In this section:


A SIP user agent (UA) is a logical network endpoint used to create or receive SIP messages and thereby manage a SIP session. A SIP UA can perform the role of a User Agent Client (UAC), which sends SIP requests, and the User Agent Server (UAS), which receives the requests and returns a SIP response. These roles of UAC and UAS only last for the duration of a SIP transaction.

A SIP phone is a SIP user agent that provides the traditional call functions of a telephone, such as dial, answer, reject, hold/unhold, and call transfer. SIP phones may be implemented as a hardware device or as a softphone. As vendors increasingly implement SIP as a standard telephony platform, often driven by 4G efforts, the distinction between hardware-based and software-based SIP phones is being blurred and SIP elements are implemented in the basic firmware functions of many IP-capable devices. Examples are devices from Nokia and Research in Motion.
In SIP, as in HTTP, the user agent may identify itself using a message header field 'User-Agent', containing a text description of the software/hardware/product involved. The User-Agent field is sent in request messages, which means that the receiving SIP server can see this information. SIP network elements sometimes store this information, and it can be useful in diagnosing SIP compatibility problems.

FXS Ports

FXS ports are standard analog telephony interfaces that are used in Public Switched Telephone Network (PSTN) networks. The FXS interface is the modular wall plug that connects the telephone to the central office (CO) of the PSTN. The CO delivers power, dial tone, and ringing capabilities by way of the FXS interface.

Warning

After a reboot or refresh from a configuration change, the FXS ports may report as "Unlicensed". This means the hardware is still starting up and loading the configuration, or there is cached web page data in the browser.

To resolve this issue, either wait 1 to 2 minutes and refresh the page, or configure the ports and submit. 

Transmit/Receive Gain

You can configure the transmit and receive gain setting for each FXS port on the EdgeMarc. Most devices operate with the default 0dB Rx gain setting, but you can adjust the setting to interoperate with user endpoints such as phones, fax, or key systems.
When an endpoint device is hooked to the FXS port, it may be necessary to adjust the receive gain settings if the port is unable to detect the digits sent from the device. Adjust the gain in steps of -4dB using the FXS port Tx/Rx gain settings until the digits are detected.

Dialed-in Prefix

Dial-in prefix is used in PBX deployments to provide a way to manipulate the incoming dial pattern for 4-digit dialing. If the dial-in prefix matches the beginning of the incoming dial string, the prefix is stripped from the incoming dial string before that dial-pattern is forwarded to the PBX. For example, if the dial-in prefix is 408555 and the dial-in string is 4085551234, the dial pattern given to the PBX is 1234 after stripping. If the dial-in prefix field is not specified, the dial pattern sent to the PBX contains all 10 digits (10-digit dialing), for example 4085551234.

Ad Hoc Conferencing

The EdgeMarc supports ad hoc SIP conferencing using conference Uniform Resource Identifier (URI). The conference URI identifies a resource in the SIP/IP network that can handle conferencing and media mixing. When the EdgeMarc is configured for ad hoc conferencing, the FXS port uses this configured network for third party conferencing. This is always the recommended way to do 3 way calling.
Analog Phone Calling Features on the FXS Port

All EdgeMarc platforms support the following features for analog phones connected to the FXS port:

  • Call Hold
  • Call Transfer – Unattended
  • Call Transfer – Attended
  • Call Waiting
  • 3-Way Calling

Configuration Optimization

Configuration optimization allows you to make GUI configuration changes on the SIP, SIP UA, and SIP UA Advanced configuration pages and effect the network restart while keeping call interruption to a minimum. Hover your mouse over the red and black asterisks on the SIP, SIP UA, and SIP UA Advanced configuration pages to see the pop-up tool tip which tells you whether or not a configuration change will interrupt calls.
SIP UA configuration port-specific configuration values can be changed without interrupting any other ports and with only a short down time by avoiding the network restart.
Selected system configuration parameters are now re-loadable so that they can be changed and put into effect without restarting the ALG module.
Configuration optimization is disabled by default.