[asterisk-users] SIP/2.0 403 Forbidden

Dovid Bender asteriskusers at dovid.net
Sun Mar 28 07:57:28 CDT 2010


You need to ask your carrier what you are not sending them that they would like. It's usually a fromdomain or authname.
  ----- Original Message ----- 
  From: Aaron chen 
  To: Asterisk Users Mailing List - Non-Commercial Discussion ; Asterisk Developers Mailing List 
  Sent: Friday, March 26, 2010 09:22
  Subject: [asterisk-users] SIP/2.0 403 Forbidden


  hi,all

  when i send a call to other server use SIP trunk,

  i got this below,

  <--- SIP read from 222.46.18.52:5060 --->
  SIP/2.0 403 Forbidden

  what's rong with is?


    Asterisk 1.4.21.2, Copyright (C) 1999 - 2008 Digium, Inc. and others.
    Created by Mark Spencer <markster at 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.
    =========================================================================
      == Parsing '/etc/asterisk/asterisk.conf': Found
    Connected to Asterisk 1.4.21.2 currently running on gd-branch (pid = 3145)
    Verbosity is at least 3
        -- Executing [015921256331 at from-internal:1] Set("SIP/75002-b7705298", "MOHCLASS=none") in new stack
        -- Executing [015921256331 at from-internal:2] Macro("SIP/75002-b7705298", "user-callerid|SKIPTTL|") in new stack
        -- Executing [s at macro-user-callerid:1] Set("SIP/75002-b7705298", "AMPUSER=75002") in new stack
        -- Executing [s at macro-user-callerid:2] GotoIf("SIP/75002-b7705298", "0?report") in new stack
        -- Executing [s at macro-user-callerid:3] ExecIf("SIP/75002-b7705298", "1|Set|REALCALLERIDNUM=75002") in new stack
        -- Executing [s at macro-user-callerid:4] Set("SIP/75002-b7705298", "AMPUSER=75002") in new stack
        -- Executing [s at macro-user-callerid:5] Set("SIP/75002-b7705298", "AMPUSERCIDNAME=75002") in new stack
        -- Executing [s at macro-user-callerid:6] GotoIf("SIP/75002-b7705298", "0?report") in new stack
        -- Executing [s at macro-user-callerid:7] Set("SIP/75002-b7705298", "AMPUSERCID=75002") in new stack
        -- Executing [s at macro-user-callerid:8] Set("SIP/75002-b7705298", "CALLERID(all)="75002" <75002>") in new stack
        -- Executing [s at macro-user-callerid:9] ExecIf("SIP/75002-b7705298", "0|Set|CHANNEL(language)=") in new stack
        -- Executing [s at macro-user-callerid:10] GotoIf("SIP/75002-b7705298", "1?continue") in new stack
        -- Goto (macro-user-callerid,s,19)
        -- Executing [s at macro-user-callerid:19] NoOp("SIP/75002-b7705298", "Using CallerID "75002" <75002>") in new stack
        -- Executing [015921256331 at from-internal:3] Set("SIP/75002-b7705298", "_NODEST=") in new stack
        -- Executing [015921256331 at from-internal:4] Macro("SIP/75002-b7705298", "record-enable|75002|OUT|") in new stack
        -- Executing [s at macro-record-enable:1] GotoIf("SIP/75002-b7705298", "1?check") in new stack
        -- Goto (macro-record-enable,s,4)
        -- Executing [s at macro-record-enable:4] AGI("SIP/75002-b7705298", "recordingcheck|20100326-141638|1269584198.62") in new stack
        -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
      recordingcheck|20100326-141638|1269584198.62: Outbound recording enabled.
      recordingcheck|20100326-141638|1269584198.62: CALLFILENAME=OUT75002-20100326-141638-1269584198.62
        -- AGI Script recordingcheck completed, returning 0
        -- Executing [s at macro-record-enable:999] MixMonitor("SIP/75002-b7705298", "/var/spool/asterisk/monitor/gd-branch/gdbranchOUT75002-20100326-141638-1269584198.62.wav||") in new stack
        -- Executing [s at macro-record-enable:1000] Set("SIP/75002-b7705298", "RecordingFileName=/var/spool/asterisk/monitor/gd-branch/gdbranchOUT75002-20100326-141638-1269584198.62.wav") in new stack
        -- Executing [s at macro-record-enable:1001] NoOp("SIP/75002-b7705298", "/var/spool/asterisk/monitor/gd-branch/gdbranchOUT75002-20100326-141638-1269584198.62.wav") in new stack
        -- Executing [s at macro-record-enable:1002] Set("SIP/75002-b7705298", "CDR(userfield)=/var/spool/asterisk/monitor/gd-branch/gdbranchOUT75002-20100326-141638-1269584198.62.wav") in new stack
        -- Executing [015921256331 at from-internal:5] Macro("SIP/75002-b7705298", "dialout-trunk|7|015921256331||") in new stack
        -- Executing [s at macro-dialout-trunk:1] Set("SIP/75002-b7705298", "DIAL_TRUNK=7") in new stack
