The Benefits of My SIP Switch

Most SIP based devices are limited regarding the number of SIP accounts they can connect to. One piece of hardware typically allows only one or two connections. So if you wish to connect to more than one SIP account your options are limited. You can either:

Reconfigure the device each time you wish to swap SIP accounts.
Alternatively you can submit a request to one of your current providers to set up a prefix to allow calls to be routed onto the additional SIP servers.

Neither case is ideal or practical for large numbers of SIP accounts and users will generally limit themselves to the services of one or two SIP providers for convenience and simplicity.

My SIP Switch has no such limitations. It is now possible to set up customisable routing and dial plans to as many different SIP providers as you need. All you need to do is create a My SIP Switch account and manage your accounts on line.

The Blueface development team thinks My SIP Switch is a great tool and it was created because you the user asked for such a service. It enables you to take advantage of 5, 10 or 20 different SIP Providers anywhere in the World. This will allow you to tailor fit a VoIP service to meet your needs from several different providers, thus guaranteeing you the best call rates possible.
For extra information, please, check our FAQ.

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How it Works

My SIP Switch's accounts configuration is made around a "Dial Plan" which is based on Asterisk's syntax. You will need to register your SIP accounts on your My SIP Switch configuration page.
A sip signalling monitoring page is there to debug and test the configuration.

Incoming calls

My SIP Switch performs 3rd party SIP registrations. That means that it registers to each of your providers in order to receive all your calls. You need to set up these registrations.
Then, you can set you configuration to receive all your calls, from all your SIP accounts, to the phone that you configured with your My SIP Switch account or you can forward all calls to a PSTN (POTS) number or a mobile or another VoIP number.
An option that you may like is that you can configured My SIP Switch to ring on different phones (for instance your home VoIP phone, your mobile and your office phone), the first one to pick up the call is connected and the others stop ringing.

Outgoing calls

If you want to benefit from the best call rates from different VoIP providers, you may want to call with a specific provider depending on the destination or depending on the time of day. With My SIP Switch you can configure your dial plan so that when you dial a prefix, it calls via a specified provider.

Moreover, for outgoing and incoming calls, you can set up backup SIP accounts to forward the call to/with. For instance, if you want to call via provider1 and for some reasons your SIP account at this provider is not available (no credit, down time of the provider), My SIP Switch will attempt to make your call via provider2 (if you have set your dial plan that way) ... There is the same feature for incoming calls.

Here is an example of "dial plan" to show some possibilities:

exten => _06X.,1,Switch(provider1)
exten => _0ZX.,1,Switch(provider2)
exten => _*1X.,1,Switch(${dst:2}@provider3)
exten => _0039X.,1,Switch(provider1&0${dst:4}@provider4)
exten => _X.,1,Switch(provider5)
exten => user,1,Switch(user@local&0039123456789@provider1)

Note that this is only an example, you can set up something simpler or more complicated. If you are not a technical person, don't be afraid! There are no specific knowledge required to start mysipswitching. If you are skilled in VoIP and especially with Asterisk, then you should be interested in playing with that.

We'll give explanations line by line but first of all, be aware that My SIP Switch will read line by line from top to bottom. Therefore mind the order of your lines, since if a dialling rule is applied, then, all the ones below won't be read. The trick is to go from specific prefixes to more general ones.

Notes :
X matches any number, Z matches any digit from 1 to 9, and N any digit from 2 to 9 (not used in this example).
provider1, provider2, provider3, provider4, provider5 represents 5 SIP accounts that are registered for this account.

- Line 1 : exten => _06X.,1,Switch(provider1)
If I dial out any number starting with 06 the call will get through provider1.

- Line 2 : exten => _0ZX.,1,Switch(provider2)
If I dial any number starting by 0 and the a digit from 1 to 9 (so 01, 02, 03, 04, 05, 06, 07, 08, 09) My SIP Switch will match this line and the call will get through provider2.
Note that if the call started by 06, My SIP Switch would have matched the line before and this line wouldn't have been read at all! If these two lines were inverted, the line which specifically matches 06 would never be used.

- Line 3 : _*1X.,1,Switch(${dst:2}@provider3)
If for a specific reason I want to call via provider3, I will need to dial *1 before dialling the destination number.
My Sip Switch will need to remove the *1 before dialling out, '${dst:2}@' allows this. the 2 there, means that the first 2 characters will be removed. Note that dst and EXTEN can be swapped with one another. That's the same keyword for My SIP Switch.

- Line 4 : exten => _0039X.,1,Switch(provider1&0${dst:4}@provider4)
If I'm calling to Italy, the call will go through via provider1 by default. The number in that case is kept as diallied (in international format). If for any reason, my SIP account at provider1 can't be used (not credit, down time of the provider ...), the call will automatically go through provider4. Note that provider4, is an Italian provider and therefore my number needs to be dialled using Italian national format, so I need to remove 0039 and to add a 0 in front. That's the aim of '0${dst:4}'.

- Line 5 : exten => _X.,1,Switch(provider5)
That's the default line of my dial plan. It matches everything, so if My SIP Switch doesn't match any of the previous line, this line will be used.

- Line 6 : exten => user,1,Switch(user@local&0039123456789@provider1)
This last line, is the line for incoming calls. 'user' represents your My SIP Switch username. There, the line means that all calls will ring on both, the phone registered with My SIP Switch and the 0039 ... phone. This last call leg will be paid to provider1. If you pic up 1st the phone registered to My SIP Switch, then the other stops ringing (and vis versa), so that's handy if you want to be always reachable.

Old syntax (for info):
exten => _*1X.,1,Switch(user,pass,${EXTEN:2}@providerproxy1.com)
exten => _*2X.,1,Switch(user,pass,${EXTEN:2}@providerproxy2.com)

There are many more possibilities to customize the way your outgoing calls are performed.