[Asterisk-Users] [suse-isdn] Major Problems UTStarcom F1000 registering -- pls help

Wed Jan 11 06:45:51 MST 2006

Hi there,
I am trying desperatly to register my WiFi Phone UTStarcomm F1000 with 
my asterisk server. I already changed the name of the user to 
"anonymous" since it looks like the phone sends that name. The WiFi 
Phone's IP is 192.168.1.217, the asterisk server's IP is 192.168.1.200
What is it that I am missing? Any help very much appreciated!!!

The error message I get is:
Jan 11 13:49:30 NOTICE[24024]: chan_sip.c:10817 handle_request_register: 
Registration from '"anonymous" <sip:anonymous at 192.168.1.200>' failed for 
'192.168.1.217' - Username/auth name mismatch

extract of [sip.conf]:
...................................
[UTStarcomF1000]        
type=friend
bindport=5060
username=anonymous
;fromuser=anonymous
secret=welcome
mailbox=1000
canreinvite=yes
context=sipout     
insecure=very
defaultip=192.168.1.217
host=dynamic
qualify=yes
nat=no
;auth=anonymous:welcome at 192.168.1.217
dtmfmode=rcfa2833
....................................................

*CLI> sip show peers
Name/username              Host            Dyn Nat ACL Port     Status
UTStarcomF1000/anonymous   (Unspecified)    D          0        UNKNOWN
omp-out-4321/419941xxxxx  212.117.200.148      N      5060     OK (64 ms)
omp-out-5211/419941xxxxx  212.117.200.148      N      5060     OK (64 ms)
omp-out-5200/419941xxxxx  212.117.200.148      N      5060     OK (64 ms)
web-de/xxxxx               217.72.200.89        N      5060     OK (64 ms)
sipgate-out/19xxxxx        217.10.79.9          N      5060     OK (68 ms)
8 sip peers [5 online , 3 offline]

*CLI> sip debug ip 192.168.1.217
SIP Debugging Enabled for IP: 192.168.1.217

*CLI> sip show registry
Host                            Username       Refresh State
sip.web.de:5060                 xxxxx              105 Registered
sipgate.de:5060                 19xxxxx            105 Registered

And here the debug message:
.....................................................................
Jan 11 14:28:38 NOTICE[24049]: chan_sip.c:10817 handle_request_register: 
Registration from '"anonymous" <sip:anonymous at 192.168.1.200
 >' failed for '192.168.1.217' - Username/auth name mismatch
Scheduling destruction of call '129842916 at 192.168.1.217' in 15000 ms

<-- SIP read from 192.168.1.217:5060:
REGISTER sip:192.168.1.200:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.217:5060;rport;branch=z9hG4bK3499846672
From: "anonymous" <sip:anonymous at 192.168.1.200>;tag=787472657
To: "anonymous" <sip:anonymous at 192.168.1.200>
Call-ID: 129842916 at 192.168.1.217
CSeq: 90 REGISTER
Contact: <sip:anonymous at 192.168.1.217:5060>;action=proxy
max-forwards: 70
expires: 60
user-agent: UTSTARCOM F1000/Device ID-0007ba253307
Content-Length: 0

--- (11 headers 0 lines)---
Using latest REGISTER request as basis request
Sending to 192.168.1.217 : 5060 (non-NAT)
Transmitting (no NAT) to 192.168.1.217:5060:
SIP/2.0 404 Not found
Via: SIP/2.0/UDP 
192.168.1.217:5060;rport;branch=z9hG4bK3499846672;received=192.168.1.217
From: "anonymous" <sip:anonymous at 192.168.1.200>;tag=787472657
To: "anonymous" <sip:anonymous at 192.168.1.200>;tag=as750293ee
Call-ID: 129842916 at 192.168.1.217
CSeq: 90 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:anonymous at 192.168.1.200>
Content-Length: 0
............................................................................
and here is the SIP and RTP Configuration of the phone: (STUN is turned 
off) (I hope this will be transmitted to the list as well since it is a 
paste from the Web Interfrace. In short it says:
Sip Terminal Use Outbound Proxy> yes
sip terminal use register> yes
sip outbound server domain name> server.x.y
sip outbound server ip address> 192.168.1.200
sip outbound server port> 5060
sip rigister server domain name> server.x.y
sip register server ip address> 192.168.1.200
sip register server port> 5060
sip authentication string> anonymous
sip user name> anonymous
sip password> welcome
sip terminal port> 5060
sip terminal use null packet> no
both sip proxy and regisister server use IP> yes
dns query type> yes
set registration duration> 60 sec
terminal audio rtp port> 10120
terminal audio packetize time> 20 milliseconds

*SIP Terminal Use Outbound Proxy:*

No

Yes
*SIP Terminal Use Register: *

No

Yes
*SIP Outbound Server Domain Name:*

*SIP Outbound Server IP Address:*

*SIP Outbound Server Port:*

*SIP Register Server Domain Name:*

*SIP Register Server IP Address:*

*SIP Register Server Port:*

*SIP Authentication String:*

*SIP User Name:*

*SIP Password:*

*SIP Terminal Port:*

*SIP Terminal Use Null Packet:*

No

Yes
*SIP Terminal Use DNS:*

Both SIP Proxy And Register Servers Use IP
Register Server Uses DNS And SIP Proxy Uses IP
Register Server Uses IP And SIP Proxy Server Uses DNS
Both Register And SIP Proxy Servers Use DNS
*DNS Query Type: *

None SRV

SRV
*Set Registration Duration:*

(sec)
*Terminal Audio RTP Port:*

*Terminal Audio Packetize Time:*

(millisecond)

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