        -- Executing [s at macro-dialout-trunk:2] GosubIf("SIP/75002-b7705298", "0?sub-pincheck|s|1") in new stack
        -- Executing [s at macro-dialout-trunk:3] GotoIf("SIP/75002-b7705298", "0?disabletrunk|1") in new stack
        -- Executing [s at macro-dialout-trunk:4] Set("SIP/75002-b7705298", "DIAL_NUMBER=015921256331") in new stack
        -- Executing [s at macro-dialout-trunk:5] Set("SIP/75002-b7705298", "DIAL_TRUNK_OPTIONS=Ttr") in new stack
        -- Executing [s at macro-dialout-trunk:6] Set("SIP/75002-b7705298", "OUTBOUND_GROUP=OUT_7") in new stack
        -- Executing [s at macro-dialout-trunk:7] GotoIf("SIP/75002-b7705298", "1?nomax") in new stack
        -- Goto (macro-dialout-trunk,s,9)
        -- Executing [s at macro-dialout-trunk:9] GotoIf("SIP/75002-b7705298", "0?skipoutcid") in new stack
        -- Executing [s at macro-dialout-trunk:10] Set("SIP/75002-b7705298", "DIAL_TRUNK_OPTIONS=Tt") in new stack
      == Begin MixMonitor Recording SIP/75002-b7705298
        -- Executing [s at macro-dialout-trunk:11] Macro("SIP/75002-b7705298", "outbound-callerid|7") in new stack
        -- Executing [s at macro-outbound-callerid:1] ExecIf("SIP/75002-b7705298", "0|SetCallerPres|") in new stack
        -- Executing [s at macro-outbound-callerid:2] ExecIf("SIP/75002-b7705298", "0|Set|REALCALLERIDNUM=75002") in new stack
        -- Executing [s at macro-outbound-callerid:3] GotoIf("SIP/75002-b7705298", "1?normcid") in new stack
        -- Goto (macro-outbound-callerid,s,6)
        -- Executing [s at macro-outbound-callerid:6] Set("SIP/75002-b7705298", "USEROUTCID=") in new stack
        -- Executing [s at macro-outbound-callerid:7] Set("SIP/75002-b7705298", "EMERGENCYCID=") in new stack
        -- Executing [s at macro-outbound-callerid:8] Set("SIP/75002-b7705298", "TRUNKOUTCID=s2") in new stack
        -- Executing [s at macro-outbound-callerid:9] GotoIf("SIP/75002-b7705298", "1?trunkcid") in new stack
        -- Goto (macro-outbound-callerid,s,12)
        -- Executing [s at macro-outbound-callerid:12] ExecIf("SIP/75002-b7705298", "1|Set|CALLERID(all)=s2") in new stack
        -- Executing [s at macro-outbound-callerid:13] ExecIf("SIP/75002-b7705298", "0|Set|CALLERID(all)=") in new stack
        -- Executing [s at macro-outbound-callerid:14] ExecIf("SIP/75002-b7705298", "0|SetCallerPres|prohib_passed_screen") in new stack
        -- Executing [s at macro-dialout-trunk:12] ExecIf("SIP/75002-b7705298", "0|AGI|fixlocalprefix") in new stack
        -- Executing [s at macro-dialout-trunk:13] Set("SIP/75002-b7705298", "OUTNUM=015921256331") in new stack
        -- Executing [s at macro-dialout-trunk:14] Set("SIP/75002-b7705298", "custom=SIP/s2") in new stack
        -- Executing [s at macro-dialout-trunk:15] ExecIf("SIP/75002-b7705298", "1|Set|DIAL_TRUNK_OPTIONS=M(setmusic^none)Tt") in new stack
        -- Executing [s at macro-dialout-trunk:16] Macro("SIP/75002-b7705298", "dialout-trunk-predial-hook|") in new stack
        -- Executing [s at macro-dialout-trunk-predial-hook:1] MacroExit("SIP/75002-b7705298", "") in new stack
        -- Executing [s at macro-dialout-trunk:17] GotoIf("SIP/75002-b7705298", "0?bypass|1") in new stack
        -- Executing [s at macro-dialout-trunk:18] GotoIf("SIP/75002-b7705298", "0?customtrunk") in new stack
        -- Executing [s at macro-dialout-trunk:19] Dial("SIP/75002-b7705298", "SIP/s2/015921256331|300|M(setmusic^none)Tt") in new stack
    Audio is at 219.235.234.238 port 17136
    Adding codec 0x4 (ulaw) to SDP
    Adding non-codec 0x1 (telephone-event) to SDP
    Reliably Transmitting (NAT) to 222.46.18.52:5060:
    INVITE sip:015921256331 at 222.46.18.52 SIP/2.0
    Via: SIP/2.0/UDP 219.235.234.238:5060;branch=z9hG4bK368b5ad8;rport
    From: "s2" <sip:Unknown at 222.46.18.52>;tag=as75543a2d
    To: <sip:015921256331 at 222.46.18.52>
    Contact: <sip:Unknown at 219.235.234.238>
    Call-ID: 5cf71e106209cf65344e24031354fbda at 222.46.18.52
    CSeq: 102 INVITE
    User-Agent: Asterisk PBX
    Max-Forwards: 70
    Date: Fri, 26 Mar 2010 06:16:38 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Supported: replaces
    Content-Type: application/sdp
    Content-Length: 244
    v=0
    o=root 3145 3145 IN IP4 219.235.234.238
    s=session
    c=IN IP4 219.235.234.238
    t=0 0
    m=audio 17136 RTP/AVP 0 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=silenceSupp:off - - - -
    a=ptime:20
    a=sendrecv

