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Incoming calls failing with 404 Not Found RRS feed

  • Question

  • Hi all,

    I've got the following scenario:

    1 x enterprise pool, called OCSPool01.domain.int, 1 mediation server called mediate.domain.int

    My User has a Server URI of sip:fred.bloggs@domain.com and a line URI of tel:1428

    telephone is 192.168.1.153, telephone number 9201, PBX is 192.168.1.11, mediation is 192.168.1.231, OCSPool01 is 192.168.1.12

     

    I send the following INVITE to the mediation server and I always receive a SIP/Trying, and then a 404 Not Found...

     

    Thanks in advance,

     

    Alan

     

    SIP Invite:

      Frame:
    + Ethernet: Etype = Internet IP (IPv4)
    + Ipv4: Next Protocol = TCP, Packet ID = 62495, Total IP Length = 1413
    + Tcp:
    - SIP: Request: INVITE sip:fred.bloggs@domain.com SIP/2.0
      + RequestLine: INVITE sip:fred.bloggs@domain.com SIP/2.0
        Record-Route: <sip:192.168.1.11:5080;lr>
        From: "SipX" <sip:9201@sipx.sipx.domain.int>;tag=t8amtg00hx
        To: <sip:fred.bloggs@domain.com>
        CallID: 47e32d46204e-zamgv991cf4g@snomSoft-000413FFFFFF
        Cseq: 1 INVITE
        Max-Forwards: 19
        Contact: <sip:gruu~28fd9dd2d9a2ebb0@sipx.sipx.domain.int>;flow-id=1
        P-Key-Flags: resolution="31x13", keys="4"
        User-Agent: snomSoft/5.3
        Accept: application/sdp
        Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
        Allow-Events: talk, hold, refer
        Supported: timer, 100rel, replaces, callerid
        Session-Expires: 3600;refresher=uas
        ContentType: application/sdp
        Content-Length: 370
        Date: Tue, 24 Apr 2007 10:57:34 GMT
        Via: SIP/2.0/TCP 192.168.1.11:5080;branch=z9hG4bK-69e6df15b1a7c3db51c93c61f55a6614
        Via: SIP/2.0/TCP 192.168.1.11;branch=z9hG4bK-c59a4c280f60d7651251aba37a4f5aa0
        Via: SIP/2.0/UDP 192.168.1.153:55291;branch=z9hG4bK-f3tjcuraa40d;rport=55291
        HeaderEnd: CRLF
    + Sdp: Session Name=call, Version=0

     

     

    Reply after a SIP Trying...

     

      Frame:
    + Ethernet: Etype = Internet IP (IPv4)
    + Ipv4: Next Protocol = TCP, Packet ID = 11421, Total IP Length = 585
    + Tcp: Flags=...PA..., SrcPort=5060, DstPort=56374, Len=533, Seq=2926951647 - 2926952180, Ack=2516741869, Win=64174 (scale factor 0) = 0
    - SIP: Response: SIP/2.0 404 Not Found
      + StatusLine: SIP/2.0 404 Not Found
        FROM: "SipX"<sip:9201@sipx.sipx.domain.int>;tag=t8amtg00hx
        TO: <sip:fred.bloggs@domain.com>;tag=c5f19cd3f
        CSEQ: 1 INVITE
        CallID: 47e32d46204e-zamgv991cf4g@snomSoft-000413FFFFFF
        MAX-FORWARDS: 70
        VIA: SIP/2.0/TCP 192.168.1.11:5080;branch=z9hG4bK-69e6df15b1a7c3db51c93c61f55a6614,SIP/2.0/TCP 192.168.168.11;branch=z9hG4bK-c59a4c280f60d7651251aba37a4f5aa0,SIP/2.0/UDP 192.168.168.153:55291;branch=z9hG4bK-f3tjcuraa40d;rport=55291
        CONTENT-LENGTH: 0
        SERVER: RTCC/3.0.0.0 MediationServer/3.0.6090.0
        HeaderEnd: CRLF

    Tuesday, April 24, 2007 11:08 AM

Answers

  • OK, With a little help from a friend, I finally got SipX to speak with OCS.

     

    Solution: I was creating a Dial Plan of 14. (which is 14 and any number of digits)

    The Dial Plan should have each OCS telephone number defined, i.e. 1428 (with 0 digits)

    Then prefix a + to the dialing rule, and it works fine.

     

    Thanks to Martin for the SipX help.

     

    Alan

    Friday, April 27, 2007 7:26 AM

All replies

  • Hi Allan,

     

    I think the request must be addressed using a valid E.164 number e.g +12345678901. The user should then have the Line URI parameter configured to "tel:+12345678901".

     

    /Janne

    Tuesday, April 24, 2007 12:53 PM
  • Hmm, OK, changed it, but still no joy. I've set up 3 different users, with the following line URIs

    tel:1428, tel:+1428 and sip:1428@domain.com

     

    My invite comes in as SIP:+1428@domain.com, but I still get the 404 error.

     

    However, I'm quite sure the problem is to do with the gateway I'm using. I've tried both SIPX and OpenSER, but neither seem to deliver the required invite header.

     

    Ah well, looks like I'll have to invest in a Dialogic...

     

    If anybody has any information on getting OCS and SIPX or OpenSer to work, I would be very interested,

     

    Thanks,

     

    Al

    Thursday, April 26, 2007 9:57 AM
  • OK, With a little help from a friend, I finally got SipX to speak with OCS.

     

    Solution: I was creating a Dial Plan of 14. (which is 14 and any number of digits)

    The Dial Plan should have each OCS telephone number defined, i.e. 1428 (with 0 digits)

    Then prefix a + to the dialing rule, and it works fine.

     

    Thanks to Martin for the SipX help.

     

    Alan

    Friday, April 27, 2007 7:26 AM
  • Hi Alan,

     

    Could you explain me how did you do to configure all about incoming and outcoming calls?

     

    Regards,

    Wednesday, May 2, 2007 6:52 PM
  • Could you please post the dial plan you've created ?

    Monday, December 31, 2007 3:41 PM
  • Any updates? Looking for info on OCS with SipX

     

    Wednesday, February 20, 2008 4:17 PM