Routes Management

In order for IPBrick to execute the routing of calls between the several network interfaces, it is necessary to define the specific routes according to a telephony numbering.

As you can see in Figure 4.120 we have this options:

• Local Routes: Represents all the local interfaces available in IPBrick by default;
• Outbound routes: Represents all the outbound routes, so it will be possible to make calls using SIP/IAX accounts;
• SIP servers list for registering: Allow to receive calls for SIP numbers associated to SIP accounts;

Figure 4.120: VoIP - Routes Management

Local routes

Local routes (Figure 4.121) allow the configuration of an interconnection between LAN, PSTN, PBX or INTERNET.

The possible options by default are:

• PSTN-LAN: It allows the call routing from the telephone network to the VoIP phones of local network. So it's a internal IPBrick route than can redirect the received calls from the PSTN to VoIP phones;
• PBX-LAN: It allows the call routing between the telephones connected to the PBX and the VoIP telephones of local network;
• LAN-PBX: It allows the call routing from the VoIP telephones in local network to the telephones of the PBX;
• LAN-PSTN: It allows the call routing from VoIP phones to telephone network;
• INTERNET-PBX: It allows to accept VoIP calls from the Internet and route them to PBX phones. It's a IPBrick internal route only for call redirection;
• INTERNET-PSTN: It allows to accept VoIP calls from the Internet and route them to the telephone network network. It's a IPBrick internal route only for call redirection;
• PBX-PSTN: This is a default internal route. It allows the call routing from the PBX to telephone network. ^4.22
• PSTN-PBX: This is a default internal route. It allows the call routing from the telephone network to the PBX.

If there are other configured interfaces (acting like trunks), they may be added to the list of routes, and for that it is necessary to click the connection Available Local Routes (Figure 4.121) and then add the necessary routes.

Figure 4.121: VoIP - Local Routes

The Insert in the top menu allows to insert one of the routes mentioned. After insertion, each type of route has a connection that allows its configuration. When acceding to this interface it is possible to choose one of these options:

• Back
• Modify: To change the type of local route;
• Delete: Remove the local route;
• Insert: It allows to add the prefixes that must be added to this route. When you indicate a prefix, all the calls whose initial digits coincide with that digit are routed by that route. Choosing Advanced Options we have this options (Figure 4.125):
  • Prefix: The numeric prefix to use to make calls using that route;
  • Include prefix in address: If Yes the prefix will be part of the destination number, so the prefix will be maintained when the call is routed. If No the prefix will be used only to identify the route. Example: To enable the use of number 6 to route a call to the Portuguese PSTN network, it is necessary to remove this prefix in order that the number stays with the correct format (the format 2XXXXXXXX instead of 62XXXXXXXX).
  • Postrouting prefix: It's a prefix added by the IPBrick when the number is received. Example: For the Portuguese PSTN network we use the format 2XXXXXXXX. If we use has main route a SIP account route it's necessary to use prefix 2, include prefix in address and use a postrouting prefix with 00351 (351 is the portuguese international code);
  • Caller IDs restriction: Will restrict the route only for the listed caller ID's;
  • Fallback routes: It's a backup route to use if the present one fails;
  • Generate local ringing tone: Will generate a local ringing tone. Can be used when it can't ring at the destination phone;
  • Priority: Define the prefix priority level.

Outbound routes

This option enables you to configure which calls shall be routed to a external server which, in turn, shall be responsible for routing them to their destination (Figure 4.122). This routing is made through prefixes that may be inserted clicking the name of the route and then the link Insert above the prefixes table. To change or remove a route you only have to click its name and then the option Modify or Delete, respectively.

To add a new outbound route click Insert. The Basic Options are:

• Type: Type of signalling protocol to use: Can be SIP, SIP with TLS, IAX or Local;
• Name: Outbound server name;
• Server Address: Server IP/name address;
• Authentication: If authentication is necessary at the server, you will have to choose the option User/Password and fill the users name and respective password;

Choosing Advanced Options the following parameters will be presented:

• Server Port: Server port to use;
• Video support: If the VoIP server supports video, you can enable that option;
• Caller identifier: Outbound caller ID masking;
• Registration realm: Realm is usually the SIP server FQDN but some SIP servers have different server address and registration realm;
• Outbound proxy: Usually not used but is a server that passes the SIP messages between the SIP client and the SIP proxy server;
• Available to Internet: With this option selected, the route shall be available for VoIP telephones outside the LAN;
• Simetric signalling: It allows to define if signalling is sent and received through the same port (5060);
• Enable ENUM lookup: It allows IPBrick to search through ENUM.^4.23

• Enable DUNDi lookup: It allows IPBrick to search through DUNDi.^4.24

• DTMF type: Type of DTMF^4.25 to use. Options: RFC2833 (default), Inband, Info and Auto;
• Call limit: Number of possible simultaneous calls using that route, that can be useful for bandwidth control. With 0 we can disable it;
• State check: This feature permits you to verify if the entity is online or not, i.e.: on a phone it will check if there's online activity, if wether by accident or deliberate the IPBrick will know that the entity is no longer available. By default this field is set to No;
• No far-end NAT detection by provider: This option applies in a context where there's a route to a operator that also sends calls, i.e. not just a an outbound operator.
• P-Asserted-Identity: Some servers are now requiring that packets include P-Asserted-Identity in the header of SIP packets. If this is the case, please activate this option.
  

