SIP/2.0 484 Address Incomplete

VitalPBX Community Support General Discussion SIP/2.0 484 Address Incomplete

  • Post
    StefaanVD
    Participant

    I’m trying to set up a SIP-trunk for incoming calls on VitalPBX 2.3.6-1. I think I have it configured ok, but it ain’t working. If I set core verbose 3, I see the following in the logs 

    == Using SIP RTP TOS bits 184

    == Using SIP RTP CoS mark 5

     

    If I turn on SIP debugging I get :

    <— SIP read from UDP:146.148.114.244:5060 —>

    INVITE sip:s@192.168.1.43:5060 SIP/2.0

    Via: SIP/2.0/UDP 146.148.114.244:5060;branch=z9hG4bK4e64e17f

    Max-Forwards: 70

    From: “0492xxxxxx” <sip:0492xxxxxx@146.148.114.244>;tag=as6e13feb4

    To: <sip:s@192.168.1.43:5060>

    Contact: <sip:0492xxxxxx@146.148.114.244:5060>

    Call-ID: 262ff56d4cb043f37a54b4af6bef3cc7@146.148.114.244:5060

    CSeq: 102 INVITE

    User-Agent: weepee

    Date: Mon, 16 Sep 2019 12:50:35 GMT

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

    Supported: replaces, timer

    X-AccountCode: 014363

    X-Switch: ssw-weepee0-d

    Remote-Party-ID: “0492xxxxxx” <sip:0492xxxxxx@146.148.114.244>;party=calling;privacy=off;screen=no

    Content-Type: application/sdp

    Content-Length: 470

     

    v=0

    o=root 1894532458 1894532458 IN IP4 146.148.114.244

    s=weepee

    c=IN IP4 146.148.114.244

    t=0 0

    m=audio 20902 RTP/AVP 8 9 110 97 18 4 3 111 0 101

    a=rtpmap:8 PCMA/8000

    a=rtpmap:9 G722/8000

    a=rtpmap:110 speex/8000

    a=rtpmap:97 iLBC/8000

    a=rtpmap:18 G729/8000

    a=rtpmap:4 G723/8000

    a=fmtp:4 annexa=no

    a=rtpmap:3 GSM/8000

    a=rtpmap:111 G726-32/8000

    a=rtpmap:0 PCMU/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:20

    a=maxptime:60

    a=sendrecv

    <————->

    — (17 headers 21 lines) —

    Sending to 146.148.114.244:5060 (no NAT)

    Sending to 146.148.114.244:5060 (no NAT)

    Using INVITE request as basis request – 262ff56d4cb043f37a54b4af6bef3cc7@146.148.114.244:5060

    Found peer ‘WeePeeOut’ for ‘0492xxxxxx’ from 146.148.114.244:5060

      == Using SIP RTP TOS bits 184

      == Using SIP RTP CoS mark 5

    Found RTP audio format 8

    Found RTP audio format 9

    Found RTP audio format 110

    Found RTP audio format 97

    Found RTP audio format 18

    Found RTP audio format 4

    Found RTP audio format 3

    Found RTP audio format 111

    Found RTP audio format 0

    Found RTP audio format 101

    Found audio description format PCMA for ID 8

    Found audio description format G722 for ID 9

    Found audio description format speex for ID 110

    Found audio description format iLBC for ID 97

    Found audio description format G729 for ID 18

    Found audio description format G723 for ID 4

    Found audio description format GSM for ID 3

    Found audio description format G726-32 for ID 111

    Found audio description format PCMU for ID 0

    Found audio description format telephone-event for ID 101

    Capabilities: us – (ulaw|alaw), peer – audio=(ulaw|gsm|g723|alaw|g722|g729|ilbc|speex|g726)/video=(nothing)/text=(nothing), combined – (ulaw|alaw)

    Non-codec capabilities (dtmf): us – 0x1 (telephone-event|), peer – 0x1 (telephone-event|), combined – 0x1 (telephone-event|)

           > 0x7fe194117990 — Strict RTP learning after remote address set to: 146.148.114.244:20902

    Peer audio RTP is at port 146.148.114.244:20902

    Looking for s in trk-2-in (domain 192.168.1.43)

     

    <— Reliably Transmitting (NAT) to 146.148.114.244:5060 —>

    SIP/2.0 484 Address Incomplete

    Via: SIP/2.0/UDP 146.148.114.244:5060;branch=z9hG4bK4e64e17f;received=146.148.114.244;rport=5060

    From: “0492xxxxxx” <sip:0492xxxxxx@146.148.114.244>;tag=as6e13feb4

    To: <sip:s@192.168.1.43:5060>;tag=as5933b29d

    Call-ID: 262ff56d4cb043f37a54b4af6bef3cc7@146.148.114.244:5060

    CSeq: 102 INVITE

    Server: VitalPBX

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

    Supported: replaces, timer

    Content-Length: 0

     

    Any idea what is going wrong here?

     

    Regards,

     

    Stefaan

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Viewing 5 replies - 1 through 5 (of 5 total)
  • Replies
    StefaanVD
    Participant
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    Just a note to myself….

    I suspect it has todo with the fact that my SIP-provider sends the phone number without international number. It sends 0492xxxxxx instead of  +32492xxxxxx. The generated dialplan has

    exten => _[+*#0-9A-Za-z].,1,NoOp(Incoming call through:

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    May you share your trunk configurations? obviously you must hide any sensitive data.

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    StefaanVD
    Participant
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    @ing-joserivera26

    I used textmode

    Description WeePee

    Trunk default

     

    Outgoing

    WeePeeOut

    disallow=all
    host=sip0-d.voice.weepee.io
    fromdomain=sip0-d.voice.weepee.io
    username=3299xxxxxxx
    secret=passwd
    fromuser=3299xxxxxxxx
    insecure=port,invite
    trustrpid=yes
    sendrpid=yes
    qualify=yes
    canreinvite=no
    type=friend
    allow=ulaw
    allow=alaw
    insecure=port,invite
    qualify=yes
    nat=yes

    Incoming

    WeePeeIn

    secret=passwd
    type=user
    allow=ulaw
    allow=alaw

    registration string

    3299xxxxxx:password@sip0-d.voice.weepee.io

     

    Asterisk Cli

    WeePeeOut/3299xxxxxx    146.148.114.244                             Yes        Yes            5060     OK (21 ms)  WeePee

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    On the peer and user section add the following parameter:

    callbackextension=YOUR_DID
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    StefaanVD
    Participant
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    @ing-joserivera26

    Thank you for pointing me in the right direction. Adding my DID to the registration string worked!

    3299xxxxxx:password@sip0-d.voice.weepee.io/DID

    I found this on the net :

    A similar effect can be achieved by adding a “callbackextension” option in a peer section.
    ; this is equivalent to having the following line in the general section:
    ;
    ; register => username:secret@host/callbackextension

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Viewing 5 replies - 1 through 5 (of 5 total)
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