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)