SIP Gateway Setup

I am using SIP as VOIP technology. I purchased Linksys PAP2 SIP gateway boxes. They allow me to use any ‘normal’ telephone for VOIP.

Originally posted on Oct 17 2008

I was in VirginMedia ADSL broadband. Boxes have arrived from a trusty e-Bay source. The default setup of them is quite ok, you just have to add your SIP ID, and similar, for identification so that PAP2 succeeds when it wants to login to PAP2 has a web based interface and everything went smoothly as ever with similar devices, like routers etc. Connected both PAP2 boxes, tested them and all was dandy. I had two SIP telephones working OK.

I took one back home (Belgrade) and connected it and it worked ok. All in all as expected: no problems whatsoever. After all SIP is by now mature and a lot of people and companies are using it.

Then I moved my London address. Which was very good, among other things, because I had the opportunity to connect to the best broadband available in UK: namely VirginMedia optical cable. The only little problem  was that SIP box could not connect. Line was dead.

My networking experts friends told me that I will ‘never succeed’ to connect to SIP while on Virgin cable. Of course, a little bit of conspiracy and unfair competition practices have been added to each and every ‘testimony’.

Then, I decided to look into this issue.

What immediately seemed odd to me was that PAP2 could not (or refused to?) establish its dynamic IP, as before. And, I still do not understand why it refuses to do so? It is connected to another  router. Another but very similar. Before it was DG834G – 54Mbps wireless-G modem router , and now my Linksys PAP2 box is connected to WPN824 – RangeMax™ Wireless Smart MIMO Router (no modem included, ideal for cable subscribers).

So my PAP2 could not (want not) establish an dialogue with new routers DHCP server and stayed stubbornly on IP … Through somewhat arcane PAP2 telephone interface I managed to give it an static IP and gateway. In case you might think this is straightforward here is the part of the manual which describes how this is done :

A. Obtain your IP Address, Default Gateway and Subnet Mask. If you are using a router, contact the manufacturer for the retrieval of this information. If you are not using a router, obtain this information directly from your ISP.
B. Pick up the telephone receiver of the phone connected to the adaptor.
C. Dial ****. You will hear a voice prompt saying “Linksys configuration menu.”
D. Press the 101# on your analog phone to access the static IP configuration menu.
E. Press 0 # to disable DHCP. Press 1 to save the setting.
F. Press 11# to access the IP address Menu.
G. Enter your IP address followed by #. Use * to indicate a period (.). For example, the IP address would be entered as 123*45*67*89#. Press 1 to save the setting.
H. Press 131# to access the default gateway menu.
I. Enter your default gateway followed by #. Use * to indicate a period (.). Press 1 to save the setting.
J. Press 121# to access the subnet mask menu.
K. Enter your subnet mask followed by #. Use * to indicate a period (.). Press 1 to save the setting.

Ok, never mind, it is doable. Now I was able to see and control my PAP2 box from the PC attached to the same router. But still I had no connection to the SIP. Phone line was dead. I sent an e-mail to Sipgate support. Engineer asked me if I did set up a port forwarding on my router. Which I did not because on my previous Netgear router this was not necessary.

I opened the router web site and setup the port forwarding like so:

router sip port forwarding

Where is PAP2 static IP, which had to be made as reserved address on this same routers DHCP control page. To add this service one has to clik on the "Add custom Service" button above. Port 5060 is so called "SIP port" and can be found on the PAP2 web interface pages, under "Line 1" settings. IMPORTANT: telephone is connected on PAP2 to the "Line 1" port. To make this even more clear here is the DHCP setup page from this router :

DHCP setup on the router

After this, I still had no working SIP phone line. PAP2 Line 1 status page , still had "can not connect to the Login server", clearly visible. Then I remembered I had not set up manually DNS server entries on the PAP2 , so I opened again PAP2 web interface and set up that also. Now this PAP2 box has its "System" (aka LAN) setup like this :


Both gateway and DNS servers are pointing to which is the router.

And finally … my SIP phone line started working.

Update 2009 Sep 21

Since then I have noticed that this PAP2 device went on and off line regularly and unexpectedly. At unpredictable times.
And when sees my device as “off line” nobody can not call me from non-sip numbers. Which means nobody using her mobile or land line, can reach me on this sip number. The call will be “rejected”.
Times is the emphasized word here. After lot of trials and tribulations, and as ever it turns out that I have not seen the woods because of the proverbial tree in front of my face. The time-out is the issue. More precisely the registration time-out.
It you observe the screen dumps above carefully you might notice this:

Last Registration At: 2/14/2003 18:38:30 Next Registration In: 2851 s

And this is definitely not OK. The connected PAP2 device has to register each 10 minutes. This is how registration server is set up. If any user installed and owned device, is not re-registering after 10 minutes, it is labelled as “off line”, and thus calls are not directed to it, but are forwarded. And if call forwarding is not set-up call is rejected. As simple as that.
The obvious question for the, is why calls to the off-line device are not voice recorded but rejected? It turns out that only on-line device calls are voice recorded?

So, the option “Register Expires” in my (and yours) PAP2 configuration should be set to 600 seconds. And after doing so, my PAP2 device and the SIP number it represents, are constantly “on line” as long as PAP2 is connected and switched on. And now people can call me on this number normally as on any other number. Which is one of the purposes of having SIP number after all.