[asterisk-users] sip authorization problem

[Ryan Murray]

Hi,

I am trying to setup a simple home voip service w/ *
I have compiled and installed the svn source
as a first step I am trying to configure SIP for inside my network.
I have a handful of softphones and a few hardphones that I want to all be
able to call each other
I have configured users.conf with a single softphone(kphone) and have tried
calling itself (ext 6000) and the demo
from the asterisk install (ext 500), both give me a 401 Unauthorized error
below I have included some debugging output and all the important config
files

*******part of extensions.conf that was added by asterisk-gui (svn)*******
[asterisk_guitools]
exten = executecommand,1,System(${command})
exten = executecommand,n,Hangup()
exten = record_vmenu,1,Answer
exten = record_vmenu,n,Playback(vm-intro)
exten = record_vmenu,n,Record(${var1})
exten = record_vmenu,n,Playback(vm-saved)
exten = record_vmenu,n,Playback(vm-goodbye)
exten = record_vmenu,n,Hangup
exten = play_file,1,Answer
exten = play_file,n,Playback(${var1})
exten = play_file,n,Hangup
hasbeensetup = Y

[DID_trunk_1]
include = default

[numberplan-custom-1]
plancomment = DialPlan1
include = default
include = parkedcalls

[timebasedrules]
*******part of extensions.conf that was added by asterisk-gui (svn)*******


*******part of users.conf that was added by asterisk-gui (svn)*******
[trunk_1]
allow = all
context = DID_trunk_1
dialformat = ${EXTEN:1}
hasexten = no
hasiax = yes
hassip = no
host = iax2.fwdnet.net
port = 4569
registeriax = yes
registersip = no
secret = rycort4e
trunkname = Custom - fwd
trunkstyle = customvoip
username = 788694

[6000]
callwaiting = yes
cid_number = 6000
fullname = proton
hasagent = yes
hasdirectory = no
hasiax = no
hasmanager = no
hassip = yes
hasvoicemail = yes
host = dynamic
mailbox = 6000
secret = proton
threewaycalling = yes
vmsecret = 1234
registeriax = no
registersip = yes
canreinvite = yes
nat = no
dtmfmode = inband
disallow = all
allow = all
context = numberplan-custom-1
*******part of users.conf that was added by asterisk-gui (svn)*******

the rest are straight from the samples that got installed at build time

*******************debugging output*************
*CLI> sip show peers
Name/username              Host            Dyn Nat ACL Port     Status
6000/6000                  192.168.0.101    D          5060     Unmonitored
1 sip peers [Monitored: 0 online, 0 offline Unmonitored: 1 online, 0
offline]

debugging output from calling 500
<--- SIP read from 192.168.0.101:5060 --->
INVITE sip:500 at 192.168.0.102 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.101;branch=z9hG4bK78B64BD
CSeq: 2212 INVITE
To: <sip:500 at 192.168.0.102>
Content-Type: application/sdp
From: "6000" <sip:6000 at 192.168.0.102>;tag=327F7192
Call-ID: 2096168429 at 192.168.0.101
Subject: sip:6000 at 192.168.0.102
Content-Length: 230
User-Agent: kphone/4.2
Contact: "6000" <sip:6000 at 192.168.0.101;transport=udp>

v=0
o=username 0 0 IN IP4 192.168.0.101
s=The Funky Flow
c=IN IP4 192.168.0.101
t=0 0
m=audio 33322 RTP/AVP 0 97 8 3
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30

<------------->
--- (11 headers 11 lines) ---
  == Using TOS bits 0
  == Using CoS mark 5
Sending to 192.168.0.101 : 5060 (no NAT)
Using INVITE request as basis request - 2096168429 at 192.168.0.101
No user '6000' in SIP users list
Found peer '6000' for '6000' from 192.168.0.101:5060

<--- Reliably Transmitting (no NAT) to 192.168.0.101:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.101;branch=z9hG4bK78B64BD;received=192.168.0.101
From: "6000" <sip:6000 at 192.168.0.102>;tag=327F7192
To: <sip:500 at 192.168.0.102>;tag=as6b3f431e
Call-ID: 2096168429 at 192.168.0.101
CSeq: 2212 INVITE
User-Agent: Asterisk PBX SVN-trunk-r81159
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5f450cef"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '2096168429 at 192.168.0.101' in 32000 ms
(Method: INVITE)

<--- SIP read from 192.168.0.101:5060 --->
ACK sip:500 at 192.168.0.102 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.101;branch=z9hG4bK78B64BD
CSeq: 2212 ACK
To: <sip:500 at 192.168.0.102>;tag=as6b3f431e
From: "6000" <sip:6000 at 192.168.0.102>;tag=327F7192
Call-ID: 2096168429 at 192.168.0.101
Content-Length: 0
User-Agent: kphone/4.2
Contact: "6000" <sip:6000 at 192.168.0.101;transport=udp>


<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '2096168429 at 192.168.0.101' Method: ACK

*******************debugging output*************


thanks in advanced
Ryan
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