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:
From Provider B:
host=10.11.30.7 --> IP address of a VoIP Gateway Device.
I am not getting any idea where to put these configuration that my SIP providers have provided me.