports 10000 - 20000 to start with the RTP is for your sound.
     
  5. Anonymous

    Anonymous Guest

    Ok. So just so I'm clear, this is the firewall on the 3cx side? And how do RTP ports translate into UDP/TCP. Because when forwarding ports, those are the options on the type of ports to forward. Thanks for the help by the way. My wife is about to kill me with this 3cx project.

    Brian
    http://www.jaydien.com
     
  6. Anonymous

    Anonymous Guest

    Partly correct, the firewall that services the VoIP traffic needs to be configured to allow the RTP traffic through.

    So if you have a firewall in your office sitting between your router/switch and the internet you have to enable that one to allow for the VoIP traffic.


    It doesn't RTP sits on the UDP. So for RTP to work enable the ports using UDP.

    LOL, you get that :) that is why I work when she sleeps and be in the office when she is awake LOL.

    PS a tip, lock up those kitchen knifes........
     
  7. 3CXsupport

    3CXsupport New Member
    3CX Staff

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    Also, your server status logs may be of great help here. If you can attempt a call and copy the logs here maybe there will be something intersting to see.
     
  8. Anonymous

    Anonymous Guest

    8) Thanks. I'll have to remember that.

    Are there any known issues with using a nonstandard SIP port for 3CX. I currently have 3CX configured to use port 5050 and can't get propper communications.

    2 Internal extension, One extension is a 3cx softphone and the other is a Cisco phone. Both phones are registered but when I place a call from the Softphone to the cisco phone, I hear no audio in from the softphone and the Cisco phone does not ring. The Cisco phone also never shows activity in 3cx line status.

    Here is the log from the 3cx soft phone as I placed the call.

    07:12:06,511: Connect Request: 2B 00 01 00 02 80 13 00 01 00 00 00 01 00 04 80 31 30 31 05 00 80 32 32 32 00 00 09 01 00 01 00 00 00 00 00 00 00 00 02 91 81 00
    07:12:06,511: Connect Request: 222 to 101
    07:12:06,746: Connect Confirm: 0E 00 01 00 02 81 13 00 01 01 00 00 00 00
    07:12:06,761: Connect Confirm
    -------------------------------------------
    07:12:06,761: T: 192.168.0.2:5050
    INVITE sip:101@192.168.0.2:5050 SIP/2.0
    Via: SIP/2.0/UDP 192.168.0.102;branch=z9hG4bK0007e818fa10dc11a297000bdbb80aaa
    From: 3CX Phone <sip:222@192.168.0.2:5050>;tag=30030
    To: <sip:101@192.168.0.2:5050>
    Call-ID: 0007E818-FA10-DC11-A296-000BDBB80AAA@192.168.0.102
    CSeq: 3 INVITE
    Contact: <sip:222@192.168.0.102>
    Max-Forwards: 70
    User-Agent: SIPPER for 3CX Phone
    Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY
    Content-Type: application/sdp
    Content-Length: 351

    v=0
    o=- 3389944326 3389944326 IN IP4 192.168.0.102
    s=SIPPER for 3CX Phone
    c=IN IP4 192.168.0.102
    t=0 0
    m=audio 5062 RTP/AVP 8 0 2 3 97 110 101
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:2 G726-32/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:97 iLBC/8000
    a=rtpmap:110 speex/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=ptime:20

    -------------------------------------------
    07:12:06,871: R: 192.168.0.2:5050
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 192.168.0.102;branch=z9hG4bK0007e818fa10dc11a297000bdbb80aaa
    To: <sip:101@192.168.0.2:5050>;tag=a76dcd58
    From: "3CX Phone"<sip:222@192.168.0.2:5050>;tag=30030
    Call-ID: 0007E818-FA10-DC11-A296-000BDBB80AAA@192.168.0.102
    CSeq: 3 INVITE
    Content-Length: 0
     

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