The following configuration has not been tested for PSTN failover. The FXS port is registering to a sipXcom 14.10 PBX and handles inbound/outbound fax transmissions as well as voice.
The first page as seen below shows the firmware and hardware version that this unit was running.
Click on Admin Login and Advanced and then click on the SIP tab under the Voice settings. The following page will be displayed:
Set RTP Packet Size to 0.020 and then click on the Submit All Changes button at the bottom of the page.
Next, click on the Line 1 tab. Set the Line Enable to yes, set any QOS settings you may require, set the SIP Transport to UDP, set the Proxy to your SIP domain (for SRV sipXcom setup), set Reigster to Yes, set Use DNS SRV to Yes, set DNS SRV Auto Prefix to Yes.
Set the Display Name to what you want local phone users to see for a user name, set User ID to the user's PBX extension, set the Password to the SIP password (not the PIN!), set Use Auth ID to no, set Auth ID to the user's PBX extension.
Set the Preferred Codec to G711u, set Use Pref Codec Only to yes.
The following dial plan will dial just about any valid number (depending on you Dial Plan in the PBX) . Just before the end of the page set the Dial Plan to:(*xx|11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.)
Click on 'Submit All Changes' when done with this page.
All of these settings can be seen in the following three screen shots:
If your DNS is configured properly the gateway should register and be able to make calls at this point.
We need more detail here...
PAP2 Settings to adjust.
Under the sip tab:
RTP Packet Size: 0.020
Under the line settings:
DTMF Process INFO: no
I think that is all I changed from the defaults, and these settings work on both of my PAP2's. For some reason this does not work consistent with all the Linksys/Sipura models, for example I ended up using different settings on a SPA 2100.