Cisco WRP500, Administration Manual

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R E V I E W D R A F T # 1 — C I S C O C O N F I D E N T I A L
3-5
Cisco WRP500 Administration Guide
Chapter 3 Configure Voice Services
ATA Software Features
Configurable Dial Plan
with Interdigit Timers The WRP500 has three configurable interdigit timers:
• Initial timeout (T)—Signals that the handset is off the hook and
that no digit has been pressed yet.
•
Long timeout (L)—Signals the end of a dial string; that is, no more
digits are expected.
• Short timeout (S)—Used between digits; that is after a digit is
pressed a short timeout prevents the digit from being recognized a
second time.
See “Configure Dial Plans,” on page 10 for more information.
Polarity Control The WRP500 allows the polarity to be set when a call is connected and
when a call is disconnected. This feature is required to support some
pay phone system and answering machines. Polarity Control is
configured in the Line and PSTN Line tabs. See
Appendix A,
“Advanced Voice Fields.”
Calling Party Control Calling Party Control (CPC) signals to the called party equipment that
the calling party has hung up during a connected call by removing the
voltage between the tip and ring momentarily. This feature is useful for
auto-answer equipment, which then knows when to disengage. CPC is
configured in the Regional, Line, and PSTN Line tabs. See
Appendix A, “Advanced Voice Fields.”
Syslog and Debug
Server Records Syslog and Debug Sever Records log more details than Report
Generation and Event Logging. Using the configuration parameters,
the WRP500 allows you to select which type of activity/events should
be logged. Syslog and Debug Server allow the information captured to
be sent to a Syslog Server. Syslog and Debug Server Records are
configured in the System, Line, and PSTN Line tabs. See
Appendix A,
“Advanced Voice Fields.”
SIP Over TLS The WRP500 allows the use of SIP over Transport Layer Security
(TLS). SIP over TLS is designed to eliminate the possibility of
malicious activity by encrypting the SIP messages of the service
provider and the end user. SIP over TLS relies on the widely-deployed
and standardized TLS protocol. SIP Over TLS encrypts only the
signaling messages and not the media. A separate secure protocol such
as Secure Real-Time Transport Protocol (SRTP) can be used to encrypt
voice packets. SIP over TLS is configured in the SIP Transport
parameter configured in the Line tab(s). See
Appendix A, “Advanced
Voice Fields.”
Feature Description