Hardware required for forwarding call via SIP

We have a main office using Avaya IP office (SIP trunk supported) installed. Now, we have a new branch in another country. We would like to forward the incoming call from analog telephone line at the new branch to our Avaya IP office in the main office.

Hardware Required:
My question is what hardware I need at the new branch in this situation for forwarding the analog call from our new branch to the main main office via SIP. For option 1, digium VoIP gateway G100. For option 2, install AsteriskNOW and configured as a VoIP gateway. In option 2, what hardware do I really need, e.g. PXS card?

Do we need fixed IP addresses on both sites? Or we just need a fixed IP address in the main office?

Hope anyone could help to answer and explain. Thanks in advance.

If you want to forward all incoming calls to HQ, you need only One analog gateway. But if you have some extensions too I think best option is IP-PBX like FreePBX with an analog card.

In both case you don’t need any fix IP for new branch. Only you need it for HQ. So SIP trunk from new branch will register to your Avaya IP office.

So what I need for IP-PBX is just a PC with Ethernet port and a 4-Port Analog Card with at least 1 PXO module. Am I correct?

Yes but you need FXO port. You can use embedded solution too.

Whether you can use dynamic addresses will depend on your VPN solution, as best security practice in this case is to use a VPN.

Asterisk can cope with a dynamic address at its side, provided that it can register with a static address at the other side. The reverse is probably also possible. However, the dynamic address must not actually change whilst Asterisk is running.