asterisk users - Dec 2009 - Error : SIP/2.0 401 Unauthorized

Hi Friends,

need to help.

*I have problem about sip : SIP/2.0 401 Unauthorized*
Is it require to nathelper module in kamailio ?
*what can i write kamailio.cfg file when kamailio and Asterisk on same
network?*

Scenario is like as :
-----------------------------
1) kamailio server on 172.18.100.74
kamailio.cfg ( nathelpler module )
-----------------
loadmodule "nathelper.so"
modparam("nathelper", "rtpproxy_sock",
"unix:/var/run/rtpproxy.sock")
modparam("nathelper", "natping_interval", 60)
modparam("nathelper", "ping_nated_only", 1)
modparam("nathelper", "rtpproxy_disable_tout", 60)
modparam("nathelper", "rtpproxy_tout", 1)
modparam("nathelper", "rtpproxy_retr", 5)
modparam("nathelper", "sipping_method", "OPTIONS")
modparam("nathelper", "received_avp",
"$avp(i:801)")

2) Asterisk server on 172.18.100.65
sip.conf
-----------
[rajnikant]
nat=yes
disallow=all
allow=alaw
allow=ulaw
allow=gsm
type=peer
context=default
host=172.18.100.74
fromdomain=rajnikant.net
mailbox=user at context

-- 
Thanks and  Regards
Rajnikant Vanza
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Hi all,

*I have problem about sip : SIP/2.0 401 Unauthorized

*domain = rajnikant.net ( its ipaddress is 172.18.100.74 - kamailio server )
when i have call from 111 at rajnikant.net user to 222 at rajnikant.net this
error
occured *SIP/2.0 401 Unauthorized*

Asterisk server on 172.18.100.65
sip.conf
-----------
[rajnikant]
nat=yes
disallow=all
allow=alaw
allow=ulaw
allow=gsm
type=peer
context=default
host=172.18.100.74
fromdomain=rajnikant.net
mailbox=user at context

*Asterisk CLI with sip set debug on *

Scheduling destruction of SIP dialog '
7d358b001cba050d2ff61d9b0e1caae1 at 172.18.100.71' in 32000 ms (Method:
OPTIONS)
Really destroying SIP dialog '
6086988b002aaea400c87044747346d3 at 172.19.100.103' Method: OPTIONS
sabseserver1*CLI>
<--- SIP read from UDP://172.18.100.74:5060 --->
INVITE sip:*89 at voicemailserver.com:5060<http://89 at
voicemailserver.sabsebolopbx.com:5060>SIP/2.0
Record-Route: <sip:172.18.100.74;lr;ftag=613522939;nat=yes>
Via: SIP/2.0/UDP 172.18.100.74;branch=z9hG4bKd50e.c5e36c53.0
Via: SIP/2.0/UDP 172.18.100.74:5061
;received=172.18.100.74;rport=5061;branch=z9hG4bK0685AC676DAF2C251C89805C4DC4E419
From: 111 <sip:111 at rajnikant.net:5061>;tag=613522939
To: <sip:222 at rajnikant.net <sip%3A222 at rajnikant.net>>
Contact: <sip:111 at 172.18.100.74:5061>
Call-ID: 1DCD3AC9-81CC-5077-7864-93C06BEDDAFB at 172.18.100.74
CSeq: 6259 INVITE
Max-Forwards: 69
Content-Type: application/sdp
User-Agent: X-Lite release 1105d
Content-Length: 276

v=0
o=111 2083978771 2083978817 IN IP4 172.18.100.74
s=X-Lite
c=IN IP4 172.18.100.74
t=0 0
m=audio 46058 RTP/AVP 0 8 3 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=nortpproxy:yes
<------------->
--- (13 headers 13 lines) ---
  == Using SIP RTP CoS mark 5
Sending to 172.18.100.74 : 5060 (NAT)
Using INVITE request as basis request -
1DCD3AC9-81CC-5077-7864-93C06BEDDAFB at 172.18.100.74
Found user '111' for '111'
sabseserver1*CLI>
<--- Reliably Transmitting (NAT) to 172.18.100.74:5060 --->
*SIP/2.0 401 Unauthorized*
Via: SIP/2.0/UDP
172.18.100.74;branch=z9hG4bKd50e.c5e36c53.0;received=172.18.100.74
Via: SIP/2.0/UDP 172.18.100.74:5061
;received=172.18.100.74;rport=5061;branch=z9hG4bK0685AC676DAF2C251C89805C4DC4E419
From: 111 <sip:111 at rajnikant.net:5061>;tag=613522939
To: <sip:222 at rajnikant.net <sip%3A222 at
rajnikant.net>>;tag=as7c7ddf69
Call-ID: 1DCD3AC9-81CC-5077-7864-93C06BEDDAFB at 172.18.100.74
CSeq: 6259 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="sabse1",
nonce="3f2f6c32"
Content-Length: 0


Thanks in advance.

-- 
Thanks and  Regards
Rajnikant Vanza
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