Trunk Problem

Discussion in 'General' started by craagle, Feb 11, 2011.

  1. craagle

    Joined:
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    I can not make VoIP calls over the service provider. When I called the reason for an extension on the SIP built-in number over the phone line going to the server. Instead, the user must go to my number, they said. How do I do? I have written the following settings in the console output and the trunk.

    Code:
    -- AGI Script recordingcheck completed, returning 0
        -- Executing [s@macro-record-enable:5] MacroExit("SIP/55-0000016c", "") in new stack
        -- Executing [02125015196@from-internal:4] Macro("SIP/55-0000016c", "dialout-trunk|2|02125015196||") in new stack
        -- Executing [s@macro-dialout-trunk:1] Set("SIP/55-0000016c", "DIAL_TRUNK=2") in new stack
        -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/55-0000016c", "0?sub-pincheck|s|1") in new stack
        -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/55-0000016c", "0?disabletrunk|1") in new stack
        -- Executing [s@macro-dialout-trunk:4] Set("SIP/55-0000016c", "DIAL_NUMBER=02125015196") in new stack
        -- Executing [s@macro-dialout-trunk:5] Set("SIP/55-0000016c", "DIAL_TRUNK_OPTIONS=tr") in new stack
        -- Executing [s@macro-dialout-trunk:6] Set("SIP/55-0000016c", "OUTBOUND_GROUP=OUT_2") in new stack
        -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/55-0000016c", "1?nomax") in new stack
        -- Goto (macro-dialout-trunk,s,9)
        -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/55-0000016c", "0?skipoutcid") in new stack
        -- Executing [s@macro-dialout-trunk:10] Set("SIP/55-0000016c", "DIAL_TRUNK_OPTIONS=") in new stack
        -- Executing [s@macro-dialout-trunk:11] Macro("SIP/55-0000016c", "outbound-callerid|2") in new stack
        -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/55-0000016c", "0|SetCallerPres|") in new stack
        -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/55-0000016c", "0|Set|REALCALLERIDNUM=55") in new stack
        -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/55-0000016c", "1?normcid") in new stack
        -- Goto (macro-outbound-callerid,s,6)
        -- Executing [s@macro-outbound-callerid:6] Set("SIP/55-0000016c", "USEROUTCID=") in new stack
        -- Executing [s@macro-outbound-callerid:7] Set("SIP/55-0000016c", "EMERGENCYCID=") in new stack
        -- Executing [s@macro-outbound-callerid:8] Set("SIP/55-0000016c", "TRUNKOUTCID=") in new stack
        -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/55-0000016c", "1?trunkcid") in new stack
        -- Goto (macro-outbound-callerid,s,12)
        -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/55-0000016c", "0|Set|CALLERID(all)=") in new stack
        -- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/55-0000016c", "0|Set|CALLERID(all)=") in new stack
        -- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/55-0000016c", "0|SetCallerPres|prohib_passed_screen") in new stack
        -- Executing [s@macro-dialout-trunk:12] ExecIf("SIP/55-0000016c", "1|AGI|fixlocalprefix") in new stack
        -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
           >  fixlocalprefix: Using pattern 0NXXNXXXXXX
      ==  fixlocalprefix: Dialpattern 0NXXNXXXXXX matched. 02125015196 -> 02125015196
        -- AGI Script fixlocalprefix completed, returning 0
        -- Executing [s@macro-dialout-trunk:13] Set("SIP/55-0000016c", "OUTNUM=02125015196") in new stack
        -- Executing [s@macro-dialout-trunk:14] Set("SIP/55-0000016c", "custom=SIP/DoruknetOUT") in new stack
        -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/55-0000016c", "0|Set|DIAL_TRUNK_OPTIONS=M(setmusic^)") in new stack
        -- Executing [s@macro-dialout-trunk:16] Macro("SIP/55-0000016c", "dialout-trunk-predial-hook|") in new stack
        -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/55-0000016c", "") in new stack
        -- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/55-0000016c", "0?bypass|1") in new stack
        -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/55-0000016c", "0?customtrunk") in new stack
        -- Executing [s@macro-dialout-trunk:19] Dial("SIP/55-0000016c", "SIP/DoruknetOUT/02125015196|300|") in new stack
        -- Called DoruknetOUT/02125015196
        -- Got SIP response 500 "account has been moved to a remote system" back from 212.58.0.122
        -- SIP/DoruknetOUT-0000016d is circuit-busy
      == Everyone is busy/congested at this time (1:0/1/0)
        -- Executing [s@macro-dialout-trunk:20] Goto("SIP/55-0000016c", "s-CONGESTION|1") in new stack
        -- Goto (macro-dialout-trunk,s-CONGESTION,1)
        -- Executing [s-CONGESTION@macro-dialout-trunk:1] GotoIf("SIP/55-0000016c", "1?noreport") in new stack
        -- Goto (macro-dialout-trunk,s-CONGESTION,3)
        -- Executing [s-CONGESTION@macro-dialout-trunk:3] NoOp("SIP/55-0000016c", "TRUNK Dial failed due to CONGESTION - failing through to other trunks") in new stack
        -- Executing [02125015196@from-internal:5] Macro("SIP/55-0000016c", "outisbusy|") in new stack
        -- Executing [s@macro-outisbusy:1] Playback("SIP/55-0000016c", "all-circuits-busy-now|noanswer") in new stack
        -- <SIP/55-0000016c> Playing 'all-circuits-busy-now' (language 'en')
      == Spawn extension (macro-outisbusy, s, 1) exited non-zero on 'SIP/55-0000016c' in macro 'outisbusy'
      == Spawn extension (from-internal, 02125015196, 5) exited non-zero on 'SIP/55-0000016c'
        -- Executing [h@from-internal:1] Macro("SIP/55-0000016c", "hangupcall") in new stack
        -- Executing [s@macro-hangupcall:1] GotoIf("SIP/55-0000016c", "1?noautomon") in new stack
        -- Goto (macro-hangupcall,s,3)
        -- Executing [s@macro-hangupcall:3] NoOp("SIP/55-0000016c", "TOUCH_MONITOR_OUTPUT=") in new stack
        -- Executing [s@macro-hangupcall:4] GotoIf("SIP/55-0000016c", "1?noautomon2") in new stack
        -- Goto (macro-hangupcall,s,6)
        -- Executing [s@macro-hangupcall:6] NoOp("SIP/55-0000016c", "MONITOR_FILENAME=") in new stack
        -- Executing [s@macro-hangupcall:7] GotoIf("SIP/55-0000016c", "1?theend") in new stack
        -- Goto (macro-hangupcall,s,9)
        -- Executing [s@macro-hangupcall:9] Hangup("SIP/55-0000016c", "") in new stack
      == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/55-0000016c' in macro 'hangupcall'
      == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/55-0000016c'
    
