Asterisk PBX SIP.conf, which works :)

sip.conf

[general]
udpbindaddr=0.0.0.0:5060

tcpenable=yes
tcpbindaddr=0.0.0.0:5060

tlsenable=yes
tlsbinddir=0.0.0.0:5060

;tlsverify=no
;tlscipher=ALL
;tlsclientmethod=tlsv1
;tlscertfile=/etc/asterisk/keys/asterisk.pem
;tlscafile=/etc/asterisk/keys/ca.crt

;dtlsenable=yes
;dtlsbinddir=0.0.0.0:5060
;dtlsverify=no
;dtlscertfile=/etc/asterisk/keys/asterisk.pem
;dtlscafile=/etc/asterisk/keys/ca.crt

port=5060

transport=udp,tcp
;transport=udp,tcp,ws,wss,tls
;dynamic_exclude_static = yes

context=incoming
allowguest=yes
allowtransfer=yes
alwaysauthreject=yes

;realm=andrius.mobi
;domain=andrius.mobi

;externhost=HOSTNAME-IF-YOU-HAVE-DYNAMIC-IP (DYNDNS)
;externrefresh=90

; NO if you have DNS issues, or install DNSMASQ
srvlookup=yes

tos_sip=cs3
tos_audio=ef
tos_video=af41
tos_text=af41

cos_sip=3
cos_audio=5
cos_video=4
cos_text=3

; http://www.asterisk.org/doxygen/trunk/sip_session_timers.html
session-timers=accept ; ["accept", "originate", "refuse"]
session-minse=90
session-expires=600
session-refresher=uas

maxexpiry=600
minexpiry=120
defaultexpiry=300

vmexten=voicemail

#include sip_allowed_codecs.conf

language=en
relaxdtmf=yes    ; Relax dtmf handling
;trustrpid=no    ; If Remote-Party-ID should be trusted
;sendrpid=yes    ; If Remote-Party-ID should be sent

promiscredir=yes ; If yes, allows 302 or REDIR to non-local SIP address
                 ; Note that promiscredir when redirects are made to the
                 ; local system will cause loops since SIP is incapable
                 ; of performing a "hairpin" call.


; Uncomment if you want to hide Asterisk in USERAGENT header
;useragent=C3412s
;sdpsession=C3412s

sdpowner=root
;usereqphone=yes  ; If yes, ";user=phone" is added to uri that contains
; a valid phone number
dtmfmode=rfc2833

regcontext=sipregistrations
;regextenonqualify=yes

rtptimeout=300
rtpholdtimeout=3600
rtpkeepalive=30

jbenable=yes

allowsubscribe=yes
subscribecontext=outgoing
notifyringing=yes
notifyhold=yes
callcounter=yes
counteronpeer=yes

localnet=192.168.0.0/255.255.0.0  ; RFC 1918 addresses
localnet=10.0.0.0/255.0.0.0  ; Also RFC1918
localnet=172.16.0.0/12  ; Another RFC1918 with CIDR notation
localnet=172.31.255.0/255.255.255.0
localnet=172.32.255.0/255.255.255.0
localnet=172.33.255.0/255.255.255.0
localnet=172.34.255.0/255.255.255.0
localnet=172.35.255.0/255.255.255.0
localnet=169.254.0.0/255.255.0.0  ; Zero conf local network

canreinvite=no
directmedia=no ; update,nonat
nat=auto_force_rport,auto_force_comedia
icesupport=no
;qualify=2000 ; 2000 msec = 2 seconds
qualify=no

; as per http://forum.snom.com/index.php?showtopic=5248
prematuremedia=no
progressinband=yes

registertimeout=20
registerattempts=0

limitonpeers=yes

;#include sip_registrations.conf

[authentication]

;#include sip_profiles.conf
;#include sip_trunks.conf
;#include sip_trunks_extra.conf
;#include sip_accounts.conf

[natted-phone-no-qualify]
transport=tcp,udp
type=friend
progressinband=yes
context=test
nat=force_rport,comedia
canreinvite=no
directmedia=no
host=dynamic
qualify=no
subscribemwi=yes
disallow=all
; allow=amr
allow=g722
allow=alaw
;allow=silk8
;allow=g729
allow=gsm

[natted-phone](natted-phone-no-qualify)
qualify=8000

[account1](natted-phone)
secret=Fawre5Ru8hUfruTAfReC

[account2](natted-phone)
secret=juCuDetaduxechameph6

sip_allowed_codecs.conf

disallow=all
allow=g722
allow=alaw
allow=ulaw
;allow=silk8
;allow=g729
allow=gsm