MySIPSwitch

My SIP Switch

My SIP Switch official site

MySIPSwitch is a VoIP open-source software under BSD licence and a free service. This is an experimental stateful SIP Proxy server sponsored by Blueface to allow multi-user management of diverse SIP providers and allow central management of any SIP based VoIP service.. It basically means that you can use many SIP accounts with a single piece of hardware (IP Phone, ATA or softphone), you can even use SIP with a normal phone. The range of possibility is wide and getting wider very often due to improvements from the team of developers.

MySIPSwitch allows as well to handle SIP phone calls from the web browser: hangup, transfer, forward, hold/resume.

"The registration is quite simple and straight forward" : complete article.

The development language is C#, the database environment is PostgreSQL and the interface is a website.
The source code can be downloaded on SIPSwitch's page on Codeplex.

Features

SIP account creation
Setting up a customised dial plan
Setting up 3rd party SIP Registrations
SIP traffic forwarding
SIP Accounts activity monitoring via the website
SIP traffic monitoring via telnet
Online switchboard: call hold/resume, call transfer/forward, call hangup
Usual security features

How it works

Set up:

First, you have to create an account. Then you have to set up your piece of hardware (SIP Phone, Softphone or ATA) with this SIP account. The SIP server should be : sip.mysipswitch.com. Then, login MySIPSwitch website and set the dial plan and the 3rd party registrations.

Making Calls ( Dial plan )

The first thing you have to set up, when using MySIPSwitch, is your customised 'dial plan'. You can list your different SIP accounts (login, password and server) and set a prefix for it. Note that you can, as well, specify an account to be used by default.

- 'Let's say you had two SIP accounts'

You can set up the dial plan so that, when you dial out, if you press *1 (as a prefix!) you'll phone via sip.my1provider.com and if you press *2 you'll phone via the other one.

The corresponding configuration lines would be :

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

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

- 'Route Calls based on number format'

If you wanted to route all North American numbers dialed in 1-xxx-xxx-xxxx format via one provider and long distance numbers via another provider:

exten => _1X.,1,Switch(user,pass,${EXTEN}@NorthAmericalProvider.com)

exten => _011X.,1,Switch(user,pass,${EXTEN}@LongDistance.com)

- 'Adjust for local dialing'

If you wanted to be able to dial all North American numbers without dialing the 1 (just xxx-xxx-xxxx) while leaving long distance dialing intact:

exten => _011X.,1,Switch(user,pass,${EXTEN}@LongDistance.com)

exten => _X.,1,Switch(user,pass,1${EXTEN}@NorthAmericalProvider.com)

There are more advanced possibilities since this is based on Asterisk's syntax but quite dedicated to technical people.

Source : How to Use My SIP Switch

Receiving Calls ( Dial Plan and 3rd party registrations )

Now, you have to set up your MySIPSwitch account in order to receive calls. MySIPSwitch can perform SIP 3rd party registrations, that means that it can register your SIP account on your provider's servers as a SIP client would do. It can as well forward SIP signalling so you can register your device with another provider and concentrate the SIP signalling in My SIP Switch anyway.

- 'Receiving Calls by registering ATA with MySipSwitch'

1. Register your ATA with MySipSwitch

2. Register provider in your MySipSwitch account (Config page, under Registrations):

The main info here is :
Contact: SipSwitchUsername@sip.mysipswitch.com

- 'Receiving Calls in your ATA registered with another provider'

Note that other the provider that will receive the call must support SIP URI calls from third party networks

Register the provider in your MySipSwitch account :
The main information here is : Contact: username@ReceivingProvider.com

- 'Receiving Calls on an actual PSTN/Telephone number'

1. Register a provider with MySipSwitch

2. Create a Special DialPlan line:

exten = SipSwitchUsername,1,SwitchCall(user,pass,NumberToCall@OutgoingProvider.com)

3. Check "Use dial plan for incoming calls" checkbox in the configuration page.

Source : How to Use My SIP Switch

Audio Codec:

MySIPSwitch has 'nothing to do with media'. It deals only with SIP Signaling so it does not affect voice quality at all.

According to Vinay here: "when we tested their SIP registration, the call quality was excellent and so you do not need to worry which codec will be used to carry your voice."

Benefits

If you are using multiple SIP accounts and if your equipment allows only one, MySIPSwitch is a great tool.
It is possible to take an incoming call from one provider and forward it onto another provider without having to have a device registered. This makes it possible to receive a call without the need for a VoIP device at all. more info

Citation from Andy Abramson : "There is something here to keep me thinking about the possibilities as more users and business get on SIP." (Main article: more info)

Source code

The code is downloadable from CodePlex : MySIPSwitch on CodePlex
They used to be on SourceForge but changed recently.

History

November 2006 : Created by Aaron Clauson from Blueface
January 2007 : Improvements thanks to user's feedback
April 2007 : New features: call management functions, new dial plan interface
July 2007 : Creation of a dedicated forum : more info
August 2007 : Bugs corrections and 3rd party registrations improvements
October 2007 : New layout (it now looks much better)
October 2007 : new feature: possibility to use the dial plan for incoming calls : more info

voip-info.org

My SIP Switch

Features

How it works

Benefits

Source code

History

References

External links

Comments