[Asterisk-Users] SIP/2.0 403 Forbidden

[Silvio Schneider]

Hi all,

I have been worriyng and googling a lot but I can't find my mistake.
I am trying to regiter an X-Lite Softphone to Asterisk, but
I am getting a SIP/2.0 403 Forbidden response:

SEND TIME: 10157385
SEND >> 10.100.249.12:5060
REGISTER sip:10.100.249.12 SIP/2.0
Via: SIP/2.0/UDP
10.100.249.86:5060;rport;branch=z9hG4bKFAC1B6F2B5414EE9855696A09A83FB22
From: Tester <sip:1000 at 10.100.249.12>;tag=3354744682
To: Tester <sip:1000 at 10.100.249.12>
Contact: "Tester" <sip:1000 at 10.100.249.86:5060>
Call-ID: 6A1715C235994196A7739A624B6D0C41 at 10.100.249.12
CSeq: 4806 REGISTER
Expires: 1800
Max-Forwards: 70
User-Agent: X-Lite release 1103m
Content-Length: 0


RECEIVE TIME: 10157385
RECEIVE << 10.100.249.12:5060
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP
10.100.249.86:5060;branch=z9hG4bKFAC1B6F2B5414EE9855696A09A83FB22
From: Tester <sip:1000 at 10.100.249.12>;tag=3354744682
To: Tester <sip:1000 at 10.100.249.12>;tag=as7ae925e2
Call-ID: 6A1715C235994196A7739A624B6D0C41 at 10.100.249.12
CSeq: 4806 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:1000 at 10.100.249.12>
Content-Length: 0 

------------------------------
MY config is:
/etc/asterisk/sip.conf :

; Note: If your SIP devices are behind a NAT and your Asterisk
;  server isn't, try adding "nat=1" to each peer definition to
;  solve translation problems.
[general]
port = 5060           ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0    ; Address to bind to (all addresses on machine)
disallow=all
allow=ulaw
allow=alaw
context = from-sip-external ; Send unknown SIP callers to this context
callerid = Unknown

#include sip_nat.conf
#include sip_custom.conf
#include sip_additional.conf

[1000]
username=1000
secret=abc123
context=mytest
host=dynamic
-----------------
/etc/asterisk/extensions.conf :

[general]
static=yes
writeprotect=yes

;Suport phones
;SUPPORTPHONES=SIP/2205&SIP/2206&SIP/2207&SIP/2208&SIP/2209

[globals]
XLITE=SIP/1000

[mytest]
exten => 1367,1Dial(SIP/1000)
exten => 2890,1,Wait(2)
exten => 2890,2,Answer
exten => 2890,3,Playback(demo-echotest)
exten => 2890,4,Echo()
exten => 2468,1,Dial($XLITE) 
-------------
Some startup messages from Asterisk:
[pbx_config.so] => (Text Extension Configuration)
  == Parsing '/etc/asterisk/extensions.conf': Found
  == Setting global variable 'XLITE' to 'SIP/1000'
    -- Registered extension context 'mytest'
    -- Added extension '1367' priority 1 to mytest
    -- Added extension '2890' priority 1 to mytest
    -- Added extension '2890' priority 2 to mytest
    -- Added extension '2890' priority 3 to mytest
    -- Added extension '2890' priority 4 to mytest
    -- Added extension '2468' priority 1 to mytest 
----------------
Someone has an idea what is causing the "SIP/2.0 403 Forbidden"?
Please help. 

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