  NOTE: At Authentication you will have to choose User/Password in order for this option to be available.

• Caller ID on blind tranfers: Support for configuring the caller ID on blind transfers. Available options are:
  • Use the caller ID of the phone that transfers
  • Use the caller ID of the transfered phone

Figure 4.122: VoIP - Outbound route definition

If the outbound route type is IAX, the only parameters are:

• Name;
• Server Address;
• Server Port;
• Available to Internet;
• Call limit.

If the outbound route type Local, the basic parameters are:

• Type: Local
• Name: Type your local server's name

Figure 4.123: VoIP - Local - Basic Options

But if you click on Advanced Options a new set of fields will appear:

• Caller identifier: Outbound caller ID masking;
• Available to internet: By default, NO;
• Call limit: (by default, 0 to disable)
• Mandatory Route: A mandatory VoIP route will have priority over any other. Dialed numbers (including prefixes) associated to a mandatory route will overlap any other match, even if this match is with a local phone, an alias or any other direct access (including IVRs, Attendances Sequences, queues, etc.);
• Caller ID on blind tranfers: Support for configuring the caller ID on blind transfers. Available options are:
  • Use the caller ID of the phone that transfers
  • Use the caller ID of the transfered phone

Figure 4.124: VoIP - Local - Advanced Options

Click on the Insert button to create the route.

Prefixes

The prefixes inserted in any of these outbound routes shall be available automatically for the SIP telephones and the telephones connected to PBX.

These are all the available options (advanced):

• Route name: Type your route's name
• Prefix-Number Pattern: Both fields define the destination's number format that enables you to make calls by using this route. Prefix will be a number and for the Number Pattern you may use this syntax:

  X matches any digit from 0-9
  Z matches any digit from 1-9
  N matches any digit from 2-9
  [1237-9] matches any digit or letter in the brackets
  (in this example, 1,2,3,7,8,9)
  [a-z] matches any lower case letter
  [A-Z] matches any UPPER case letter
  . wildcard, matches one or more characters
  ! wildcard, matches zero (none) or more characters immediately
  

• Include prefix in address: By default YES
• Postrouting prefix: It's a prefix added by the IPBrick when the number is received. e.g: For the Portuguese PSTN network we use the format 2XXXXXXXX. If we use has main route a SIP account route it's necessary to use prefix 2, include prefix in address and use a postrouting prefix with 00351 (351 is the portuguese international code);
• Caller IDs restriction: Will restrict the route only for the listed caller ID's. Click on the ADD button to insert a restriction;
• Fallback routes: Should any failure occur, you may set an alternative route. Click on the ADD button to insert a fallback route;
• Generate local ringing tone: It generates a ringing tone to the calling party. By default, NO

Figure 4.125: VoIP - Prefix definition

Codecs

For each outbound route it's possible to set which codecs are going to be used (click on Modify) as well as their priority (Order option).

Figure 4.126: VoIP - Codecs

SIP servers list for registering

Here it's possible to visualize the SIP^4.26 address list which have already been configured (Figure 4.127). When inserting a new one, the page generated asks for the following data:

• Name: Server name;
• SIP server address: SIP server IP or address. It is possible to specify the port number along with the SIP IP address as: <server address>:<port number> (e.g: 212.12.34.1:5090)

After typing in the data, it is necessary to click the button Insert to confirm the address. The next step is to register accounts to the local SIP server. Pressing Insert we have this options:

• Login: SIP account login. Normally it's the nomadic SIP number;
• Authentication user: Usually the same as the login;
• Password: SIP account password;

Note: To configure the internal number to where the operator will transfer the incoming calls from the Internet to that nomadic number, just create a Phone at Phones Management, IVR, Call Group etc. at Functions > Inbound.

Note: In order to define the destination of the received calls, you should configure a Speed Dial or use a function to that purpose.

Figure 4.127: VoIP - SIP server for registering

Footnotes

... network.^4.22

It's possible to call from phones connected to PBX and, if IPBrick is connected to PSTN and to a PBX, you can also answer calls. IPBrick will work in a transparent mode, switching all the traffic from PBX to PSTN and vice-versa.

... ENUM.^4.23

Group of protocols that aims to associate the telephonic numbering to a new register in DNS. This way, a telephone number shall correspond to a SIP address.

... DUNDi.^4.24

A peer-to-peer system for locating Internet gateways to telephony services. Unlike traditional centralized services (such as the remarkably simple and concise ENUM standard), DUNDi is fully distributed with no centralized authority whatsoever.

... DTMF^4.25

Dual-tone multi-frequency

... SIP^4.26

Session Initiation Protocol