Digium Asterisk SIP Setup Guide

STEP 1

Complete the following guide: Creating a SIP Peer for PBX

STEP 2

Sample "sip.conf" file:

[general]
relaxdtmf=true
rfc2833compensate=yes ;Recommended for Asterisk 1.4

;Only one peer is required per PBX
register => USERNAME:PASSWORD@PROXY ;Proxy may vary, please refer to the selection made in Peers / Trunks.

[voicemeup-trunk]
type=friend
context=inbound ;Change this to your own context if required

host=PROXY ;Proxy may vary, please refer to the selection made in Peers / Trunks.
fromuser=USERNAME
username=USERNAME
secret=PASSWORD

nat=yes
canreinvite=no ;Change to 'yes' if you have no audio
disallow=all
allow=ulaw
sendrpid=yes
trustrpid=yes
insecure=very,port,invite ;This is important or call will be refused

STEP 3

Sample "extensions.conf" file:

[default]
;This is for your outbound calls
exten => _X.,1,Dial(SIP/${EXTEN}@voicemeup-trunk)
exten => _X.,n,Hangup

[inbound]
;This if for your inbound calls
exten => PHONE_NUMBER_A,1,Dial(SIP/LOCAL_PHONE_A) ; 
exten => PHONE_NUMBER_A,n,Hangup

exten => PHONE_NUMBER_B,1,Dial(SIP/LOCAL_PHONE_B) ; 
exten => PHONE_NUMBER_B,n,Hangup

STEP 4

Enjoy !