Hello Good People,
I am having problems with Elasticx so I am trying to migrate into asterisk complied from source code.
Thus far, I have been able to create some extensions and establish local calls. Now, I want to route my calls outside. I am completely puzzled from where to start off..
I have two SIP providers:
From Provider A:
Outbound Settings
Peer Details:
type=peer
qualify=yes
port=5060
host=116.68.210.20
disallow=all
allow=g729&ulaw&alaw&gsm
dtmfmode=info
relaxdtmf=yes
Inbound Settings
User Details:
type=peer
qualify=yes
port=5060
host=116.68.210.20
disallow=all
allow=g729&ulaw&alaw&gsm
dtmfmode=info
relaxdtmf=yes
insecure=invite,port
From Provider B:
Peer Details
type=friend
qualify=yes
secret=password
host=10.11.30.7 --> IP address of a VoIP Gateway Device.
context=from-trunk
port=5060
dtmfmode=rfc2833
canreinvite=no
disallow=all
allow=ulaw
I am not getting any idea where to put these configuration that my SIP providers have provided me.
Please help.
Thank you.
Regards,
Ganesh