The quickest way to set-up your PAP2T is to use the wizard found here:

https://www.voipfone.co.uk/IP_Phone_Setup.php

If that doesn't work for you or you wish to do it manually, please follow the following steps:

1. Find the IP of your device:

Pickup an attached telephone and dial **** 110#, make a note of the numbers which are read out.

2. Visit your PAP2T’s configurations page:

Now that you know the IP of your device, type it into a browser on a PC attached to the network. For example Internet Explorer, Mozilla Firefox.
You may need to type in http:// before the IP.

3. Login as an administrator, and view configuration in advanced view

Click the 'Admin Login' button at top right of page, then click 'Switch to advanced view' in the centre of page.

4. Setup the line you wish to use

Click Line 1 or Line 2 depending on which line you wish to configure for the Voipfone service. Once you have the page up that you need to configure you need to enter the following information

A: Proxy and Registration;

Proxy: sip.voipfone.co.uk

Use outbound proxy: No

Outbound Proxy: nat.voipfone.co.uk:5065

Use OB Proxy in Dialog: Yes

Register: Yes

Make Call without Reg: No

Register Expires: 60

Everything else can be left on its defaults in the Proxy and Registration section

B: Subscriber Information

Display Name: Name you wish to have displayed when making calls to other VoIP users

User ID: Your user ID starting 30 and consisting of 8 digits. E.g. 30100000 [for PBX users enter user ID with *200 at the end, where your extension is 200. E.g. 30100000*200]

Password: Your 6 digit Voipfone password. This can be found in your sign up confirmation email. If you're setting up an extension - it's the password for your extension. You can find it here https://www.voipfone.co.uk/extensions.php#ext

Use Auth ID: Yes

Auth ID: Same as User ID (see above)

Everything else can be left on its defaults in the Subscriber Information section.

Once this information is entered submit changes by clicking save settings or submit changes at the bottom of the page. Your device should now be working, if not try enabling the outbound proxy. See 4.A above for information on how to do this.

5. Other Settings
To get the best performance out of your device you can edit some of the default settings.

A. Registration Retry Interval

This setting determines how long your device should wait before trying to register again if you lose registration for any reason. To change this value:

Choose SIP tab, under SIP Timer Values (sec) change Reg Retry Long Intvl to 120

B. Regional Settings

The Linksys PAP2T can be set to give regional sounds; by default it has American sounds. The settings can be changed in the Regional Tab. UK settings are as follows:

Call Progress Tones

Dial Tone: 350@-19,440@-19;30(*/0/1+2)
Second Dial Tone 420@-19,520@-19;30(*/0/1+2)
Outside Dial Tone: 420@-16;10(*/0/1)
Prompt Tone: 520@-19,620@-19;10(*/0/1+2)
Busy Tone: 400@-20;10(.375/.375/1)
Reorder Tone: 400@-19;20(*/0/1)
Ring Back Tone: 400@-20,450@-20;*(.4/.2/1+2,.4/2/1+2)
Confirm Tone: 600@-16;1(.25/.25/1)
MWI Dial Tone: 350@-19,440@-19;8(.1/.1/1+2);10(*/0/1+2)
Cfwd Dial Tone: 350@-19,440@-22;10(.75/.75/1+2)
DND Dial Tone: 350@-19,440@-22;10(.80/.75/1+2)
Holding Tone: 600@-19;*(.1/.1/1,.1/.1/1,.1/2.2/1)
Conference Tone: 350@-19;20(.1/.1/1,.1/9.7/1)
Secure Call Indication Tone: 397@-19,507@-19;15(0/2/0,.2/.1/1,.1/2.1/2)
Feature Invocation Tone: 350@-16;*(.1/.1/1)
SIT1 Tone: 985@-16,1428@-16,1777@-16;20
(.380/0/1,.380/0/2,.380/0/3,0/4/0)
SIT2 Tone: 914@-16,1371@-16,1777@-16;20
(.274/0/1,.274/0/2,.380/0/3,0/4/0)
SIT3 Tone: 914@-16,1371@-16,1777@-16;20
(.380/0/1,.380/0/2,.380/0/3,0/4/0)
SIT4 Tone: 985@-16,1371@-16,1777@-16;20
(.380/0/1,.274/0/2,.380/0/3,0/4/0)
Off Hook Warning Tone: 480@-10,620@-16,1400@0,2060@0,2450@0,2600@0;60
(.2/0/1,.2/0/2);240(.1/.1/3+4+5+6)


Distinctive Ring Patterns

Ring Cadence 1: 60(.4/.2,.4/2)

Miscellaneous

Caller ID Method: ETSI FSK With PR (UK)

The wizard seems to set some of the PAP2T's settings incorrectly.

In the proxy and registration settings it sets Use outbound proxy to Yes.