    ---
        -- Called s2/015921256331
    gd-branch*CLI> 
    <--- SIP read from 222.46.18.52:5060 --->
    SIP/2.0 403 Forbidden
    Via: SIP/2.0/UDP 219.235.234.238:5060;branch=z9hG4bK368b5ad8;received=58.247.12.18;rport=11028
    From: "s2" <sip:Unknown at 222.46.18.52>;tag=as75543a2d
    To: <sip:015921256331 at 222.46.18.52>
    Contact: <sip:015921256331 at 222.46.18.52:5060>
    Call-ID: 5cf71e106209cf65344e24031354fbda at 222.46.18.52
    CSeq: 102 INVITE
    Max-Forwards: 70
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Supported: timer
    Server: VOS2009 V2.1.1.8


    <------------->
    --- (11 headers 0 lines) ---
    Transmitting (NAT) to 222.46.18.52:5060:
    ACK sip:015921256331 at 222.46.18.52 SIP/2.0
    Via: SIP/2.0/UDP 219.235.234.238:5060;branch=z9hG4bK368b5ad8;rport
    From: "s2" <sip:Unknown at 222.46.18.52>;tag=as75543a2d
    To: <sip:015921256331 at 222.46.18.52>
    Contact: <sip:Unknown at 219.235.234.238>
    Call-ID: 5cf71e106209cf65344e24031354fbda at 222.46.18.52
    CSeq: 102 ACK
    User-Agent: Asterisk PBX
    Max-Forwards: 70
    Content-Length: 0


    ---
        -- SIP/s2-088f72e8 is circuit-busy
      == Everyone is busy/congested at this time (1:0/1/0)
        -- Executing [s at macro-dialout-trunk:20] Goto("SIP/75002-b7705298", "s-CONGESTION|1") in new stack
        -- Goto (macro-dialout-trunk,s-CONGESTION,1)
        -- Executing [s-CONGESTION at macro-dialout-trunk:1] GotoIf("SIP/75002-b7705298", "1?noreport") in new stack
        -- Goto (macro-dialout-trunk,s-CONGESTION,3)
        -- Executing [s-CONGESTION at macro-dialout-trunk:3] NoOp("SIP/75002-b7705298", "TRUNK Dial failed due to CONGESTION - failing through to other trunks") in new stack
        -- Executing [015921256331 at from-internal:6] Macro("SIP/75002-b7705298", "outisbusy|") in new stack
        -- Executing [s at macro-outisbusy:1] Playback("SIP/75002-b7705298", "all-circuits-busy-now|noanswer") in new stack
        -- <SIP/75002-b7705298> Playing 'all-circuits-busy-now' (language 'en')
    Really destroying SIP dialog '5cf71e106209cf65344e24031354fbda at 222.46.18.52' Method: INVITE
        -- Executing [s at macro-outisbusy:2] Playback("SIP/75002-b7705298", "pls-try-call-later|noanswer") in new stack
        -- <SIP/75002-b7705298> Playing 'pls-try-call-later' (language 'en')
        -- Executing [s at macro-outisbusy:3] Macro("SIP/75002-b7705298", "hangupcall") in new stack
        -- Executing [s at macro-hangupcall:1] GotoIf("SIP/75002-b7705298", "1?skiprg") in new stack
        -- Goto (macro-hangupcall,s,4)
        -- Executing [s at macro-hangupcall:4] GotoIf("SIP/75002-b7705298", "1?skipblkvm") in new stack
        -- Goto (macro-hangupcall,s,7)
        -- Executing [s at macro-hangupcall:7] GotoIf("SIP/75002-b7705298", "1?theend") in new stack
        -- Goto (macro-hangupcall,s,9)
        -- Executing [s at macro-hangupcall:9] Hangup("SIP/75002-b7705298", "") in new stack
      == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/75002-b7705298' in macro 'hangupcall'
      == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/75002-b7705298' in macro 'outisbusy'
      == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/75002-b7705298'
      == End MixMonitor Recording SIP/75002-b7705298



  -- 
  Best Regards!

  Aaron Chen 



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