Call Routing

LanScape's Centrex Proxy Server® includes call routing support so that you can define sequential call routing plans for each user who registers with your VOIP domain. In addition, incoming calls can also be routed to non-registered network destinations. Call routing plans are a powerful  feature and are used when the proxy receives a new incoming phone call. Call routing rules can also be defined using regular expression syntax. This powerful capability will allow you to route whole blocks of numbers anywhere you like (even to another VOIP domain). An example of this is to route all calls beginning with the digit '9' to a PSTN gateway.

Generally speaking, the proxy determines who the call destination is and performs a lookup for a valid call routing plan for that user. If a call routing plan is located for the destination user, the call routing plan is propagated sequentially until the call is answered. If no call destination in the route plan answers the incoming call, the initiator of the call will receive an error indicating that the call destination is not available.

Note: Call "forking" is not supported. Calls propagate sequentially.

There is no limit imposed on the number of call points you can define in a single call routing plan. There is of course a practical limit as to how long you want your callers to wait for someone to answer the VOIP call. You can route an incoming call to your primary location, bounce the call to a few other SIP call endpoints, and then route the call back to your primary location. The sequence does not matter. What you should always do however is make the last call destination of the routing plan a voicemail server that resides in your VOIP domain. That way, when an incoming call is processed and no one answers, the call will be vectored to voicemail.

Call routing uses call processing timeout values that are defined in the Call Processing Timeouts configuration dialog. The important value to keep in mind when creating your call routing plans is the value of the "Call rollover timeout". This value specifies the number of seconds an incoming phone call will be presented to each call endpoint defined in your route plan. So, if you set the "Call rollover timeout" value to 10 seconds, the incoming call will be presented to each endpoint in the route plan for a maximum of 10 seconds.

Note: Any call endpoint in the route plan that is for a registered extension but is not registered, will be skipped. Also be aware that the call will be presented to the final call endpoint of the route plan for a duration of "Final answer timeout" as specified in the Call Processing Timeouts configuration dialog.



The following dialog is used to set up call routing plans:

Enable/disable the ability of SIP user agents to call their own extension numbers. If this is disabled, then a user agent will not be able to call itself even if it's extension appears in its own routing list.




The call routing configuration page consists of a single list control that is used to specify call routing lists. To modify entries in the route list, use your mouse and right click anywhere in the list control. When you right click in the list control, you will see a pop up menu like the one shown in the dialog below:







The operations you can perform are:

Add - Adds a new call routing entry.
Edit - Edit an existing call routing entry.
Delete - delete an entry.
Copy - Allows you to duplicate an entry.
Save - Saves the current changes to the call routing database.

Defining call routing plans - The basics


The "User Name" field:

When you define a new call routing plan, you have two choices. You can specify the "User Name" as a known registered user name or you specify the "User Name" as a regular expression.

Stated another way: For the purposes of call routing, the "user name" of the call routing list can be the name of a SIP user agent that will register with this proxy or it can be a regular expression. Regular expressions are good when large blocks of phone numbers must be routed to a specific destination (like to a PSTN gateway that may or may not register with this proxy).

Note: Regular expression routes are evaluated and applied only after all non-regular expression routes are exhausted.



The "Route List" field:

The call "Route List" basically contains all of the call endpoints that will be used when processing an incoming phone call. The entries in the route list can be the names of other registered users or it can be an IP address and port value for a non registered endpoint. You also can specify that a call endpoint be the fully qualified domain name (FQDN) of another fully proxied VOIP domain. To summarize, the call "route list" can contain any of the following types of destination specifiers:

Registered Endpoints:
The names of SIP user agents in your domain that will register
with the proxy.

Static IP locations not behind another proxy:
Raw IP address and optional port specifier as in x.x.x.x:port.
(If port is not specified, the value 5060 is assumed) This would be
the static IP address of a SIP user agent that is not behind its own proxy.
For example: "1.2.3.4:5060"

Static IP locations behind another proxy:
Raw IP address and optional port specifier as in x.x.x.x:port:p.
(If port is not specified, the value 5060 is assumed) This would be
the static IP address of a SIP user agent that is behind its own proxy.
Note the ":p" (colon - p) suffix at the end of the location specifier.
For example: "1.2.3.4:5060:p"

DNS (or other name resolved) locations not behind another proxy:
The FQDN or UNC host names in the form of \\HostName:port.
(i.e. \\PII450:7000 or \\myphone.dnsalias.com:6000, etc...
If port is not specified, the value 5060 is assumed)
Using this "\\" syntax implies that the call endpoint is a SIP user
agent that is not behind its own proxy server.

DNS (or other name resolved) locations behind another proxy:
The FQDN  in the form of \\\HostName:port.
(i.e. \\\support.lanscapecorp.com:7000 or \\myotherdomain.dnsalias.com:6000, etc...
If port is not specified, the value 5060 is assumed)
Using this "\\\" syntax implies that the call endpoint is a SIP user
agent that is behind its own proxy server (in another VOIP domain).


 

By supporting this capability, you will be able to route calls to any other registered call endpoint and also to other SIP devices or gateways that do not specifically register with the Centrex Proxy Server®.





Defining call routing plans - Using registered user names


Defining call routing plans using registered user names is a simple task. For example, selecting to add a new call routing plan from the pop up menu shown above, will display the following dialog:






Example1 - Routing to registered users and to voicemail:
Lets assume you want to create a call routing plan for "user name" 1234. The term "user name" is synonymous with a phone extension. And lets say you want to route incoming calls for extension 1234 to call endpoints 1234,1236 and then finally to voicemail. Remeber: a SIP call endpoint can have a number or a name assigned to it.



So, to define the call routing plan we just described above, you would need to enter the information that appears in the following dialog:











Defining call routing plans - Using regular expressions


Creating a call routing plan based on a regular expression is a very powerful capability. With this functionality, you will be able to manage whole blocks of user agent names or phone numbers. One particularly good use of this is to route SIP calls to a PSTN gateway only if they consist of the proper number and types of digits.



Example1 - Routing to the PSTN gateway:
Lets assume you have a PSTN gateway located at your facility. You have determined that whenever a user dials '9' before the phone number, they are attempting to make an outgoing phone call to the legacy public switched telephone network. Lets also assume that the PSTN gateway can not register with the LanScape Centrex Proxy Server®. We will also assume that your PSTN gateway has a host name of "PstnGateway", it's IP address can be obtained using DNS or some other name resolution and it listens for SIP traffic on UDP port 5060. We will also assume that the user is trying to call a valid PSTN number in the US - 1-952-943-8250.

The following dialog shows typical information that can be entered into the dialog in order to adhere to the previous description:



Note that when using a regular expression, you must check the "User name\phone number is specified as a regular expression" option. The field in the dialog that is labeled "optional dialing prefix" allows you to tell the Centrex Proxy Server® that it should strip off a leading '9' digit from the destination phone number before it attempts to route the call to the PSTN gateway.

When you use a regular expression as the "user name" in the call route, you can test the validity of your regular expression against real destination SIP user names or phone numbers. Lets assume a user is at their soft phone and they wanted to make the PSTN call in the above VOIP system. They would dial 9 and then the number 19529438250. When their soft phone sends the SIP INVITE to the Centrex Proxy Server®, the proxy sees that the user is attempting to call the number 919529438250. The proxy then matches this number to the regular expression routing rule defined above. The proxy also removes the "prefix" dialing digits from the call destination number before routing the call onto the PSTN gateway. The PSTN gateway then receives a SIP INVITE message with the destination phone number 19529438250. The PSTN gateway then processes the outgoing call as required.