In the Subscriber information section it sets the User ID to ="30123456" instead of just 30123456 (that's obviously not my real user id). The equals sign and the quotation marks need to be removed. In the same section it sets the Auth ID correctly.[/i]

On incoming calls my phone does not ring. Outgoing calls are fine. Incoming diverted to mobile is fine. Incoming to voicemail is fine. But incoming to PAP2T adaptor and deskphone does not ring. If I pick up receiver I can still hear dialtone. Caller hears ringing tone but I don't know what or where it is ringing. Can anyone help?

I am looking to use these adaptors in my workshop.

I will want to put 3-4 extensions in. I did have a "clamshell" phone from you guys but it was rubbish! (hence why you don't sell them anymore I think!!)

I am now sure wireless is that great for VoIP? Maybe the clamshell just put me off!

I wanted to get some DECT phones which them plug into the Linksys PAP2T VoIP Telephone Adaptor. What do you think?

Or should I use a wireless phone device?

The trouble with the PAP2T is that you can only connect two handsets, cables all over the place. I think wireless works fine for VOIP except the batteries on the wireless phones run down very quickly and the phones aren't that cheap.

You could try something like a Siemens Gigaset S450IP

http://gigaset.com/shc/0,1935,hq_en_0_1 ... rN,00.html

You can connect upto 6 Dect handsets.

PS the handsets don't necessarily have to be the Siemens handsets

Thanks for that!

The thing is I will only need one 4 handsets, each one on a seperate extension. The most important thing is the DECT phone range really.

There won't be many cables as they just will go directly to the cat 5 socket and sit on the desk. Also I do not want advanced phones - my guys will not need or use the functionality and it will make things harder! KISS is the word of the day for this project.

The most important thing is the dect range of the phone and the VoIP connection. I have fibre between the workshop and office as there is about 50M, so as long as the DECT is good, the quality should be fine.

Oh, I also need the phones to be able to be LOUD. Some of these phones are made for use in a soudproof room I think htesedays and do not suit a workshop enviroment!

If the two buildings share the same consumer unit you could try one of the mains network products like Devolo.

http://www.devolo.com/co_EN/produkte/dl ... lussk.html

I like it!

Sadly the workshop and the office are on different lines, so that won't work!

I believe that they can be on different circuits - I think they may even work on differing consumer units if the phase is the same but I might be making that up. Why not call them?

I did.

Sadly I am on a different phase all together! God damn it!


What is the Dect range on the SNOM M3 like? Will you be offering it shortly? If you think its a good phone I might try it!

We don't like the snom M3 yet - or we'd sell it. We're waiting for a firmware update that we're told will fix a few things.

OK. Do you know when the firmware update will arrive?

In your opinion, when the firmware update arrives, will it be a phone worth having?

Don't know when we'll get it - we've been waiting several months.

When the bugs are fixed, we'll sell it as we are generally very big fans of snom - but the M3 is quite a step away from their wired phones and seems quite expensive compared to the Siemens DECTs.

We're not that keen on it's look and feel either but that's subjective.

OK, fair assesment.

You know what the next question is don't you. Oh yes you do.

Siemens DECTS, when will they be back in stock? )

FWIW my Siemens DECT handsets will do about 50-70 meters. It's not a noisy environment (electrically), though. You can set the volume on the handset pretty easily.

Also When setting up the base station you just select Voipfone and voila. Not that you would want to buy them elsewhere.

PS for anyone else regarding power plugs. I use the Netgear HDX101 on a three phase setup. It works OK on different phases but with a 10-15% loss in throughput

This is a bit of a problem - they're not out of stock but we're not happy with their power supplies - we've had too many fail. Siemens are looking into this - as soon as we have an answer from them we'll put them back on sale.

Sounds good! One thing that is not clear, is the products strictly point to point or if you plug say 10 in does it act like a full network?

OK! It is an odd area, there seems to be far too few good quality VoIP products around for mobile use at the moment.

Are the phones going to be back on sale anytime soon!? No answer from them still?

I'd like to buy some, I need to get moving on this project. What do you reckon guys, have the failures really been that bad?

We still don't wish to sell Siemens phones right now.

As well as the power supplies failing on lots of customers we've also seen problems with their later firmware versions which cause problems with incoming calls not always working.

OK!

So no more dect/sip phones available from you then. Are you planning on introducing a new one, or are you sticking with the adaptors and standard DECT phones now?

Not quite sure what you mean. You plug the base station into an Ethernet port and you can hang six handsets off that base station. You can have multiple base stations.

Surprised about the power supplies failing and firmware updates, I haven't had any problems. Then again I'm not that heavy a user.

We've just heard that the Siemens have identified the fault to be a bad batch so we're putting them back in stock.

Good news. I bought mine in France over a year ago, I assume it's a different power unit.

How about the firmware, though - that worries me.

Siemens released new firmware yesterday which is supposed to correct this problem.

We've not yet fully tested it.