Problem with bridge

Discussion in '3CX Phone System - General' started by Alexord, May 17, 2010.

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

    Joined:
    May 28, 2008
    Messages:
    19
    Likes Received:
    0
    Hi,

    I successfully connected two 3cx with bridge but I unable to make calls between office brunches. In test I call ext 222 (I dial 7222 due to outbound rules). I can see following error on caller side:

    12:07:03.782 [CM503016]: Call(1926): Attempt to reach <sip:7222@192.168.0.200> failed. Reason: Not Found
    12:07:03.782 [CM503003]: Call(1926): Call to sip:222@192.168.1.5:5060 has failed; Cause: 404 User unknown.; from IP:192.168.1.5:5060
    12:07:03.220 [CM503025]: Call(1926): Calling PBXline:222@(Ln.20000@Main)@[Dev:sip:192.168.1.5:5060]
    12:07:03.173 [CM503004]: Call(1926): Route 1: PBXline:222@(Ln.20000@Main)@[Dev:sip:192.168.1.5:5060]
    12:07:03.157 [CM503010]: Making route(s) to <sip:7222@192.168.0.200>
    12:07:03.157 [CM505001]: Ext.208: Device info: Device Identified: [Man: Linksys;Mod: SPA Series;Rev: General] Capabilities:[reinvite, no-replaces, able-no-sdp, recvonly] UserAgent: [Linksys/SPA942-6.1.5(a)] PBX contact: [sip:208@192.168.0.200:5060]
    12:07:03.157 [CM503001]: Call(1926): Incoming call from Ext.208 to <sip:7222@192.168.0.200>


    And this on called side:

    09:59:21.159 [CM500002]: Unidentified incoming call. Review INVITE and adjust source identification:
    INVITE sip:222@192.168.1.5:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bK-d8754z-2e0722318329eb64-1---d8754z-;rport=5060
    Max-Forwards: 70
    Contact: <sip:192.168.0.200:5060>
    To: <sip:222@192.168.1.5:5060>
    From: "Alexander Kitaev"<sip:192.168.1.5:5060>;tag=3e2f7b27
    Call-ID: NDdkNmEwMjQxNmYwYmFhNzBmMTFlZDZlOWUwMmRjZmY.
    CSeq: 2 INVITE
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
    Proxy-Authorization: Digest username="20000",realm="3CXPhoneSystem",nonce="414d535c0201885863:8c94e2130d0146d65bce7c676e9ea867",uri="sip:2222@192.168.1.5:5060",response="b575df407907e6d15ce612d6317ff7c8",algorithm=MD5
    User-Agent: 3CXPhoneSystem 8.0.10708.0
    Content-Length: 0
    Remote-Party-ID: "Alexander Kitaev"<sip:Alex%20Kitaev@sip.pditgroup.net:5060>;party=calling

    09:59:21.159 [CM302002]: Authentication failed due to unidentified source of: SipReq: INVITE 222@192.168.1.5:5060 tid=2e0722318329eb64 cseq=INVITE contact=192.168.0.200:5060 / 2 from(wire)
    09:59:20.846 [CM500002]: Unidentified incoming call. Review INVITE and adjust source identification:
    INVITE sip:222@192.168.1.5:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bK-d8754z-0354204fac00c51a-1---d8754z-;rport=5060
    Max-Forwards: 70
    Contact: <sip:192.168.0.200:5060>
    To: <sip:222@192.168.1.5:5060>
    From: "Alexander Kitaev"<sip:192.168.1.5:5060>;tag=3e2f7b27
    Call-ID: NDdkNmEwMjQxNmYwYmFhNzBmMTFlZDZlOWUwMmRjZmY.
    CSeq: 1 INVITE
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
    User-Agent: 3CXPhoneSystem 8.0.10708.0
    Content-Length: 0
    Remote-Party-ID: "Alexander Kitaev"<sip:Alex%20Kitaev@sip.pditgroup.net:5060>;party=calling
     
  2. leejor

    leejor Well-Known Member

    Joined:
    Jan 22, 2008
    Messages:
    10,371
    Likes Received:
    230
    This might help...http://www.3cx.com/blog/docs/source-identification-issues/
     
  3. Alexord

    Joined:
    May 28, 2008
    Messages:
    19
    Likes Received:
    0
    Yes looks reasonable though I do not understand what should be in this invite message in case of bridge. I don't want to dial external number, just extension. Secondly, there is no option to adjust these parameters as in case of PTSN device or VoIP provider.
     
  4. leejor

    leejor Well-Known Member

    Joined:
    Jan 22, 2008
    Messages:
    10,371
    Likes Received:
    230
    Have you tried sending the call to a ring group rather than one extension? What about the options in Settings/Advanced/Settings for Direct SIP calls. Could it be something in there?
     
Thread Status:
Not open for further replies.