Christoph Merk
2006-Jan-11 06:45 UTC
[Asterisk-Users] [suse-isdn] Major Problems UTStarcom F1000 registering -- pls help
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@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@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@192.168.1.200
>' failed for '192.168.1.217' - Username/auth name mismatch
Scheduling destruction of call '129842916@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@192.168.1.200>;tag=787472657
To: "anonymous" <sip:anonymous@192.168.1.200>
Call-ID: 129842916@192.168.1.217
CSeq: 90 REGISTER
Contact: <sip:anonymous@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@192.168.1.200>;tag=787472657
To: "anonymous" <sip:anonymous@192.168.1.200>;tag=as750293ee
Call-ID: 129842916@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@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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KokMeng Loh
2006-Jan-12 18:14 UTC
[Asterisk-Users] Major Problems UTStarcom F1000 registering -- pls help
Hi, You can try changing your section name ([UTStarcomF1000]) to the user name, i.e. [anonymous]. I also noticed that you had a typo in the 'dtmfmode' line; it should be 'rfc2833' and not 'rfca2833'. -kokmeng. Christoph Merk wrote:> 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@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@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@192.168.1.200 > >' failed for '192.168.1.217' - Username/auth name mismatch > Scheduling destruction of call '129842916@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@192.168.1.200>;tag=787472657 > To: "anonymous" <sip:anonymous@192.168.1.200> > Call-ID: 129842916@192.168.1.217 > CSeq: 90 REGISTER > Contact: <sip:anonymous@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@192.168.1.200>;tag=787472657 > To: "anonymous" <sip:anonymous@192.168.1.200>;tag=as750293ee > Call-ID: 129842916@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@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) > > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >