Asterisk 1.8 realtime - SIP and SPA3102 not working.

On 8.10, i had three trunks working: spa3102 for landline sipgate.de voipcheap On 10.04 none is working: spa3102 does not register, according asterisk log. voipcheap registers but sound one way only. sipgate is also not registering. I'm sending the original configuration files from the 8.10 for analisys, as i'm not being able to make them work on 10.04.

The sip.conf from the working config: Code:

[sip_general_additional.conf

bindport = 5060           ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0    ; Address to bind to (all addresses on machine)
alwaysauthreject=yes ; required by fail2ban
disallow=all
allow=ulaw
allow=alaw
; If you need to answer unauthenticated calls, you should change this
; next line to 'from-trunk', rather than 'from-sip-external'.
; You'll know this is happening if when you call in you get a message
; saying "The number you have dialed is not in service. Please check the
; number and try again."
context = from-sip-external ; Send unknown SIP callers to this context
callerid = Unknown
tos=0x68

; Reported as required for Asterisk 1.4
notifyringing=yes
notifyhold=yes
limitonpeers=yes

; enable and force the sip jitterbuffer. If these settings are desired
; they should be set in the sip_general_custom.conf file as this file
; will get overwritten during reloads and upgrades.
;
; jbenable=yes
; jbforce=yes

; #, in this configuration file, is NOT A COMMENT. This is exactly
; how it should be.
#include sip_general_custom.conf
#include sip_nat.conf
#include sip_registrations_custom.conf
#include sip_registrations.conf
#include sip_custom.conf
#include sip_additional.conf


sip_additional.conf
Code:

[sip.voipcheap.com](general]
#include)
type=friend
qualify=yes
insecure=invite,port
host=sip.voipcheap.com
dtmfmode=auto
disallow=all
context=from-pstn
allow=ulaw
allow=alaw
allow=g729

[sipgate]
username=username
type=peer
secret=xxxxxxxxxxxxxxxxxxx
qualify=yes
port=5060
nat=yes
insecure=invite,port
host=sipgate.de
fromuser=username
fromdomain=sipgate.de
dtmfmode=auto
disallow=all
context=from-trunk
canreinvite=yes
authuser=username
allow=alaw
allow=ulaw
allow=g726
allow=gsm
allow=g729
call-limit=50

[sipgate_de]
username=username
type=friend
secret=xxxxxxxxxxxxxxxxxx
qualify=yes
port=5060
insecure=invite,port
host=sipgate.de
dtmfmode=auto
disallow=all
context=from-trunk
canreinvite=yes
allow=alaw
allow=ulaw
allow=g726
allow=gsm
allow=g729

[spa3102]
username=spa3102
type=friend
secret=lmce
qualify=yes
port=5061
nat=never
incominglimit=1
host=dynamic
dtmfmode=auto
context=from-trunk
canreinvite=no
allow=ulaw
call-limit=50

[voipcheap]
username=username
type=friend
sendrpid=yes
secret=xxxxxxxxxxxxxxx
qualify=yes
port=5060
nat=yes
insecure=invite,port
host=sip.voipcheap.com
fromuser=username
fromdomain=sip.voipcheap.com
dtmfmode=auto
disallow=all
context=from-pstn
canreinvite=yes
authuser=username
allow=ulaw
allow=ulaw
allow=g729
call-limit=50


sip_registrations.conf
Code:

register=usname:xxxxxxxxxx@sip.voipcheap.com/2062036594
register=username:xxxxxxxxxx@sipgate.de/054138594676


sip_nat.conf
Code:

nat=yes
externip=myhostdyndns.homeunix.org
externrefresh=10
localnet=192.168.80.0/255.255.255.0


localprefixes.conf
Code:

[trunk-4]
rule1=00+XXXXXXX.

[trunk-2]
rule1=XXXXXXXX
rule2=08+08|00XXXXX.
rule3=005521|XXXXXXXX
rule4=031+0055|XXXXXXXXXX
rule5=031+0|XXXXXXXXXX
rule6=031+XXXXXXXXXX
rule7=031+011XXXXXXXXX

[trunk-3]
rule1=00+XXXXXXX.