Alcatel Unleashed

Guys , Im going crazy trying to configure an 8012 with OXE.
Because 8770 does no have License for configuration(only Billing ), I have to do this manually using the famous sipconfig-mac file and a Http server.
The Http server is working because when I connect the phone , the phone reads the sipconfig file and downloads the binary files .
The problem is when the phone reaches 5/5 connecting , it doesn´t register with my sip extension that I have configured for the phone (992).
After 5/5 I get a message saying no registration then the phone shows the date and time .
On the the OXE I check the SIP Extension and I see that the IP address is unused.

+-Consult/Modify: IP SIP Extension-----------------------------------------------------------------------+
| |
| Node Number (reserved) : 1 |
| Directory Number : 992 |
| Directory Number : 992 |
| |
| Set Type + SIP extension |
| IP Address : Unused |
| |
+--------------------------------------------------------------------------------------------------------+

I did a trace with wireshark and I found that there is an authentication problem .

If I disable the "Minimal authentication method + None" on the OXE the phone gives me an error message after 5/5 no registration erro and after this it shows me the date and time.
If enable the "Minimal authentication method + Digest" on the OXE the gives me the no registration error but then no time or date .

If you ask yes I have read the manual more than once TC1862 ed.01 (8002/8012 DESKPHONE : MANUAL DEPLOYMENT GUIDE ON OMNIPCX ENTERPRISE).


Can someone please correct my sipconfig file I have tried so many ways and still no luck.
Something is missing I think there is a problem with the authentication name and password.

NOTE : wireshark trace is when I have = "Minimal authentication method + None"
Thanks in advanced

Can´t anybody help here ?

please post the configuration of the sip gateway and sip proxy.. also attach a full wireshark pcap file...

regards...

OK , I will get the the full wireshark pcap file , but I don´t have a sip gateway configured because for SIP Extensions don´t need to be managed by a local SIP Gateway.
That is what I read from the TG00069.
In the phone config file you have the proxy configuration.

The SIP Extension is used only for voice calls, is considered as an Internal SIP user, so it is possible to use
phone features and facilities from the OXE.
It is not necessary to manage the local SIP gateway if you want to use it, only the proxy has to be (for authentication).

ok.
in proxy configuration there is no auth.. realm configured whereas in phone configuration file the realm is "authentication_realm=192.168.150.2"
also i think the username has to be only the usernumber (992). the device url has to be the name or ip address of the oxe.

i think this is the reason of the '403 forbidden' message.

regards...

Reenter the Password of the phone.
In Sipconfig: authentication_password=000000
In Phoneconfig: Password : ****** (0000000)

If your documentation is correct your password is not matching.

Hunf I checked the Password it was Ok, I did the alterations that tgn sugested and still no luck untill I did some more reading and Alcatel Suggest that when we great a sip extension we must no fill in the SIP part .
Só I deleted and created the extension again the phone automatically connected .

Conclusion : With Tgn suggestion and recreating the extension without filling in the SIP part the phone connects perfectly .
Authentication is solved .

TG00069

The declaration of a SIP user binds the information configured in the SIP set with the information stored into the database of the OmniPCX Enterprise.
If you don‟t fill in the SIP part in the OmniPCX Enterprise user configuration, the default values will be : URL User Name = MCDU of the user. URL Domain = SIP domain name of the OmniPCX Enterprise, i.e. the SIP set is considered as registered on the OmniPCX Enterprise.
This is usually exactly what we want so you shouldn‟t modify anything here.
After the creation of the user a corresponding entry will automatically be added to the SIP Dictionnary.
Note: The value for the URL (<username>@<domainname>) configured on the SIP set and in the OmniPCX Enterprise SIP Dictionnary MUST match. This can be an issue if you modified one of these parameters by hand and not the other one.


I have now another problem , I able to call from the 8012 extension , but when I call from another extension to the 8012 the call does no go through .

I get an Invite then a get I 488 message error " not acceptable here "

Did some more reading and it was a codec problem .
I changed to G729 and all is working
Thanks Guys for your Help

Thanks for sharing. Good luck there!

Enviado de meu C6602 usando Tapatalk

nice to hear that you succeed

regards...