[Freeswitch-users] SIP Trunk

[Richard Mace]

Hi,
I have a trunk that currently works with Asterisk, and I am trying to get
it working with FreeSWITCH. The Asterisk config is:

[out_trunk]
disallow=all
host=sip.voip-unlimited.net
username=username
fromuser=username
secret=password
type=peer
dtmfmode=rfc2833
nat=no
context=incoming-sip
insecure=invite
allow=alaw
fromdomain=voip-unlimited.net

Any idea how I would configure the same in within FreeSWITCH please, as my
current attempt doesn't seem to be working?

Thanks very much in advance

Richard
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