Below is an example 'SIP peer' definition for Asterisk allowing you to route calls to 2talk via your SIP trunk:
The main setting you need to configure is the 'host' or 'proxy' address of your outbound trunk. You should set this to:
peering.2talk.co.nz
NOTE: This is different to the default host name you use to connect to 2talk (which is 2talk.co.nz). The rest of your settings will likely be specific to your PBX platform. If you must enter an IP address as your host then perform a lookup of the above host name to find out it's IP address (In windows you can open a command windows (Star -> Run -> 'cmd') and type 'nslookup peering.2talk.co.nz'. The IP address may differ depending on your geographical location etc.
One of the most common IP-PBX platforms in use today is Asterisk (or one of it's variants such as TrixBox). Below is an example 'SIP peer' definition for Asterisk allowing you to route calls to 2talk via your SIP trunk:
[2talk]
type=friend
context=from-trunk
host=peering.2talk.co.nz
dtmfmode=rfc2833
insecure=very
nat=never
qualify=no
canreinvite=no
disallow=all
allow=gsm
allow=alaw
This is only an example. You will need to decide which codecs and context you wish to use in your own setup. You may then route calls out through your 2talk trunk by adding a directive such as the one below into your extensions configuration:
exten => _X.,1,Dial(SIP/2talk/${EXTEN},,T)
peering.2talk.co.nz
NOTE: This is different to the default host name you use to connect to 2talk (which is 2talk.co.nz). The rest of your settings will likely be specific to your PBX platform. If you must enter an IP address as your host then perform a lookup of the above host name to find out it's IP address (In windows you can open a command windows (Star -> Run -> 'cmd') and type 'nslookup peering.2talk.co.nz'. The IP address may differ depending on your geographical location etc.
One of the most common IP-PBX platforms in use today is Asterisk (or one of it's variants such as TrixBox). Below is an example 'SIP peer' definition for Asterisk allowing you to route calls to 2talk via your SIP trunk:
[2talk]
type=friend
context=from-trunk
host=peering.2talk.co.nz
dtmfmode=rfc2833
insecure=very
nat=never
qualify=no
canreinvite=no
disallow=all
allow=gsm
allow=alaw
This is only an example. You will need to decide which codecs and context you wish to use in your own setup. You may then route calls out through your 2talk trunk by adding a directive such as the one below into your extensions configuration:
exten => _X.,1,Dial(SIP/2talk/${EXTEN},,T)