Sunday, July 03, 2005

 

Linksys PAP2-NA - Day 2

After inspecting the firewall logs I could see incoming traffic bouncing off the firewall that was from sipgate IP addresses and on SIP ports, so I put the Linksys into the DMZ on the router exposing it to the ills of the internet.

Calling it from the landline I got ringing tone but the DECT phone on the Linksys didn't ring, and there was no flashing LEDs etc to indicate activity. On inspecting the Info page I could however see that it was ringing. Inserting an ADSL microfilter in between the DECT phone and the PAP2 brought it to life - clearly the output needs an RJ11-BT adaptor that includes a ringing capacitor (provided in my case by the microfilter). So it is actually working with Sipgate on Line 1 while in the DMZ !!

Without the ring capacitor you can answer the call if you know its ringing, but that is only evident from the web interface :-(

Reading up about the Linksys PAP2-NA I discover that it is allegedly the same unit as a SIPURA SPA2000

The forums at http://voxilla.com/PNphpBB2.html were very helpful, and include configuration tools for common ATAs and poplar VoIP service providers, very useful !

I turned off the DMZ on the router and it still works. Not sure it will survive its hourly re-registration though. I have port forwarding of 5060 and 5061 to the Linksys but no other router tweaks. May need to test its resilience.

Inspired by getting the first line to work I set about connecting the second up to beta VoIP service from 18866.co.uk - this sat saying "can't connect to registration server" and declined to do anything - no blue LED :-(

After messing around with Line 1 for a while I looked at the settings in Line 2 and saw that they were corrupted - what should have been "sip.call18866.co.uk" in the Proxy field had morphed into something like "985@-16,1428@-16,1777" which is a tone definition or dial plan config - so the moral here is to check what is actually saved when you hit "Save" by revisiting the same page.

Corrected the settings and Line 2 leapt into life, made a call out and all worked well. I tested it by ringing my answering machine, to ensure 2-way audio was present.

Cool - 4 blue LEDs all shining nicely at me. Lets see if it lasts. Still not in the DMZ.

I found a load of NAT and STUN settings at the bottom of the SIP page when viewed in Advanced mode as Admin. Currently they are blank however the voxilla config wizard gives the settings for sipgate. I'm never quite happy with these 2-line devices having one set of parameters for something, nagging doubts that it'll work for both providers. For reference I have Firmware Version: 2.0.12(LS)

Off to buy some cake to celebrate.

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