thx for the answer, and now to the details:
The first thing you need to do is get all the "401" & "403" response codes fixed.
I'm seeing that your VoIp provider is not rejecting(401 code) then accepting. Educate me again on some things:
How does you machine communicate with the "outside" world? From LMCE to a router, to a DSL/Cable modem and are you using a static IP? If dynamic IP have you registered with someone like dyndns.org or are you just dependant on your Internet Service Provider?
LMCE box - external nic -> router -> adsl modem
IP setting dynamic and i also have dyndns setted on the external router.
tcp ipv4 443 core_input Delete
tcp ipv4 2000 core_input Delete
udp ipv4 2000 core_input Delete
udp ipv4 4569 core_input Delete
udp ipv4 5060 core_input Delete
udp ipv4 10001 to 20000 core_input Delete
Both external sip providers (sipgate and voipcheap) uses udp 5060.
On the spa, if I am correct I'm seeing 401 & 403s, you need to check that the user name & password are the same as what LMCE has entered in its database. It could be that it is and the database is not storing the passwords the same. I have had this issue before in the past, a corrupt database.....
I'm also seeing that the spa is trying to communicate on port 5061, is this the only port in the spa that is assigned, if so you need to have it at beginning at 5060 and end with blank or at least 5061.
I'm assuming that it is set for 5061, asterisk normally starts with 5060....
The spa3102 is connected in the internal network, as the log shows (192.168.80.30).
The spa configuration has two parts: pstn and line 1.
Line 1 is defined as extension, and registers as a sip phone.
The pstn is defined with port 5061 and user and password were checked and are the same. The whole configuration of the spa3102 is the same as the wiki, http://wiki.linuxmce.org/index.php/Linksys_SPA3102
, including username and password
Display Name - Unknown Caller (this is what is displayed when a call comes in without caller ID info)
UserID - spa3102 (this is what you set when creating the Phone Line in the LinuxMCE webmin)
Password - lmce (this is what you set when creating the Phone Line in the LinuxMCE webmin)
In the 8.10 release, it worked well, with the new release, it does not authenticates, and i'm not finding out the reason, as userid, password, port and settings are the same.
Had to look your post over pretty quick as I have to leave for church but will check it again when I return but in the passing time check for all your 401 & 403 and try to correct them, as I say it's a authentication issue. I have a website that will give you the sip responses at: http://en.wikipedia.org/wiki/List_of_SIP_response_codes and you can always go to asterisk web site for additional info.
The other issues are:
cisco sccp extension 203 calling sip on spa3102 extension 204: 204 rings, i can hear from 203, but 203 does not hears from 204
sip on spa3102 extension 204 calling cisco sccp extension 203: always busy.
Well, any help is welcome in order to solve it all.
BTW, where should i define the dialplans according to the trunk?
Best regards and thx again.