1.4.3 Destinations

The Destinations menus allow to create a destination endpoint with its settings which then can be reused for many routes defined in the Routing plan. The entries in the routing refer to the destinations by name. Each menu let you add a new instance of a given type and manage the existing definitions grouped in the lists.

Gateway

By a gateway we mean here an endpoint which location is known and explicitly provided as an IP address or FQDN. A gateway will not register to voipswitch’s SIP registrar. Usually they are SIP servers, softswitches or PSTN gateways which provide a bridge between SIP and traditional telephony. The gateway type objects are stored in the voipswitch.gateways table.

The most often used scenario is connecting to a carrier providing calls termination. For example if we have traffic to PSTN numbers originated by our customers from the mobile VoIP softphone and we need to terminate the calls. The carrier will give you its SIP interface’s IP address or FQDN and the port and protocol (usually UDP). Now you will need to add a new entry in the Gateway menu.

To add a new gateway click on the ADD button located above the list of gateways.

In the popup window enter the basic data which include a name, IP number and port and the calls limit. The name is how you name this particular carrier, it will be used then in the reports and CDRs.

General:

• Calls limit – max number of calls (both connected and pending) going thru the gateway

           Note: 0 means there is no limits.

• Calculate costs – allows to assign a cost tariff, i.e. a tariff with rates which we pay to the carrier, the tariff must be in the base currency.
  The interstate/intrastate option in the cost section is for the US based providers which originate and terminate calls within the USA. 

           When the check-box is selected a tariff from the combo-box must be chosen. This tariff will be used to calculate the costs of connections after finishing every call terminated with this gateway.

           The tariff used there is defined in the same way as any other tariff in VoipSwitch. A good practice is to name such tariffs differently from tariffs being used for clients.

           Cost calculations allows comparing bills received from carriers or to see the profit from calls made by clients

In the edit mode there are more options grouped in the General and Advanced tabs:

Advanced General options:

• Calls limit – max number of calls (both connected and pending) going thru the gateway
• Calculate costs – allows to assign a cost tariff, i.e. a tariff with rates which we pay to the carrier, the tariff must be in the base currency.
  Note: 0 means there is no limits.
  The interstate/intrastate option in the cost section is for the US based providers which originate and terminate calls within the USA. 
• Dialing number prefix – the dialed number modification rules, the number is modified by voipswitch before the call is sent to the gateway. For example some carrier may ask you to add a special prefix before the number.
• Tariff prefix – when the calculate cost option is chosen and a cost tariff assigned this field will be shown in the Dialed number manipulation section. VoipSwitch will modify the dialed number before calculating the gateway cost.
• SIP username and password – in most cases when sending traffic to a wholesale carrier the authentication will be based on the IP address, it means you will have to provide your voipswitch’s IP address or addresses which will be allowed on the carrier side. In some cases however we may need to use SIP digest authentication with username and password – they have to be provided by the carrier. If the authentication is by the IP then these fields can remain blank.
• Send Remote-Party-ID - the Remote-Party-ID header required by some carriers

Codecs:

•  Codecs– the list of allowed codecs. These codecs are supported by the gateway, however we can limit them to only desired ones. For example even if the gateway supports g711 we may want to force it to use g729. The voipswitch will be trying to find a gateway with a codec matching one of the codecs offered by the client. If not found the transcoding module can be used – this feature is described in a dedicated chapter

If the carrier has more than one gateway you should create separate entries in the gateways list each with different host address and other details.

Registrar

This type of destination endpoint is for SIP registrars in the meaning that voipswitch itself is the client endpoint registering to the remote registrar server.

All the settings are identical to those of the gateway type instances with the difference that there is an additional section Registration. This section contains all parameters required to register to a registrar.

A registrar type will be used when you get a SIP trunk from a provider and it will require that you keep SIP registration. In such case you are like a retail client, the provider cannot send calls to you until you appear in his location server.

The registrars are stored in voipswitch.gatekeepers table. The table name is from old times of h323 protocol.

If the registration is not required you should always use the gateway type.

Lookup

The Lookup is a voipswitch function allowing to run a customized SQL query on voipswitch database or to send an http request to a specified URL. When adding a new lookup entry you have to enter the query string. The variable in the query is the dialed number and several other call details (the full list described further). When a call comes the dialed number is passed to the lookup function which triggers the query. The result is sent back to the routing procedure as a new number so the voipswitch can make a new search in the Routing plan to find the destination for the new number.

The lookup feature enables implementation of many advanced scenarios. One of them is a lookup for ported numbers where a provider has to find the current mobile operator of the dialed number.

Lookups are stored in the voipswitch.lookups table.

In the VSM the list of the lookup instances is in the Lookups menu.

Each lookup consists of the following properties:

• Description - name of the lookup scenario
• query_string - the lookup string, two types of strings are allowed – sql query and http request with prefixes  "sql://" or http:// respectively

When creating a query string you can use the below variables. Their values are taken from the call.

• #number#       the dialed number after the modifications using the dialing rules in the Routing plan
• #cli#    caller ID
• #caller_ip_number#    IP address of the calling endpoint
• #id_client#      ID of the client, form voipswitch.clientsshared table
• #client_type#  numeric type of the client, from voipswitch.clientsshared table

Examples:

• Lookup scenario for ported numbers
• How to setup a dynamic list of blacklisted numbers

Enum route

An enum route is a DNS server which resolves phone numbers to SIP URIs. The examples of enum providers are e164.arpa and e164.org. They store information on user’s phone numbers and their mappings to SIP accounts on different domains.

Callback route

The Callback route type instances allow to define different settings of callback services. The instances are triggered by associated with them callback scenarios. More on the Callback configuration in a dedicated chapter further in the document.