Successfully set up service Using Billion 7402VGP with BT Duet 20 as Phone 1 and Panasonic KX-TCD965EC Dect with 2 handsets.

The essential step to ensure success is to log into the router, go to Configuration/VoIP/Wizard/VoIP User-defined Profiles/Create - here you enter the information supplied in your email to create the User-defined profile that completes the other fields. Then apply changes, save configuration and then restart.

Phone 1 (Codec G729)

PSTN service works completely normally.
Outgoing calls routed via broadband are normal.
Incoming calls to geographic number are normal.
SIP calls to 123 and 160 are normal.
Other SIP services not tested.

Phone 2 (Codec G729)

Similar results to Phone 1 but the phone does not ring when calls are received although caller id works normally.
Sound quality is not the best.

I've successfully set up X-Lite 3 to be able to dial out to 123 and I've successfully made an outgoing call to a landline but can't currently receive incoming calls on my PSTN number (caller eventually goes to answerphone but X-Lite doesn't ring). This may be the office firewall or even the fact that it's X-Lite 3 I'm using - will experiment with this later. I'll leave this until tomorrow to see if there's any change as I've only just registered. and I've successfully received an incoming call to my PSTN number with thanks to Pendragon for this post --> http://community.plus.net...ex.php/topic,58422.0.html

I've managed to register the SIP account on my N91 which is a big step forward but I need to change the firmware to get the internet call option. I'm currently with o2 and though I can enter SIP settings and register there doesn't seem to be an internet call option at present.

A little unexpected progress with Fring on the N91 as well: I can make a 123 call. Can't call PSTN numbers (the call hangs up almost immediately) and I don't have a test SIP number to call (is there a test number to call other than 123?)

Fring settings that vaguely worked were:

Username: 1234567
Password: *******
Proxy server: lilac.gradwell.net
Note sip2.plus.net didn't work and nat.plus.net (with or without :5082 ) didn't work, the IP addresses didn't work but lilac.gradwell.net did.
I've yet to try lon-ppc-3.gradwell.net (= nat.plus.net )

Good old ping -a 193.111.200.56 eh?

I am using a Zoom VOIP adapter patched into my router (Netgear DG834).  The settings were sent to you via the Plusnet My Questions portal (QID:  23509140) but if I need to send that file again I will happily do so.  As a result of that Question, I succeeded in getting outgoing calls working.

So far, I have only tried calling in using my landline phone.  I do hope you are not going to tell me that can't be done as it shares the same line!  Certainly, I can call the landline from the VOIP phone.

Copied from another thread - highly useful for PAP2 users....

Dialplan on the PAP2 - the supplied Linksys dialplan will not allow you to dial "**"

A suitable dialplan is listed below - this is the dialplan that a Gradwell engineer uses on his own SPA941 at home ;

( *x. | **x. | 0xxx xxx xxx. | 186 xxx xxxx | [1-8]xx | [1-8]xxx | [1-9]xx xxxx | 00 xxxxx x. | 141 0 x. | 1470 0 x. )

Explanation:
*x. Any star code
**x. Any double star code
0xxx xxx xxx. Any UK PSTN number
186 xxx xxxx 186 VM access (takes you to voicemail for given extension)
[1-8]xx 3 digit short code
[1-8]xxx 4 digit short code
[1-9]xx xxxx 7 digit extension
00 xxxxx x. International number
141 0 x. Withhold CLI on PSTN call
1470 0 x. Present CLI on PSTN call

Just for reference, Gradwell have configuration and setup guides for a number of pieces of kit at the link below:

GradWell eSupport

Try and make this page your first port of call.

Cheers
Rich

[Moderators Note by Daniel (Assos), shortened long link as it was streching the page.

I would like to report success with the Billon BiPAC 7404VGO 5.52g VoIP VPN router.

It configured as you would expect, with one major exception:

Where is asks for your Phone Number, and you think you would enter the geographic number here, you actually need to enter your username again (which is your extension number on Plus Trial)!

(In fact this really caught me out using the other phone port to register my voipuser.org account, as it needed my alphabetic username entering into this "phone number" entry. But again with those settings it works a treat.)

With my geographic number entered it was reporting as "registered" at my end, but not showing up as registered at Gradwell. Any attempted calls were greeted with beeps. Changing this to my username, solved all the problems, and calling is working correctly inbound & outbound.

