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Thread: Registered but cannot dial number

  1. #1

    Default Registered but cannot dial number

    Hi I have sucessfuly setup and registered to callcentric.com.

    The problem is I cannot dial any number as I always get dialed number doesn't exist or is temporary inavailible. On the other hand I can receive calls !?!?!?

    I think the problem is my Epygi doesn't identify itself properly.

    in the log it says Anonymous is calling (anonymous@anonymous).

    How can I change that? I tried everything.

    I am attaching SIP UA log. 17771234567 is a test number at callcentric. Calling number should be 17772448397 which is my number



    ***************************** SIP message buffer start *****************************
    INVITE sip:17771234567@callcentric.com SIP/2.0
    Via: SIP/2.0/UDP 109.93.15.17:51759;rport;branch=z9hG4bKEPSVBUS0958 26da-6b26-493f-8442-ff3170c5e05e
    To: <sip:17771234567@callcentric.com>
    From: "anonymous" <sip:anonymous@anonymous>;tag=12581313454cb54694-20bf-427c-91f4-2d2b5fea645f
    CSeq: 172 INVITE
    Call-ID: 71379654925772460_9be07cb9-3922-48a9...ygi-config.com
    Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, UPDATE
    Contact: "Nikola Simic" <sip:17772448397@109.93.15.17:51759>
    Content-Type: application/sdp
    Supported: replaces, norefersub, timer
    User-Agent: Epygi Quadro SIP User Agent/v5.1.19 (QUADRO-2X)
    Min-SE: 20
    Session-Expires: 1800
    Max-Forwards: 70
    Content-Length: 199

    v=0
    o=- 132 178 IN IP4 109.93.15.17
    s=-
    c=IN IP4 109.93.15.17
    t=0 0
    m=audio 51768 RTP/AVP 18 101
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    ***************************** SIP message buffer end ******************************

    18:29:16 Receive SIP message # (13/11/2009 17:29:16:163 GMT) # UDP # 378 bytes # buff size 1 # from: 204.11.192.23:5060 # to: 192.168.1.3:5060

    ***************************** SIP message buffer start *****************************
    SIP/2.0 403 User does not exist
    v: SIP/2.0/UDP 109.93.15.17:51759;rport=51806;branch=z9hG4bKEPSVB US095826da-6b26-493f-8442-ff3170c5e05e
    f: "anonymous" <sip:anonymous@anonymous>;tag=12581313454cb54694-20bf-427c-91f4-2d2b5fea645f
    t: <sip:17771234567@callcentric.com>
    i: 71379654925772460_9be07cb9-3922-48a9...ygi-config.com
    CSeq: 172 INVITE
    l: 0

    ***************************** SIP message buffer end ******************************

    18:29:16 Try to send SIP message # (13/11/2009 17:29:16:170 GMT) # UDP # 485 bytes # buff size 1 # from: 192.168.1.3:5060 # to: 204.11.192.23:5060

    ***************************** SIP message buffer start *****************************
    ACK sip:17771234567@callcentric.com SIP/2.0
    Via: SIP/2.0/UDP 109.93.15.17:51759;rport;branch=z9hG4bKEPSVBUS0958 26da-6b26-493f-8442-ff3170c5e05e
    To: <sip:17771234567@callcentric.com>
    From: "anonymous" <sip:anonymous@anonymous>;tag=12581313454cb54694-20bf-427c-91f4-2d2b5fea645f
    CSeq: 172 ACK
    Call-ID: 71379654925772460_9be07cb9-3922-48a9...ygi-config.com
    User-Agent: Epygi Quadro SIP User Agent/v5.1.19 (QUADRO-2X)
    Max-Forwards: 70
    Content-Length: 0

    ***************************** SIP message buffer end ******************************

    18:29:16 UA --> CM # ReportError # Error: RequestRejected # Sip error: 403 # SID: 71379654925772460

  2. #2

    Default addition

    when I try all the same with X-Pro softphone everything works fine, so it is not firewall/NAT issue

  3. #3

    Default

    Hi,

    The call may be made from "anonymous" if the "Hiding Caller Information" is enabled for the extension from which you are making the call. You can find that setting under the "Caller ID Based Services" of the extension (User->Extensions Management->click on the extension link) under the "Supplementary Services" menu.
    That "anonymous" in the From filed really can be the reason of the problem, because Callcentric may not allow the calls from "anonymous" users. So find that setting, change it and report about results.

    Regards,
    Ruben
    Last edited by RubenS; 11-13-2009 at 01:39 PM.

  4. #4

    Default

    Hi
    You do not say if you have set up call routing, as if I want to dial at my sip provider I add a dial out number first then the sip number you wish to call, it sounds like you are just dialing the sip number only, and the Quadro cannot find the correct route for the call
    Derek

  5. #5

    Default Routing is correct

    Quote Originally Posted by derekkmark View Post
    Hi
    You do not say if you have set up call routing, as if I want to dial at my sip provider I add a dial out number first then the sip number you wish to call, it sounds like you are just dialing the sip number only, and the Quadro cannot find the correct route for the call
    Derek
    Hi routing is correct. I haven't had a chance to try to fix what RubenS suggested. I'll do it on monday. Anyway I didn't put it to hide caller id so I guess that is not default setting, but I will check tomorrow.

    rgds.

  6. #6

    Default

    RubenS,

    you were right, this solved the problem. On the extension I was dialing from "hiding caller information" was set to yes. So when I changed it to NO everything work fine.

    Thanks a lot!!!
    Goran

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