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

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

Hi,

Here is a quick configuration guide for the IP Phone : Siemens C450IP.

Important : Note that there maybe differences in naming or on the path to reach such or such screen depending on the firmware of your phone.

1. Get the connected physically

1st connect the phone to your router and then connect the power cable. This phone is simple so nothing tricky there.

Your phone must get an IP address, so if you are not using a DHCP, make sure to allocate an IP to your phone.

2. Get the phone’s IP address

Go to Menu > Settings > Base > Local Network (or VoIP and then Local Network, depending on your firmware) > Enter the system PIN (default is 0000) > IP Address

3. Connect the the web configuration interface

Open your favorite web browser and in the address bar enter the IP address of your phone

Write down the IP address of your phone and press enter.

You should see the interface below:

4. Enter the system pin

Click on “Settings”.

In IP configuration, by default, you won’t need to change anything. If you are not using a DHCP, then make sure your phone will always get its IP otherwise, you may encounter problems …

5. Telephony settings:

Select “Telephony”, and you should see the screen below:

In “Dial Plans”, you should not have anything to change (since you can handle that better with My SIP Switch).

In “Advanced Settings”, you may want to change things, but later : DTMF options especially (they should NOT be set as “Inband” if you want to use it with My SIP Switch)

Click on “Connections” and then “Edit”

You should see an interface containing the screen below:

Replace yourMSSusername by your actual MySIPSwitch username and edit your password.

Domain and registrar server should be sip.mysipswitch.com

Then press “Set” at the bottom.

If you go back to the page “Connections”, you should see the status as “Registered” (it may take up to 30 seconds)

5. Testing

Now that the phone is registered, you test incoming and outgoing calls.

If you don’t have sound or if the sound is only 1-way. That is more than likely a NAT problem. The 1st thing to try there is to add a STUN server in the configuration of your IP Phone, there are many free ones on the market. One I recommand often is : stun.xten.com

If you need further help, please, post on My SIP Switch’s forum

By Guillaume

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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