I successfully have my DECT handsets & internal fixed phones all on port 1 of the router (PlusTalk VoiP Trial), so they all ring with a different ring tone depending on whether it is an incoming call via BT or VoIP. I can ring out using either two using the same phone.

All other settings are taken from the signup emails.
I needed no stun or proxy settings. No firewall or port forwarding settings. It just works now.

If anyone needs help with a Billion router then you can PM me here.

Mike

I moved to SJPhone (version 1.60) from X-lite (both bersion 2 and 3 tested).

X-Lite Version 2 results

Connection worked fine but had audio problems (stuttering).

X-Lite Version 3 results

More hungry on system resources and again the same audio problem.

SJPhone 1.6 result

Worked first time and audio OK.

All three worked with my USB phone, Yealink P1K, but SJPhone works best.

I have tested other softphones but they do not function as well as SJPhone.

Hope this is of some help.

Ray

In the absence of a full SJPhones support page (coming very soon!) I thought i'd put some of the settings here for visibility....

SJPhones can be found at this link:
http://www.sjlabs.com/sjp.html

-you can choose to download the Mac or PC version here as well (the top list of links)
-there are some links to PDA versions but i would suggest sticking to the PC versions for now (we havent tested the PDA versions yet)

-main settings required:

N.B: You get prompted for your SIP ID and password at a later stage to this main bulk of the settings

Under SIP proxy tab:
-domain/realm: sip2.plus.net
-check box for 'use outbound proxy'
    -Proxy (URI): nat.plus.net:5082
    -Proxy usage mode: SMART
-check box for use separate outbound proxy for NAT
    -NAT proxy (URI): nat.plus.net:5082
    -NAT Proxy mode: SMART

STUN tab:
-server address field: <remove the default stun settings>

Everything else should be as default - the default settings are pretty good on this which makes it so cool.

Cheers
Rich

Where some of you may be experiencing "unresolved destination" errors on outbound calls - please ensure that you have not defined the SIP realm as sip2.plus.net:5060 - you do not need the ':' unless your device/router is not SIP aware. Gradwell have confirmed that you should not need to define this port as its a default port anyway.

Give it a go - it worked for me!

It has to be said, It (voip trial) won't get your average joe rushing to get this going. Too much of a pain in the BTM 
Just tried setting up X Lite (already had the old system running but no joy with voip trial.)
SJ phone was a none starter for me too.

FYI Apple Mac OS X is the OS being used and modem router Netgear DG834n voip is far from intuitive to set up. Think I'll stick with my iphone it's easier.
 
Any Mac setup pages around?

We're on the process of creating a SJPhone Mac setup guide which should be on our portal pages early next week with screenshots included. In the meantime, try these settings below:

http://www.sjlabs.com/sjp.html

-you can choose to download the Mac or PC version here as well (the top list of links)

You get prompted for your SIP ID and password at a later stage to this main bulk of the settings

Under SIP proxy tab:
-domain/realm: sip2.plus.net
-check box for 'use outbound proxy'
    -Proxy (URI): nat.plus.net:5082
    -Proxy usage mode: SMART
-uncheck box for use separate outbound proxy for NAT

STUN tab:
-server address field: <remove the default stun settings>

This is an updated version to the ones posted previously - you do not have to define the SIP domain/realm port or use the outbound proxy for nat.

cheers
Rich

Thanks for the reply and great on the setup info. However, all those parameters were on the initial email when I registered with voip trial but couldn't fathom out how to get it into SJphone unless the version I got was faulty.
Will try downloading SJphone again. Don't hold your breath

For windows use this one:
http://www.sjphone.org/so...one/SJphone-1.65.2637.exe

I am having a little trouble setting this up. I have a Speedtouch 780 and configured it as follows:


Registrar sip2.plus.net
Registrar port 5060
Proxy nat.plus.net
Proxy port 5060
Expire time 3600

The router does not show a light on Voice. If I lift the handset I hear a fast engaged tone.

As suggested here for other Speedtouch routers, I changed the Proxy port to 5082.

That gave the same result except the tone was a very slow engaged tone.

I have received an e-mail telling me I am registered but the VoIP beta trial account page shows me as not registered. Is there something else I have to do, please?

hneil,

I think you need to set your proxy port to 5082

OK, I've gone back to 5082. The phone now gives a normal engaged tone so something is still not right.