SIP Incoming Calls Rejected with Message “SIP/2.0 484 Address Incomplete”

VitalPBX Community Support General Discussion SIP Incoming Calls Rejected with Message “SIP/2.0 484 Address Incomplete”

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    anush.intech
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    Hi,

    I have configured a SIP Trunk, the outbound calls are working as required but whenever an Incoming Call is placed, my VitalPBX sends “SIP/2.0 484 Address Incomplete” message to the SIP Provider.

    I doubt that the problem is with the way the To: Header is passed by the provider as it uses tel: URI format.

    For Example, Incoming Call Invite:

    INVITE sip:s@10.17.108.201:5062 SIP/2.0
    Via: SIP/2.0/UDP 10.237.192.35:5060;branch=z9hG4bK-*2*d5d8143fc41e2792a5dctaN0
    To: <tel:+91XXXXX>
    From: <tel:+91XXXX>;tag=ztesipIP61CD6Vxh*1-3-20481*bbcb.1
    Call-ID: 9dAC03QA_qReUcc8d0ewCkVN2py6avL3iadg@zteims
    CSeq: 1000 INVITE
    Max-Forwards: 65
    Contact: <sip:10.237.192.35:5060;b_p=DIAG_2_0_02277f66;zte-did=1-3-20481-16523-12-523>
    Supported: 100rel,timer
    P-Early-Media: supported
    Session-Expires: 1800;refresher=uac
    Min-SE: 90
    P-Asserted-Identity: <tel:+91XXXX>
    Privacy: none
    Allow: INVITE,PRACK,ACK,UPDATE,CANCEL,BYE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY
    Accept: application/sdp, application/isup, multipart/mixed, application/dtmf, application/dtmf-relay
    Content-Type: application/sdp
    Content-Disposition: session
    Content-Length: 362

    v=0
    o=- 676358270 1796814569 IN IP4 10.237.192.98
    s=-
    c=IN IP4 10.237.192.98
    t=0 0
    m=audio 27678 RTP/AVP 8 0 18 4 97 101
    c=IN IP4 10.237.192.98
    a=rtpmap:8 PCMA/8000/1
    a=rtpmap:0 PCMU/8000/1
    a=rtpmap:18 G729/8000/1
    a=rtpmap:4 G723/8000/1
    a=rtpmap:97 AMR/8000/1
    a=fmtp:97 mode-set=2,7
    a=rtpmap:101 telephone-event/8000/1
    a=fmtp:101 0-15
    a=sendrecv
    <————->
    — (19 headers 16 lines) —
    Sending to 10.237.192.35:5060 (no NAT)
    Sending to 10.237.192.35:5060 (no NAT)
    Using INVITE request as basis request – 9dAC03QA_qReUcc8d0ewCkVN2py6avL3iadg@zteims
    Found peer ‘mum-sip’ for ‘XXXX’ from 10.237.192.35:5060
    == Using SIP RTP TOS bits 184
    == Using SIP RTP CoS mark 5
    Got SDP version 1796814569 and unique parts [- 676358270 IN IP4 10.237.192.98]
    Found RTP audio format 8
    Found RTP audio format 0
    Found RTP audio format 18
    Found RTP audio format 4
    Found RTP audio format 97
    Found RTP audio format 101
    Found audio description format PCMA for ID 8
    Found audio description format PCMU for ID 0
    Found audio description format G729 for ID 18
    Found audio description format G723 for ID 4
    Found unknown media description format AMR for ID 97
    Found audio description format telephone-event for ID 101
    Capabilities: us – (ulaw|alaw|g729), peer – audio=(ulaw|g723|alaw|g729)/video=(nothing)/text=(nothing), combined – (ulaw|alaw|g729)
    Non-codec capabilities (dtmf): us – 0x1 (telephone-event|), peer – 0x1 (telephone-event|), combined – 0x1 (telephone-event|)
    Peer audio RTP is at port 10.237.192.98:27678
    Looking for s in trk-4-in (domain 10.17.108.201)

     

    The Response sent by VitalPBX:

    <— Reliably Transmitting (no NAT) to 10.237.192.35:5060 —>
    SIP/2.0 484 Address Incomplete
    Via: SIP/2.0/UDP 10.237.192.35:5060;branch=z9hG4bK-*2*d5d8143fc41e2792a5dctaN0;received=10.237.192.35
    From: <tel:+91XXXXX>;tag=ztesipIP61CD6Vxh*1-3-20481*bbcb.1
    To: <tel:+9122XXXXX>;tag=as2a6a1c23
    Call-ID: 9dAC03QA_qReUcc8d0ewCkVN2py6avL3iadg@zteims
    CSeq: 1000 INVITE
    Server: VitalPBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
    Supported: replaces, timer
    Content-Length: 0

     

    Can any help me understand the problem and if possible a work around for same ?

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  • Replies
    PitzKey
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    To: <tel:+91XXXXX>

    Asterisk does not accept tel URIs

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