Configuring IP and SIP Settings

If you are connecting Vocera to an IP PBX or a VoIP gateway, set the Integration Type field to IP, then specify settings in the following fields:

Table 1. IP and SIP settings

Field

Value

Signaling Protocol

Specify the signaling protocol that your IP PBX uses. Currently only SIP Version 2.0 is supported.

Call Signaling Address

Enter the call signaling address for your IP PBX or VoIP gateway. For PBX failover support, enter a comma-separated list (up to 256 characters) of call signaling addresses for two or more PBXs or gateways in order of preference. Enter each call signaling address in this format:

IP_Address:Port

The port is optional. If you do not specify a port, port 5060 (the default) is used.

Vocera SIP Telephony Gateway uses only one PBX or gateway at a time. If you specify multiple call signaling addresses, Vocera SIP Telephony Gateway tries each PBX or gateway in the order specified and uses the first one that responds. If that PBX or gateway goes down, Vocera SIP Telephony Gateway switches to another one.

The preference order of call signaling addresses is important. If the Vocera SIP Telephony Gateway is currently using the PBX for the second call signaling address, and then the PBX for the first call signaling address becomes active, Vocera SIP Telephony Gateway automatically switches to the first PBX.

Note: The Vocera SIP Telephony Gateway uses the response to a SIP OPTIONS message to determine if the PBX or gateway is currently available. The OPTIONS message is sent every 30 seconds by default. For more information on how to configure Vocera SIP Telephony Gateway to use an OPTIONS message for keep-alive, see Detecting the Connection to the IP PBX. If the PBX or gateway is not configured to support SIP OPTIONS, then entering a second call signaling address has no effect. In some situations, using TCP as the signaling transport protocol reduces the length of time required for the VSTG to recognize that the current PBX is down and move to the next PBX in the list.

Calling Party Number

Enter the DID number, including the area code, of the Vocera trunk (the number of digits depends on the locale). Outgoing calls use this value as the caller ID. However, you can configure Vocera SIP Telephony Gateway to use caller information contained in the dial signal from the Vocera Voice Server as the caller ID.

Enable Call Trace

Click Enable Call Trace to enable tracing for a number of calls specified in the Vocera SIP Telephony Gateway configuration file (vgwproperties.txt). The default number of calls traced is five. To view the trace, see the vtg-dlog*.txt log on the Vocera SIP Telephony Gateway.

Note: If you increase the number of lines and then save changes, it will cause the Vocera SIP Telephony Gateway to restart. If you decrease the number of lines, change the call signaling address, or change the calling party number and then save changes, those changes will be reflected in subsequent calls made through the Vocera SIP Telephony Gateway.