VoIP. MS – No matching endpoint found on inbound calls.

VitalPBX Community Support General Discussion VoIP. MS – No matching endpoint found on inbound calls.

  • Post
    JSLLC
    Participant
    Multi-tenant trunk issue, how is this possible?

     

    pbx*CLI> pjsip show registrations

     

    <Registration/ServerURI…………………………>  <Auth……….>  <Status…….>

    ==========================================================================================

     

    T4_274431/sip:seattle2.voip.ms:5060                     T4_274431-oauth   Registered

    T5_274431_interwest/sip:104.129.57.250:5060             T5_274431_interwest-oauth  Registered

    Objects found: 2

     

    pbx*CLI> pjsip show endpoints

     

    Endpoint:  T5_274431_interwest                                  Not in use    0 of inf

    OutAuth:  T5_274431_interwest-oauth/274431_interwest

    InAuth:  T5_274431_interwest-auth/T5_274431_interwest

    Aor:  T5_274431_interwest                                1

    Contact:  T5_274431_interwest/sip:274431_interwest@1 0e2dccc13f Avail        20.357

    Transport:  transport-udp             udp      0      0  0.0.0.0:5060

     

     

    PJSIP Logging Enabled for host: 104.129.57.250

    <— Received SIP request (980 bytes) from UDP:104.129.57.250:5060 —>

    INVITE sip:4063125787@66.62.204.19 SIP/2.0

    Via: SIP/2.0/UDP 104.129.57.250:5060;branch=z9hG4bK511302bf;rport

    Max-Forwards: 70

    From: <sip:4065776345@104.129.57.250>;tag=as7a46804b

    To: <sip:4063125787@66.62.204.19>

    Contact: <sip:4065776345@104.129.57.250:5060>

    Call-ID: 2fb5710b1b21a187136db7a03abfe99b@104.129.57.250:5060

    CSeq: 102 INVITE

    User-Agent: voip.ms

    Date: Wed, 23 Sep 2020 19:54:37 GMT

    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

    Supported: replaces, timer

    X-Dest-User: 274431_interwest

    Remote-Party-ID: “4065776345” <sip:4065776345@104.129.57.250>;party=calling;privacy=off;screen=no

    Content-Type: application/sdp

    Content-Length: 272

     

    v=0

    o=root 246819294 246819294 IN IP4 104.129.57.250

    s=voip.ms

    c=IN IP4 104.129.57.250

    t=0 0

    m=audio 17840 RTP/AVP 0 18 101

    a=rtpmap:0 PCMU/8000

    a=rtpmap:18 G729/8000

    a=fmtp:18 annexb=no

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:20

    a=sendrecv

     

    [2020-09-23 13:54:37] NOTICE[15404]: res_pjsip/pjsip_distributor.c:676 log_failed_request: Request ‘INVITE’ from ‘<sip:4065776345@104.129.57.250>’ failed for ‘104.129.57.250:5060’ (callid:2fb5710b1b21a187136db7a03abfe99b@104.129.57.250:5060) – No matching endpoint found

    <— Transmitting SIP response (496 bytes) to UDP:104.129.57.250:5060 —>

    SIP/2.0 401 Unauthorized

    Via: SIP/2.0/UDP 104.129.57.250:5060;rport=5060;received=104.129.57.250;branch=z9hG4bK511302bf

    Call-ID: 2fb5710b1b21a187136db7a03abfe99b@104.129.57.250:5060

    From: <sip:4065776345@104.129.57.250>;tag=as7a46804b

    To: <sip:4063125787@66.62.204.19>;tag=z9hG4bK511302bf

    CSeq: 102 INVITE

    WWW-Authenticate: Digest realm=”asterisk”,nonce=”1600890877/1537af5fcf49874cf8fd316d7f059247″,opaque=”200609083c90161d”,algorithm=md5,qop=”auth”

    Server: NetPBX

    Content-Length:  0

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  • Replies
    Up
    -1
    Down
    Here’s a blog from VOIP.MS that shows you how to configure a PJSIP trunk. The only thing you have to do is to translate plane settings to VitalPBX GUI.

    https://wiki.voip.ms/article/Asterisk_PJSIP

    Also, make sure PJSIP is running on port 5060. This is the default port on VitalPBX 3. However, if you have migrated from v2 to v3, the default port is 5062.

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