Marratech Manager, SIP Usage Administration

This part of the admin administration user interface controls the SIP interface that can be used for calling SIP enabled services, such as a IP telephones or phone gateways for inclusion of land or cell phones (PSTN) into e-meetings.

	Start by opening the Advanced section of the administration interface. From here, press the Add New Module to add a new Dial Out gateway module. In the first step, add the new module to the central node. It is preferable to choose *Automatic* startup in the next step and then confirm the settings with Finish. From the left hand side menu, please select Dial Out to add a SIP line.


	To add a new SIP line press Add line under Dial Out which will take you to a new web page for adding a new SIP line. Note that a Manager might have several defined and active SIP lines at the same time. Here is a description of the various fields.

*Type*	Select SIP if your are creating a SIP line for your Manager. Consult the H.323 section for information on how to Dial Out to a H.323 Video Conference end point.
Name	This is the name of the SIP line. This is the only field that is compulsory to fill in. All the following fields are optional depending on usage. Example: SIPProvider1
Description	This is a longer description of the SIP line. Example: This line is used to connect to SIPProvider1.com for access to external phone lines.
SIP Server	This is the service main server that handles the requests for this line. Example: sipprovider1.com
Outbound Proxy	This can be used if a SIP proxy is used for communication with the SIP Server. Example: sipproxy.mycompany.com:5060/UDP If other than the default, 5060, proxy port may also be specified. Example: sipproxy.mycompany.com:5070
SIP user ID	The SIP user identity to use when using this SIP line. Example: Peter.Parnes@marratech.com
Authenticate ID	If authentication against the SIP server is required then enter the user name here. Example: qWk43J
Authenticate Password	If authentication against the SIP server is required then enter the password here: Example: jj42gy
Register	Is registration against the SIP server needed before it can used? Some SIP services require this. Example: Yes
Multiple calls	Does this SIP service allow for several simultaneous outgoing calls? Example: Yes
Bind address	Use this function to bind to a specific network interface if your server has more than one active interface. Example: 10.1.1.85
Call Port Range	At least a single UDP port is needed. If multiple SIP lines are defined then make sure that each SIP line gets its own port. If using NAT, this port(s) must be directly mapped. For a SIP line it will be the first port in the range that is used. Example: 5060 to 5160 Note that the call port range for a H.323 uses TCP ports (see the chapter regarding H.323).
Data port range	Two UDP ports are used per active SIP call and this entry configures the first data port in the range. Note, that when no more ports are available then no more calls can be made on this SIP line. Example: From 53000 To 53099
External (NAT) IP	If behind NAT, what is the external IP the calls originating from this Manager will have? Example: 195.196.47.2
Allowed Users	Which users are allowed to use this SIP line. This is the normal user handling in the Manager.
Enabled	Is this SIP line enabled or not. Obviously, a disabled SIP line cannot be used and this setting allows the administrator to disable a line but keeping the settings. Example: Yes

One or more SIP lines can be added and to edit or view the settings of a SIP line, press Edit in the main Linesmenu. To remove a SIP line, press Remove in the main Lines menu.

To view and control active phone calls press Call status. Note that this call status will show all active calls and will allow the administrator any active call.

Please consult the Marratech support pages and the Marratech forum for more information about compatible SIP-phone gateway providers.