Asterisk (Free PBX 10 or above) Configuration Settings as a registered sip extension
This is the basic method that 99% of asterisk users connect. There are 2 other methods that can be conected. Some people tends to mix tthem and they all call them as sip trunks, but in reality they are all different FreePBX 10 setttings FPBX-2.10.0(1.8.13.0) but should work with earlier versions
username=XXXX*XXXXX type=peer secret=PASSWORD qualify=yes pedantic=no insecure=very host=VOIP_IP_SERVER (received in the activation email) fromuser=XXXX*XXXXX fromdomain=VOIP_IP_SERVER (received in the activation email) disallow=all allow=alaw
XXXX*XXXXX
username=XXXX*XXXXX type=user secret=PASSWORD qualify=no insecure=very fromuser=XXXX*XXXXX context=from-trunk canreinvite=no
XXXX*XXXXX:PASSWORD@VOIP_IP_SERVER/XXXX*XXXXX
With this on you need to have the phones also use G711a codec. To use G729a from the phone you would need to have a licence from diguim installed in asterisk and in the cofig add [allow=g729] without brackets. It is not recommended to use G729a free licence in Asterisk as it seems that the quality is not the same as the paid version. Please Note this info is supplied as is and we are not liable for any loss or hacking to your Asterisk. If you do not know how to secure properly an Asterisk Installation please contact a Proffessional to do it or contact us and we can reffer you to a Proffessional Installer. |