[trixbox1.localdomain ~]# cat /etc/asterisk/sip.conf            
;--------------------------------------------------------------------------------;
; Do NOT edit this file as it is auto-generated by FreePBX. All modifications to ;
; this file must be done via the web gui. There are alternative files to make    ;
; custom modifications, details at: http://freepbx.org/configuration_files       ;
;--------------------------------------------------------------------------------;
;

[general]

; These files will all be included in the [general] context
;
#include sip_general_additional.conf

;sip_general_custom.conf is the proper file location for placing any sip general 
;options that you might need set. For example: enable and force the sip jitterbuffer. 
;If these settings are desired they should be set the sip_general_custom.conf file.
;
; jbenable=yes
; jbforce=yes
;
;It is also the proper place to add the lines needed for sip nat'ing when going
;through a firewall.  For nat'ing you'd need to add the following lines: 
; nat=yes , externip= , localhost= , and optionally fromdomain= .
;
#include sip_general_custom.conf

;sip_nat.conf is here for legacy support reasons and for those that upgrade 
;from previous versions.  If you have this file with lines in it please make
;sure they are not duplicated in sip_general_custom.conf, if so remove them 
;from sip_nat.conf as sip_general_custom.conf will have precedence.
#include sip_nat.conf

;sip_registrations_custom.conf is for any customizations you might need to do to
;the automatically generated registrations that FreePBX makes.
;
#include sip_registrations_custom.conf
#include sip_registrations.conf

; These files should all be expected to come after the [general] context
;
#include sip_custom.conf
#include sip_additional.conf

;sip_custom_post.conf If you have extra parameters that are needed for a 
;extension to work to for example, those go here.  So you have extension 
;1000 defined in your system you start by creating a line [1000](+) in this 
;file.  Then on the next line add the extra parameter that is needed.  
;When the sip.conf is loaded it will append your additions to the end of 
;that extension. 
;
#include sip_custom_post.conf


# cat /etc/asterisk/sip_custom.conf     

# cat /etc/asterisk/sip_general_additional.conf 

vmexten=000
disallow=all
allow=ulaw
allow=alaw
allow=h263
allow=h263a
allow=h264
videosupport=yes
context=from-sip-external
callerid=Unknown
notifyringing=yes
notifyhold=yes
limitonpeers=yes
tos_sip=cs3
tos_audio=ef
tos_video=af41

# cat /etc/asterisk/sip_general_custom.conf 

# cat /etc/asterisk/sip_additional.conf 

[200004908502121277]

[990]
deny=0.0.0.0/0.0.0.0
type=friend
secret=z
qualify=yes
port=5060
pickupgroup=
permit=0.0.0.0/0.0.0.0
nat=yes
mailbox=990@device
host=dynamic
dtmfmode=info
dial=SIP/990
context=access
canreinvite=no
callgroup=
callerid=device <990>
accountcode=
call-limit=50

[993]
deny=0.0.0.0/0.0.0.0
disallow=all
type=friend
secret=z
qualify=yes
port=5060
pickupgroup=
permit=0.0.0.0/0.0.0.0
nat=no
mailbox=993@device
host=dynamic
dtmfmode=info
dial=SIP/993
context=access
canreinvite=no
callgroup=
callerid=device <993>
allow=g729
allow=ulaw
accountcode=
call-limit=50

[995]
deny=0.0.0.0/0.0.0.0
disallow=all
type=friend
secret=z
qualify=yes
port=5060
pickupgroup=
permit=0.0.0.0/0.0.0.0
nat=no
mailbox=995@default
host=dynamic
dtmfmode=info
dial=SIP/995
context=access
canreinvite=no
callgroup=
callerid=device <995>
allow=g729
allow=ulaw
allow=alaw
accountcode=
call-limit=50

[996]
deny=0.0.0.0/0.0.0.0
disallow=all
type=friend
secret=z
qualify=yes
port=5060
pickupgroup=
permit=0.0.0.0/0.0.0.0
nat=no
mailbox=996@default
host=dynamic
dtmfmode=info
dial=SIP/996
context=from-internal
canreinvite=no
callgroup=
callerid=device <996>
allow=gsm
accountcode=
call-limit=50

[999]
deny=0.0.0.0/0.0.0.0
disallow=all
type=friend
secret=z
qualify=yes
port=5060
pickupgroup=
permit=0.0.0.0/0.0.0.0
nat=no
mailbox=999@device
host=dynamic
dtmfmode=info
dial=SIP/999
context=access
canreinvite=no
callgroup=
callerid=device <999>
allow=alaw
accountcode=
call-limit=50

[aktuna-istanbul]
disallow=all
username=200004908502121277
type=peer
secret=xxx
insecure=port,invite
context=from-trunk
host=193.243.202.97
dtmfmode=info
dtmf=info
canreinvite=yes
allow=g729
allow=ulaw
allow=alaw

[pstn sip]
auth=md5
canreinvite=no
context=from-pstn
dtmfmode=rfc2833
host=192.168.254.5
insecure=invite
nat=no
port=5061
secret=z
type=peer
username=PSTN

[pstn-incoming]
canreinvite=yes
context=from-pstn
host=192.168.254.5
insecure=invite
nat=no
port=5061
type=user
username=PSTN

# cat /etc/asterisk/sip_registrations_custom.conf 
minexpiry=150
defaultexpiry=300
maxexpiry=600

# cat /etc/asterisk/sip_registrations.conf        

register=200004908502121277:xxx@193.243.202.97/200004908502121277