ServerView VoIP-SIP monitoring alerts you the moment your organization's VoIP communications system has availability or performance issues. Quickly respond to issues with your critical business communication systems and troubleshoot the root causes of poor VoIP performance.
SIP Monitoring is an online VoIP monitoring service that proactively monitors the ability of VoIP infrastructure components to establish and maintain VoIP calls. Proactive VoIP monitoring is realized using Session Initiation Protocol (SIP) a signaling protocol typically used for VoIP.
The SIP Monitoring service acts like an end client, by periodically placing VoIP telephone calls (once per minute, or 3, 5-minute etc...) to a specified number and then checking the results of that call. To accomplish this, SIP Monitoring is provisioned as either an extension, or a client, on the VoIP system and configured to call a specific number using a specified SIP server with certain parameters. The expected result of the call is setup as "Answer", "No Answer", "Busy", or an Error Condition (if there is an unexpected result).
Setting up a monitoring task with the ServerView SIP Monitoring tool is a simple process. First, provision a SIP Monitoring extension on the telephony systems. Next, add a new device (for example "Asterisk") to the Dotcom-Monitor control panel. Then, specify the "Asterisk" device as a "SIP" task. Finally, set the target and parameters of the new task of "SIP" type.
Task Name - Name that identifies the task, for example "Asterisk".
Maximum Connection Timeout (in Seconds) - Time to wait for an expected call result.
Server - Name of VoIP server to be monitored. For example: asterisk.yourcompany.com
Port - This is an optional field. If not specified, Dotcom-Monitor uses default SIP Port 5060
User Name/Password - Credentials used to authenticate within the server. For example: 209 - username and "password". The task does not keep state. SIP Monitoring will login into the system, authenticate, place a phone call, and logout. Dotcom-Monitor does NOT register with the service (instead, SIP Monitoring makes a temporary connection, conducts a test, and th en disconnects)
ANI - This is an optional field. This is a number that used as caller ID. It can be set for some systems to accomplish custom routing for calls generated from SIP Monitoring. For example, the system can be programmed to dial a specific number and if "ANI=209" set a busy tone.
Phone to Dial - Number to dial. This can be any number as long as the telephony system knows how to route it.
Expected Call Result - If a call is successful and did not produce any SIP communication errors, then SIP Monitoring analyzes the result of the call. The analysis can be setup as "No Answer", "Answer", or "Busy" depending on the expected outcome.