[Asterisk-Users] Major Problems UTStarcom F1000 registering --
pls help
KokMeng Loh
kokmeng at asgent-tech.com
Thu Jan 12 18:14:22 MST 2006
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 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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