[Asterisk-Users] 403 FORBIDDEN Help!

[Bartosz Jozwiak]

Some more details....
When I am dailing an extension on Asterixsk PBX


Maybe it will help some how....


66.178.36.15 -> 66.178.36.220   SIP/2.0
66.178.36.15 -> 66.178.36.220   SIP/2.0/UDP 66.178.36.15:5060;branch=2b0b0ed72ad04c5615dcab707e0fbe4a.4
66.178.36.15 -> 66.178.36.220   SIP/2.0/UDP 66.178.36.15:5060;branch=49cec9c54d88703154f83cb6acc7b397.2
66.178.36.15 -> 66.178.36.220   SIP/2.0/UDP 66.178.37.103:5060
:V:#.Af.8.6."0.P`> f.8.6..P`:  udp i4 (DF)
E@@iB$B$qINVITE sip:1002 at sip.greentone.com SIP/2.0
Via: SIP/2.0/UDP 66.178.36.220:5060;branch=z9hG4bK44889137
From: "1002 at sip.greentone.com" <sip:asterisk at 66.178.36.220>;tag=as4046c5f2
To: <sip:1002 at sip.greentone.com>
Contact: <sip:asterisk at 66.178.36.220>
Call-ID: 0d94bac54d736a42709e590a6dee22d8 at 66.178.36.220
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 238

v=0
o=root 18960 18960 IN IP4 66.178.36.220
s=session
c=IN IP4 66.178.36.220
t=0 0
m=audio 15100 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
66.178.36.220 -> 66.178.36.15   SIP/2.0
66.178.36.220 -> 66.178.36.15   SIP/2.0/UDP 66.178.36.220:5060;branch=z9hG4bK44889137
:V:$."Qf.8.6."0.P`> f.8.6..P`:  udp i4 (DF)
E@@iB$B$qINVITE sip:1002 at sip.greentone.com SIP/2.0
Via: SIP/2.0/UDP 66.178.36.220:5060;branch=z9hG4bK44889137
From: "1002 at sip.greentone.com" <sip:asterisk at 66.178.36.220>;tag=as4046c5f2
To: <sip:1002 at sip.greentone.com>
Contact: <sip:asterisk at 66.178.36.220>
Call-ID: 0d94bac54d736a42709e590a6dee22d8 at 66.178.36.220
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 238

v=0
o=root 18960 18960 IN IP4 66.178.36.220
s=session
c=IN IP4 66.178.36.220
t=0 0
m=audio 15100 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
66.178.36.220 -> 66.178.36.15   SIP/2.0
66.178.36.220 -> 66.178.36.15   SIP/2.0/UDP 66.178.36.220:5060;branch=z9hG4bK44889137
:V:%.wPIf.8.6..P`> f.8.6."0.P`:  udp 02 (DF)
EJ@@kTB$B$6SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 66.178.36.220:5060;branch=z9hG4bK44889137
To: <sip:1002 at sip.greentone.com>;tag=2ff28aef
From: "1002 at sip.greentone.com" <sip:asterisk at 66.178.36.220>;tag=as4046c5f2
Call-ID: 0d94bac54d736a42709e590a6dee22d8 at 66.178.36.220
CSeq: 102 INVITE
Content-Length: 0

66.178.36.15 -> 66.178.36.220   SIP/2.0 403 Forbidden
66.178.36.15 -> 66.178.36.220   SIP/2.0/UDP 66.178.36.220:5060;branch=z9hG4bK44889137
:V:%.wiCf.8.6."0.P`> f.8.6..P`:  udp 81 (DF)
E@@kB$B$vACK sip:1002 at sip.greentone.com SIP/2.0
Via: SIP/2.0/UDP 66.178.36.220:5060;branch=z9hG4bK44889137
From: "1002 at sip.greentone.com" <sip:asterisk at 66.178.36.220>;tag=as4046c5f2
To: <sip:1002 at sip.greentone.com>;tag=2ff28aef
Contact: <sip:asterisk at 66.178.36.220>
Call-ID: 0d94bac54d736a42709e590a6dee22d8 at 66.178.36.220
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

  ----- Original Message ----- 
  From: Bartosz Jozwiak 
  To: asterisk-users at lists.digium.com 
  Sent: Monday, August 18, 2003 11:50 AM
  Subject: Re: [Asterisk-Users] 403 FORBIDDEN Help!



  Asterix PBX is loggin to Vocal and the extension number is also loggin on the same vocal server.
  I cannot make it work.... :(

    ----- Original Message ----- 
    From: Josh Roberson 
    To: asterisk-users at lists.digium.com 
    Sent: Monday, August 18, 2003 11:43 AM
    Subject: Re: [Asterisk-Users] 403 FORBIDDEN Help!


    I'm new too, but alot of my 403 forbidden messages when adding extensions were due to context rules..   make sure that the client dialing the extension is included in the same context your extension is in.  

    just my thoughts on it, as it resolved a lot of 403 errors for me.


      ----- Original Message ----- 
      From: Bartosz Jozwiak 
      To: Asterisk-Users at lists.digium.com 
      Sent: Monday, August 18, 2003 9:31 AM
      Subject: [Asterisk-Users] 403 FORBIDDEN Help!


      Hello,

      I have a question.
      I set up an extension to 1234

      exten => 1234,1,Dial(SIP/1234 at sip.greentone.com:5060)

      And when I dial that extension I got in SIP message "403 FORBIDDEN"
      Can somebody tell me why I cannot call that extension? When I am not using Asterisk I can call that extension without any problems.
      My SIP proxy is VOCAL.
      I am new here so I do not know a lot yet.

      Thank you in advance.

      Bartosz Jozwiak
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