k.parrent,
Well I guess I should clear a few things up first and then you can make a better decision on how you need to proceed.
When you say that you have a PS-1 attached to your system then most likely I would also assume that you have a PEC-1 as well for the extra DSP resources; this is not mandatory although a good practice.
So, with a PS-1 added to the system nothing really changes on how remote (teleworker) phones work unless they are SIP phones. The big difference is if you have a PEC-1 added to the system as well and that is where there are differences on how the firewall will need to be setup to support that configuration.
All call data is handled by the PM (Base Processor Module), but the audio can be on either the PM or the PEC-1 and they use different UDP ports for each one. Typically this happens at the same time, viola one-way audio. This means that you would need [2] Public IP Addresses with one pointing to the PM and one pointing to the PEC-1. I am going to assume that you know all the ports needed, but the difference is really between the RTP Audio like this.
RTP Audio towards the PM: 6004-6261 [UDP]
RTP Audio towards the PEC-1: 6604-7039 [UDP]
Now, if you have SIP then that has to go to the PS-1 if there is one equipped otherwise it goes to the PM, but you also have to put the Public IP in System > IP Settings > System NAT IP Address or you will have no audio after the call is made; clear as mud.
May I also suggest that if you have QoS setup on your network that you change your Audio RTP Type of Service and Data Type of Service from 0 [IP Precedence] to 184 [Expedited Forwarding] for the DSCP when going through Differentiated Services.
Hopefully that clears things up a bit for you.
Thanks,
TE