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Thread: Manual ITSP receiving 404

  1. #1

    Default Manual ITSP receiving 404

    I thought I understood the IP Carrier route setup, but obviously this newbie doesn't.

    I ran the Voip Carrier Wizard, which created an ext 99. It also created two lines in the Call Routing - one for 911 and one for *. When I place a call to the DID number, I receive a busy. Looking at the logs, I get this line -

    00:42:27 UA --> CM # MakeCall # from: 2015314448@, to: 4154586969, child: (empty), media exist, GUID: (empty) # ContactInfo: # SID: 5565670079607115754
    00:42:27 CM --> UA # OnReportError # Error: UserNotFound # SIP error: 0 # OID: 3452 # SID: 5565670079607115754
    SIP/2.0 404 Not Found

    Not Found I get it, but its the manipulation of the incoming call destination where I seem to be lost.

    From what I read, it is trying to send the call to 4154586969. When I set extension 99 to call forward to 00, nothing changes. As if it is not being routed. WHen I go into call route and modify the route with the destination of 4154586969 with NDS of 10, and a Pre of 00, I thought it would strip the destination and place the call to the auto attendant. No joy, the message reads the same, as if it is not there.

    I've tried with a tick mark on and off of route all incoming sip calls to CR.

    Questions - what am I missing in my logic - I think I'm missing one small step somewhere.

    In the Extension management for 99, should there be an attached line?
    Last edited by thodgen; 01-24-2011 at 04:12 AM.

  2. #2


    Assign 4154586969 to ext. 99 as the SIP User Name. After that the ext. 99 is should receive the call from ITSP and forward it to 00. There is no need to attach the line to ext. 99. What you did with the call route seems to be correct, strange it didn't work: since there was no extension with 4154586969 user name, the call would go to call routing table and via the record with "destination number pattern" = 4154586969 go to pbx extension 00.
    If it still doesn't work please send me the SIP INVITE message received from carrier.

  3. #3


    Thanks for the response, I really appreciate it.

    [QUOTE=hrant;13228]Assign 4154586969 to ext. 99 as the SIP User Name. QUOTE]

    Unless I'm mistaking your comment here, the SIP User Name is a part of the Registration for this ITSP. I'm unable to put the phone number in there as it would no longer register for the ITSP.

    I did try it in the Authorized User field, no joy. If I do put it in the SIP user name field, the results are as expected - no connectivity to the ITSP.

    Here is the invite (names changed to protect the semi-innocent):

    INVITE sip:2084576969@ SIP/2.0
    Via: SIP/2.0/UDP;branch=z9hG4bK2df68a2c;rport
    From: "Tom Hogen" <sip:2084314344@>;tag=as6b7c3a47
    To: <sip:2084576969@>
    Contact: <sip:2084314344@>
    Call-ID: 21c8fe2b023130977e07eb1b70507544@
    CSeq: 102 INVITE
    User-Agent: VoIPMS/SERAST
    Max-Forwards: 70
    Remote-Party-ID: "Tom Hogen" <sip:2084314344@>;privacy=off;screen =no
    Date: Tue, 25 Jan 2011 04:51:07 GMT
    Supported: replaces
    Content-Type: application/sdp
    Content-Length: 242

    o=root 2959 2959 IN IP4
    c=IN IP4
    t=0 0
    m=audio 16216 RTP/AVP 0 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=silenceSuppff - - - -

    I don't see anything obvious in this invite.

  4. #4


    Ok, you are right, the SIP user name is used for registration and if it is not the same as DID then you cannot put it there. However, you can create another virtual extension, for example, 98 and assign the DID to its SIP user name. After that you can configure the unconditional forwarding on that extension to 00.
    In your INVITE message i see that the call is made to 2084576969. Is it the DID? What is the 4154586969 then?

  5. #5


    Sorry, I've been masking the true number, and I was not consistent in how I was masking it. Neither are the true number.

    Your suggestion did indeed do the trick.

    Here is a summary of what is working -

    New External PSTN Provider is setup in "Manual" mode ( This ITSP is VOIP.MS )
    New extension is created, and new Call Control rules automatically
    I create a 2nd New Extension with the SIP User ID setup with the DID number.
    2nd New Extension is call forwarded to the 00 extension for auto attendant.

    Thanks for your help on this. I appreciate it!

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