Hi,
I have a scenario where on 1st asterisk i have PRI line (with no internet connectivity) from which i want to transfer call to 2nd asterisk server(In same LAN of 1st asterisk but with internet connectivity) and then from this 2nd server i want call to be transferred to 3rd asterisk server which is on AWS.
Can i perform this? If yes please suggest the way forward…
Treat the other Asterisk like a service provider, except that you can use static addresses and both way authentication.
You can also treat the first one like a local phone, with a dynamic address, in the second.
Hi David,
scenario here I am trying is, to connect PRI line to webrtc client on AWS server.
I managed to receive the call on extension logged in on chrome with sipml5 demo application.
But, after successful calls establishment there is not audio in the call.
You are probably not using “extension” with its Asterisk meaning and there is no information (dialplan, logs, etc.) based on which a diagnosis could be made.
As a general principle, get things working with SIP phones, before attempting WebRTC.
I seem to remember that there was recently posted information on the complexity of the NAT environment for AWS. I would suggest getting the system working on real hardware before trying to run it in the cloud.
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