# Geek # # Voip # # Linux #

voipstunt with elastix 2.4 asterisk for outgoing calls

elastix trunks

Update on migration (2021): While I have used Elastix for years, it has also been a few years since I stopped using their software - actually... since they started to charge based on number of extensions.
I am a geek, I test stuff, ... leading to many sip devices... went back to something that is free like in free beer - but another platform switch planned... more on that later! - End of update.

Connecting Elastix to a voip provider like voipstunt is not complicated, but it is important to put all the right parameters, particularly, the registration string.

First, lets create a SIP trunk, define a name, an outbound caller ID and a max number of channels
Then fill the Trunk Name and peer details as in:

username=username
type=friend
secret=password
qualify=1000
nat=yes
insecure=port,invite
host=sip.voipstunt.com
fromdomain=sip.voipstunt.com
dtmfmode=inband
disallow=all
canreinvite=no
allow=ulaw&alaw&gsm&g726

Then proceed to set the Register String as in:

username:password@sip.voipstunt.com/username

Submit changes and Apply
Finally, add your trunk in an outbound route


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