Incoming calls rejected with 403 forbidden

VitalPBX Community Support General Discussion Incoming calls rejected with 403 forbidden

  • Post
    Tom
    Participant

    Hi,

     

    I have a pbx running incrediblepbx-2.3.8-3.x86_64. It has a callcentric trunk and a voipms trunk that were working but for some reason that I cannot determine, they have stopped accepting incoming calls. Outgoing calls work as expected. Both trunks show registered on both the asterisk command line and the providers control panel.

    I have inbound routes setup for both trunks that were working. I am not sure what changed but sngrep shows 403 forbidden on the invite when I place a call. Can someone give me an idea how to troubleshoot this problem?

     

    SIP/2.0 403 Forbidden
    204.11.192.39:5060 192.168.0.7:5060 │Via: SIP/2.0/UDP 204.11.192.39:5060;branch=z9hG4bK-9771bc9eadc848eaa718deb8ae334112;
    ──────────┬───────── ──────────┬─────────│ceived=204.11.192.39;rport=5060
    │ INVITE (SDP) │ │From: <sip:1XXXXXXXXXX@66.193.176.35>;tag=3781623535-44806
    14:58:55.344179 │ ──────────────────────────> │ │To: <sip:18623200030@ss.callcentric.com>;tag=as6fdffe6a
    +0.002274 │ 403 Forbidden │ │Call-ID: 2775960-3781623535-44778@msw1.telengy.net
    14:58:55.346453 │ <────────────────────────── │ │CSeq: 1 INVITE
    +0.027690 │ ACK │ │Server: VitalPBX
    14:58:55.374143 │ ──────────────────────────> │ │Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, M
    +5.452484 │ ACK │ │SAGE
    14:59:00.826627 │ ──────────────────────────> │ │Supported: replaces, timer
    │ │ │Content-Length: 0

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Viewing 15 replies - 1 through 15 (of 20 total)
  • Replies
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    To be on the same page, you should share your trunk configurations (Hiding any sensitive data) to determine if something is configured in a wrong way.

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    Tom
    Participant
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    @ing-joserivera26

    Yes, sorry I forgot that. The trunk configurations follow:

    Callcentric trunk:

    videosupport=no
    type=peer
    secret=XXXXXXXX
    insecure=port,invite
    host=callcentric.com
    fromuser=1777XXXXXXXX
    fromdomain=callcentric.com
    disallowed_methods=UPDATE
    disallow=all
    directmedia=no
    defaultuser=1777XXXXXXXX
    allow=ulaw

    Register string looks like the following: 1777XXXXXXX101:XXXXXXX@callcentric.com/609XXXXXXX

     

    voip.ms trunk:

    username=XXXXX
    type=peer
    trustrpid=yes
    sendrpid=yes
    secret=XXXXXXX
    qualify=yes
    nat=yes
    insecure=invite
    host=newyork4.voip.ms
    fromuser=XXXXXXX
    disallow=all
    canreinvite=nonat
    allow=ulaw

    my_account_number:XXXXXXX@newyork4.voip.ms:5060

     

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    Both are failing?

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    Try changing the

    type=peer to type=friend

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    Tom
    Participant
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    @ing-joserivera26

    yes both are failing with the 403 forbidden.

    I just changed the type to friend on both trunks but I am still getting 403 forbidden on both.

     

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    What about the SIP registry:

    asterisk -rx"sip show registry"
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    Tom
    Participant
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    @ing-joserivera26

    (vitalpbx-1 pts9) # asterisk -rx”sip show registry”
    Host dnsmgr Username Refresh State Reg.Time
    callcentric.com:5060 N 1777XXXXXXXX 46 Registered Fri, 01 Nov 2019 16:48:19
    newyork4.voip.ms:5060 N XXXXXXXXXXX 105 Registered Fri, 01 Nov 2019 16:47:34
    2 SIP registrations.
    (vitalpbx-1 pts9) #

     

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    Everything looks good. May you share a call trace (of an incoming call) from the Asterisk CLI to see if we can get some hint about what is happening.

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    Tom
    Participant
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    @ing-joserivera26

    Sure If you tell me what to do. I am an experienced Linux admin but pretty green what it comes to this pbx stuff. Do I just turn on debugging (something like “sip set debug  on”) or do you need me to do something else?

    If I just do asterisk -vvvr all I get is the following:

    vitalpbx-1*CLI>
    == Using SIP RTP TOS bits 184
    == Using SIP RTP CoS mark 5
    > 0x7f9168009e20 — Strict RTP learning after remote address set to: 72.251.239.207:11794
    vitalpbx-1*CLI>

     

     

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    First, go the asterisk CLI (asterisk -rvvvvvvvvvvvvvvv), then make an incoming call to any of your DIDs, copy the output generated on the Asterisk CLI and share here (hide any sensitive data).

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    Tom
    Participant
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    @ing-joserivera26

    (vitalpbx-1 pts2) # asterisk -rvvvvvvvvvvvvvvv
    Asterisk 16.6.0, Copyright (C) 1999 – 2018, Digium, Inc. and others.
    Created by Mark Spencer <markster@digium.com>
    Asterisk comes with ABSOLUTELY NO WARRANTY; type ‘core show warranty’ for details.
    This is free software, with components licensed under the GNU General Public
    License version 2 and other licenses; you are welcome to redistribute it under
    certain conditions. Type ‘core show license’ for details.
    =========================================================================
    Connected to Asterisk 16.6.0 currently running on vitalpbx-1 (pid = 2640)
    == Using SIP RTP TOS bits 184
    == Using SIP RTP CoS mark 5
    > 0x7f9168009e20 — Strict RTP learning after remote address set to: 204.11.192.170:50548
    vitalpbx-1*CLI>

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    @vitalpbx

    There’s no call trace

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    Tom
    Participant
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    @ing-joserivera26

    I did not think there was any useful info either but all I get when I place a call is what I pasted above. sngrep shows what I pasted in my first post. Do you want a trace with debug on or should I try something else?

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    Yes, a trace with debugging will be good.

    There’s no more info on sngrep?

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    Tom
    Participant
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    @ing-joserivera26

    Not really anything more in sngrep.

    If I understand things correctly even the trace only shows the invite being rejected.

    Please let me know what you think of the info in the attached file.

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