How to use My SIP Switch

Coordinator
Nov 1, 2007 at 7:02 PM
Edited Nov 2, 2007 at 10:18 AM
Hi all,

If you need help or feedback from other users or project admins, please, use our forum: http://www.mysipswitch.com/forum/index.php. Thanks!

Here is a quick user guide done by one of our user. Thanks Emoci.


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

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

  • 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.
Apr 7, 2009 at 11:34 AM
Is mysipswitch working on live IP