mysipswitch Wiki & Documentation Rss Feed

Updated Wiki: Home

aaronc — Sat, 12 Sep 2009 12:11:51 GMT

This project has been superseded by http://sipsorcery.codeplex.com/.

Project Description
A SIP signalling consolidation tool that allows multi-user management of diverse SIP providers and allows central management of any SIP based VoIP service. Included in this project are a SIP Stack, SIP Registrar, SIP Registration UAC, SIP Stateful Proxy, STUN Server and more. The components are all written in C#.

This project is currently being used to provide the live service at :
http://www.mysipswitch.com

We have a forum to discuss the service, the technical issues, feature requests : MySIPSwitch forum do not hesitate to register and contact My SIP Switch users and developers.

My SIPSwitch 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.

This project is sponsored by Blueface : VoIP

Follow us on Twitter @Blueface_voip

Updated Wiki: Home

gbonnet — Sat, 28 Mar 2009 18:10:18 GMT

Updated Wiki: Home

gbonnet — Fri, 27 Mar 2009 16:48:44 GMT

UPDATED WIKI: Home

gbonnet — Wed, 19 Mar 2008 09:00:46 GMT

UPDATED WIKI: Home

gbonnet — Wed, 19 Mar 2008 08:59:19 GMT

Project Description
A SIP signalling consolidation tool that allows multi-user management of diverse SIP providers and allows central management of any SIP based VoIP service. Included in this project are a SIP Stack, SIP Registrar, SIP Registration UAC, SIP Stateful Proxy, STUN Server and more. The components are all written in C#.

This project is currently being used to provide the live service at :
http://www.mysipswitch.com

We have a forum to discuss the service, the technical issues, feature requests : http://www.mysipswitch.com/forum/index.php do not hesitate to register and contact My SIP Switch users and developers.

My SIPSwitch 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.

This project is sponsored by Blueface : http://www.blueface.ie

UPDATED WIKI: Home

gbonnet — Thu, 14 Feb 2008 10:45:41 GMT

Project Description
A SIP signalling consolidation tool that allows multi-user management of diverse SIP providers and allows central management of any SIP based VoIP service. Included in this project are a SIP Stack, SIP Registrar, SIP Registration UAC, SIP Stateful Proxy, STUN Server and more. The components are all written in C#.

This project is currently being used to provide the live service at :
http://www.mysipswitch.com

We have a forum to discuss the service, the technical issues, feature requests : http://www.mysipswitch.com/forum/index.php do not hesitate to register and contact My SIP Switch users and developers.

My SIPSwitch 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.

User Guide

Thanks Emoci for this quick user guide

Note : You will have better chances to get an up to date version on our forum http://www.mysipswitch.com/forum/viewtopic.php?t=139

Making Calls

1. Register your Software or ATA with MySipSwitch (SSUser, SSPassword, sip.mysipswitch.com)
2. Configure your Dial Plans

a) Let one provider handle all your calls (no *1, *2 etc. required)

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

b) Use Multiple Providers by choosing among them using *1, *2 etc, :

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

So if I wanted to call 14161112222 via provider 1, I would dial *114161112222
If I wanted to dial 14161112222 via provider 2, I would dial *214161112222

c) Route Calls based on number format:

If I 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)

d) Adjust for local dialing

If I 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)
Note the order of Dial Rules

Receiving Calls

- Receiving Calls by registering ATA with MySipSwitch

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

Username: Provider Username
Password: Provider Password
Server: ProviderServer.com or ProviderServer.com:5060 (use actual server as required by your provider)
Domain/Realm [optional] :
Expiry Seconds (60 to 3600) : 3600 (this is the default value, change if required)
Contact: SipSwitchUsername@sip.mysipswitch.com

Add it, and use the Monitoring page to see that everything went fine. Now calls to the third party provider will ring your ATA/SoftPhone registered with MySipSwitch

- Receiving Calls in your ATA registered with another provider

Other provider that will receive the call must support SIP URI calls from third party networks
This is an example on how to use MySipSwitch to forward calls coming in from certain providers via SIP URI to a different location
Register provider in your MySipSwitch account (Config page, under Registrations):

Username: Provider Username
Password: Provider Password
Server: ProviderServer.com or ProviderServer.com:5060 (use actual server as required by your provider)
Domain/Realm [optional] :
Expiry Seconds (60 to 3600) : 3600 (this is the default value, change if required)
Contact: username@ReceivingVSP.com (eg. 123456@voxalot.com)

- Receiving Calls on an actual PSTN/Telephone number

1. Register a provider with MySipSwitch

2. Create a Special DialPlan (preferrably near the end of the list if you have other Dial Plans you are using):

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

3. Check "Use dial plan for incoming calls" checkbox

Now calls to any providers which have been pointed to SipSwitchUsername@sip.mysipswitch.com, or SIP URI calls to SipSwitchUsername@sip.mysipswitch.com will ring the Phone Number you setup.

UPDATED WIKI: Home

gbonnet — Thu, 08 Nov 2007 10:12:06 GMT

Project Description
A SIP signalling consolidation tool that allows multi-user management of diverse SIP providers and allows central management of any SIP based VoIP service. Included in this project are a SIP Stack, SIP Registrar, SIP Registration UAC, SIP Stateful Proxy, STUN Server and more. The components are all written in C#.

This project is currently being used to provide the live service at :
http://www.mysipswitch.com

We have a forum to discuss the service, the technical issues, feature requests : http://www.mysipswitch.com/forum/index.php do not hesitate to register and contact My SIP Switch users and developers.

My SIPSwitch 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.

User Guide

Thanks Emoci for this quick user guide

Making Calls

1. Register your Software or ATA with MySipSwitch (SSUser, SSPassword, sip.mysipswitch.com)
2. Configure your Dial Plans

a) Let one provider handle all your calls (no *1, *2 etc. required)

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

b) Use Multiple Providers by choosing among them using *1, *2 etc, :

c) Route Calls based on number format:

d) Adjust for local dialing

Receiving Calls

- Receiving Calls by registering ATA with MySipSwitch

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

Other provider that will receive the call must support SIP URI calls from third party networks
This is an example on how to use MySipSwitch to forward calls coming in from certain providers via SIP URI to a different location
Register provider in your MySipSwitch account (Config page, under Registrations):

Username: Provider Username
Password: Provider Password
Server: ProviderServer.com or ProviderServer.com:5060 (use actual server as required by your provider)
Domain/Realm [optional] :
Expiry Seconds (60 to 3600) : 3600 (this is the default value, change if required)
Contact: username@ReceivingVSP.com (eg. 123456@voxalot.com)

- Receiving Calls on an actual PSTN/Telephone number

1. Register a provider with MySipSwitch

2. Create a Special DialPlan (preferrably near the end of the list if you have other Dial Plans you are using):

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

3. Check "Use dial plan for incoming calls" checkbox

Now calls to any providers which have been pointed to SipSwitchUsername@sip.mysipswitch.com, or SIP URI calls to SipSwitchUsername@sip.mysipswitch.com will ring the Phone Number you setup.

UPDATED WIKI: Home

gbonnet — Thu, 01 Nov 2007 18:15:12 GMT

UPDATED WIKI: Home

gbonnet — Thu, 01 Nov 2007 17:17:51 GMT

UPDATED WIKI: Home

aaronc — Sun, 28 Oct 2007 08:52:49 GMT

Project Description
A SIP signalling consolidation tool that allows multi-user management of diverse SIP providers.