My SIP Switch

Zoiper is nice little softphone that works well on MAC OS (note that they also have Windows and Linux versions). It has been tested with My SIP Switch and is working more than fine.

You can download it for free on http://www.zoiper.com/

It’s quickand easy to set it up and if you don’t have firewall or connection issues you should be able to get started quickly.

Here is a step by step configuration guide.

1) This is the default interface, click on the spanner to open the “Options” window.

2) Select “SIP accounts” and then “Add new SIP account”. You’ll be prompted to enter a name. MySipSwitch in our example, but that can also be your name or something else.

You should see the interface below, to add your My SIP Switch account details.

The domain is : sip.mysipswitch.com (you can also specify the port : sip.mysipswitch.com:5060 but this should be optional)

Then mention your MySIPSwitch username and password.

The Caller ID is your Username, so write it again.

Once you are happy with the settings you specified, press OK at the bottom and you should be ready.

If the line one shows “green” (as below) that indicates a successful registrations. Now you need to take a look at your dial plan and get calling.

If you need help regarding the dial plan, you can refer to older post on this blog :

Default Ruby Dial Plan and  Inbound Call Management in Ruby

Do not hesitate to use our forum if needs be.

Hi all,

Here are some screen shots of the configuration interface of a Grandstream Handytone 286 ATA that with indications on how to set it up with My SIP Switch service.
Note that it is very similar to Budget Phone 100 / 101 configuration screens.

Thanks a lot to Luc for the screen shots.

Note that the IP settings are specific here. This ATA is connected behind a Linksys WR54G router with NAT enabled ! You will want to change that depending on your network configuration.

The DHCP options means that the IP settings will be assigned dynamically by the router. It may be handier for less technical people.

The DNS values are specific to your broadband provider so mind these  values.

Note that the “local SIP Port” has been changed here for a specific reason. 5060 is in most cases fine. You need to make sure that the router which is in front of this ATA allows SIP traffic on this port.

The STUN server can be changed of course. There are a bunch of free ones you can use (see : http://www.voip-info.org/wiki-STUN)

Press Update and give it a go !

On the status page, you should see the SIP packets getting through and RTP ones if you are making calls.

You can also take a look at the configuration guide on Blueface’s site : HandyTone286 Configuration

Any questions , refer to our Technical Forum

Following my previous article about PhonerLite I wanted to write a quick review of his elder brother : Phoner

http://www.phoner.de/index_en.htm

It’s a Windows only softphone (95/98/ME/NT40/2000/XP)

The configuration is also made with a step by step Wizard (nearly the same one as PhonerLite), so it is really convenient and easy. I tested it with My SIP Switch and my impressions were the same as for PhonerLite : really fast to get started, easy to use and I cannot complain at all on the audio quality.

It is fully compatible with SIP and has all the main options/feature you would expect on a softphone. What makes it a bit special is that it has ISDN facilities (which is not that common on a softphone). Note that you need an ISDN adapter.

The complete list of features is available here : Phoner Features

I discovered today (Thanks to our user : deeknow) this nice, very easy to use and free softphone : PhonerLite.

Official site and download

The offical English web page is :

http://www.phonerlite.de/index_en.htm

PhonerLite is the lighter version of Phoner. It shares most of the code but with a (much) lighter interface in order to save resources. It was using 8Mo of RAM on my Windows XP while idle which is much lower than the idle XLite which was using up to 32Mo.

The donwload page is :

http://www.phonerlite.de/download_en.htm

If you download the ZIP file, you don’t need to do any installation (just don’t mess with the DLL); you can start using the phone by double clicking on PhonerLite.exe .

Screenshot and a quick review

It does all the necessary thing : easy control of settings and options (sound settings, codecs, security, network settings) … Regarding the phone features : call on hold (that’s the little hand near the green ‘pick up’ button), redial (the yin-yang like image below the hang up button), do not disturb, conference calls , call forwarding (I didn’t test these 2 features so far), logs, phone book …

Another thing I like about it is that it provides live statistics about the line such as Jitter and packet loss for instance. If you are having call quality issues, you can easily check if it is due to your network or broadband connection.

I have not tested it for long enough to judge it properly but the voice quality seemed nice; as nice as you can expect it to be on a softphone.

My SIP Switch Configuration

Now here are the steps to configure it with My SIP Switch:

Note that if you are behind a firewall, you may want to enter a STUN server (for instance : stun.xten.com).

Enter your My SIP Switch username and password. The authentication name is not necessary.

Select your audio devices.

Confirm and start My SIP Switching

Is SIP/VoIP easy ? depends…. most of the time with the common “Betamax providers” it is,

You should have not much problems, due to standard port addresses, etc… but when you do run into trouble, do you know where to look ? Should you search forever ? No, big chance your router is out of date, and your ATA isn’t the problem, and you then ask yourself, what should I set and where ? Upgrading your (modem)router is a good idea, big chance the new one has allready the “NAT things” set good at its default values, but if you have to set things for another sip-provider or a different setup, it starts all over again, so what did I recently do ? I ….bought the …. Fritz!Box 7170 !

Also because I heard a lot of good things about this one, by users and also as a “by ISP provided device” (snooping around in the webinterface, I now see the point of that) (voip over ADSL…)

Follow the paperguide, that comes with the Fritz! (7170) and look for your “situation”

The box uses a weird looking cable to connect to your splitter, (or other if appropiate) it’s a ‘Y’ cable, that puts your telephone line connection and ADSL connection to one connector, that goes into the Fritz!Box, so you must read this carefully…but everything is marked…

Dont get confused because the landline connector gets mentioned seperately in “the story”‘,

I guess this makes the Fritz!box also compatible in other configurations…. once the hardware is installed, you put in the disc that comes with the Fritz! in your pc drive, and fill in your ADSL inlog info…and the …MySipSwitch inlog account info at “Internet Telephony provider, and you’re up and running,

All phone calls will now go through the internet, exept the emergency numbers.

After this the real fun starts…and you can make it very fancy ! if you want to go that far,

But i guess most of it you will manage from MySIPSwitch,

Use your dialplan settings as always, at MySipSwitch.

Set the Fritz! in “Expert Mode” to see all features and settings, you now should snoop around in the Fritz! web interface, and use the help function, to see what’s possible with that particular function…It’s all reasonably clear to use unlike with the SPA3102, which i found “a bit tricky”

By TheFug

Here is a quick, getting started guide for X-Lite 3.0 for WIndows XP.

1. After the installation, you should have the following screen:

2. Click on the small white arrow which says “Show Menu” and then select : “SIP Accounts Settings”

3. Click on “Add”

4. Now, you need to register your My SIP Switch account settings:

Display name : your name, for instance Joe Bloggs

Username : it is your My SIP Switch username

Password : your My SIP Switch password

Domain : sip.mysipswitch.com

Register with Domain and receive incoming calls must be ticked

Send outbound via : Proxy : sip.mysipswitch.com

Press “Apply” and the OK.

You notice that X-Lite is trying to register and upon success will display “Ready” and “Your username is …”.

Once this registration is fine you can start making test calls.

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