Dismiss Notice
We would like to remind you that we’re updating our login process for all 3CX forums whereby you will be able to login with the same credentials you use for the Partner or Customer Portal. Click here to read more.

Inbound calls drop after 1 ring

Discussion in '3CX Phone System - General' started by mobilesoundz, Mar 5, 2013.

Thread Status:
Not open for further replies.
  1. mobilesoundz

    Joined:
    Oct 24, 2008
    Messages:
    19
    Likes Received:
    0
    Having issues with a new voip provider and the inbound number, it drops the call after one ring. I have sent this nuber to another 3cx install and it works as required so i know its not the provider, please see log below that might help someone to point me in the right direction. I have tried changing where the call goes from auto attendant to a voicemail but still drops after one ring

    05-Mar-2013 18:56:06.499 [CM500002]: Unidentified incoming call. Review INVITE and adjust source identification:
    Invite-UNK Recv Req INVITE from 84.45.107.53:5060 tid=yBZKFSm9BQeQm Call-ID=f607f50f-0068-1231-ce87-0026b95a467f:
    INVITE sip:fred.smith@sip.mydomain.com SIP/2.0
    Via: SIP/2.0/UDP 84.45.107.53;rport=5060;branch=z9hG4bKyBZKFSm9BQeQm
    Max-Forwards: 68
    Contact: <sip:mod_sofia@84.45.107.53:5060>
    To: <sip:fred.smith@sip.mydomain.com>
    From: "078784****8"<sip:07878453628@84.45.107.53>;tag=S7Ugvc8crtrXp
    Call-ID: f607f50f-0068-1231-ce87-0026b95a467f
    CSeq: 40918772 INVITE
    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY
    Content-Disposition: session
    Content-Type: application/sdp
    Supported: timer, precondition, path, replaces
    User-Agent: TTNC VoIP Network
    Allow-Events: talk, refer
    Content-Length: 291
    Remote-Party-ID: "078784****8" <sip:07878*****8@84.45.107.53>;party=calling;screen=yes;privacy=off
    X-FS-Support: update_display

    v=0
    o=FreeSWITCH 1362478353 1362478354 IN IP4 84.45.107.53
    s=FreeSWITCH
    c=IN IP4 84.45.107.53
    t=0 0
    m=audio 31320 RTP/AVP 8 0 3 101 13
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=rtpmap:13 CN/8000
    a=ptime:20

    I have added my 3cx server to the DMZ just incase its a firewall issue but experience the same issue, any help would be appreciated
     
  2. leejor

    leejor Well-Known Member

    Joined:
    Jan 22, 2008
    Messages:
    11,263
    Likes Received:
    358
    Confirm that you have datafilled your domain name INVITE sip:fred.smith@sip.mydomain.com SIP/2.0 in 3CX.

     
  3. mobilesoundz

    Joined:
    Oct 24, 2008
    Messages:
    19
    Likes Received:
    0
    Yes data is in there correctly i just removed it sorry
     
  4. leejor

    leejor Well-Known Member

    Joined:
    Jan 22, 2008
    Messages:
    11,263
    Likes Received:
    358
    Have you read this ?

    http://www.3cx.com/blog/docs/source-identification-issues/
     
  5. mobilesoundz

    Joined:
    Oct 24, 2008
    Messages:
    19
    Likes Received:
    0
    Spot on the article did the job, i am thinking it worked on my 3cx install as it is a lot older version and this secuirty setting has come in on the newer release, many thanks
     
Thread Status:
Not open for further replies.