    Code:
    Outgoing Settings
    Trunk Name = Doruknet OUT
    host=cscvoip.doruk.net.tr
    username=77721254458770000
    secret=XXXXXXXX
    type=peer
    allow=ulaw&alaw&g729&gsm&g723
    disallow=all
    nat=auto
    insecure=very
    dtmfmode=rfc2833
    context=default
    
    Register String
    77721254458770000:XXXXXX@cscvoip.doruk.net.tr/77721254458770000
    
    You call the extension number 55 through 55 on the opposite side say that instead, the user should have my number.
     
  2. mm.alpha2k

    Joined:
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    add "canreinvite = no"

    test
    [NOMBRE_USUARIO_SIP]
    type=peer
    context=SUCONTEXTOENTRADA
    canreinvite=no
    fromuser=NOMBRE_USUARIO_SIP
    host=cscvoip.doruk.net.tr
    insecure=invite
    nat=yes
    disallow=all
    dtmfmode=rfc2833
    allow=ulaw&alaw&g729&gsm&g723
    qualify=yes
    sendrpid=yes
    trustrpid=no
     
  3. jgutierrez

    Joined:
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    What I would do is contact my provider and ask them, why they are returning me the following message:
    Got SIP response 500 "account has been moved to a remote system" back from 212.58.0.122
    And make them to check what is happening on their side
     
  4. dicko

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    You have disallowed all your codecs after allowing the ones you want.
     
  5. craagle

    Joined:
    Feb 4, 2011
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    Hi;

    @mm.alpha2k
    I did not change, the problem continues :)

    @jgutierrez
    When I enter the IP address of the server did not register. I think I have a different process, going to see in working hours. But I have a situation like this; extensions number while you search and see how VoIP providers?

    @dicko
    But I did not understand exactly what you wanted, "disallaw = all" command removed not much has changed.
     
  6. dicko

    Joined:
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    I am just telling you that when you get your system working the calls will still fail with your posted configuration

    ref
    :

    http://www.voip-info.org/wiki/view/Aste ... g+sip.conf

    .
    .
    allow = <codec> : Allow codecs in order of preference (Use DISALLOW=ALL first, before allowing other codecs)
    .
    .

    but

    @mm.alpha2k
    I did not change, the problem continues


    was possibly a mistake, you SHOULD try making that change.
     

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