Most settings are in sip.conf and extensions.conf
; For each ORBTALK account - IN sip.conf (or equivalent) add registration details
register=ORBTALKACCOUNTID:ORBTALKACCOUNTPASSWORD@talk.orbtalk.co.uk/ORBTALKTELEPHONENUMBER
; In sip.conf
; For each extension in you'll need something like this - take care to get 'context=' setting correct -
; (this context is 'from-internal' for TRIXBOX - it could be 'default' for other Asterisk versions)
; LOCAL Extension - on same private subnet as PBX
[100]
type=friend
secret=100
record_out=Adhoc
record_in=Adhoc
qualify=yes
port=5060
pickupgroup=
nat=no
mailbox=100@default
host=dynamic
dtmfmode=rfc2833
disallow=
dial=SIP/100
context=from-internal
canreinvite=no
callgroup=
callerid=device <100>
allow=
accountcode=1234
; 101 is an example for Remote Extension -
[default]
type=friend
secret=ORBTALKACCOUNTNUMBER
record_out=Adhoc
record_in=Adhoc
qualify=yes
port=5060
pickupgroup=
nat=yes
host=dynamic
dtmfmode=rfc2833
disallow=
dial=SIP/ORBTALKACCOUNTNUMBER
context=from-internal
canreinvite=no
callgroup=
callerid=device <ORBTALKACCOUNTNUMBER>
allow=
accountcode=
; Is your PBX behind a firewall - but got DynDns or Domain Name on you Wan IP Address?
; Add following in to you sip.conf (or equivalent)
; Note for your router subnet - you may need to change 192.168.1.0 to something like 192.168.0.0 or 192.168.2.0
externhost=YOURDOMAIN.gotdns.com
localnet=192.168.1.0/255.255.255.0
nat=yes
Problem with Remote extension registration or no voice packets being heard?
Don't forget to set Nat = yes for these extensions
There could be firewall issues - effecting your account - (eg. blocking of normal reg sip port by some draytek routers)
You may need to change bindport for something like 5070 and back sure you root these packets to your PBX
Make sure all UDP voice packets get to your PBX (eg. forward RTP (UDP) packets from ports defined in rtp.conf (default 10000 to 20000 )
On the Draytek 2910 router - we forward the following to the PBX which sits behind the router.
UDP 5060-5090
UDP 4569 (Only if you use IAX)
UDP 5038 (Only if you use * Manager API)
UDP 10000-20000 (matches rtp.conf - RTP voice traffic)
Be aware of the order your router handles incoming traffic - it will probably go through all the firewall and redirect rules 1st (find from router dealer which order these two are triggered) before landing onto to your 'DMZ host'
Use Asterisk Command Line to Debug SIP traffic- ('sip debug'/'sip no debug'/'sip show peers'/'sip show registry')
You'll need to set incoming rules on your extensions.conf for incoming calls!
register=user:password@IPaddress/user
[OrbTalk]
host=IPaddress
secret=password
type=peer
username=user
