ZyXEL P-2802HW-I1 User Guide - Page 301

Logs, P-2802HWL-I Series User's Guide, RTP Logs, FSM Logs: Caller Side, FSM Logs: Callee

Page 301 highlights

Chapter 22 Logs Table 133 SIP Logs (continued) LOG MESSAGE DESCRIPTION SIP UnRegistration Success by SIP:SIP Phone Number The listed SIP account's registration was deleted from the SIP register server. SIP UnRegistration Fail by SIP:SIP Phone Number An attempt to delete the listed SIP account's registration from the SIP register server failed. Table 134 RTP Logs LOG MESSAGE Error, RTP init fail Error, Call fail: RTP connect fail Error, RTP connection cannot close DESCRIPTION The initialization of an RTP session failed. A VoIP phone call failed because the RTP session could not be established. The termination of an RTP session failed. Table 135 FSM Logs: Caller Side LOG MESSAGE DESCRIPTION VoIP Call Start Ph[Phone Port Number] Outgoing Call Number Someone used a phone connected to the listed phone port to make a VoIP call to the listed destination. VoIP Call End Phone[Phone A VoIP phone call made from a phone connected to the listed Port] phone port has terminated. Table 136 FSM Logs: Callee Side LOG MESSAGE DESCRIPTION VoIP Call Start from SIP[SIP Port Number] A VoIP phone call came to the ZyXEL Device from the listed SIP number. VoIP Call Established Ph[Phone Port]

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Chapter 22 Logs
P-2802H(W)(L)-I Series User’s Guide
301
SIP UnRegistration
Success by SIP:SIP Phone
Number
The listed SIP account’s registration was deleted from the SIP
register server.
SIP UnRegistration Fail
by SIP:SIP Phone Number
An attempt to delete the listed SIP account’s registration from the
SIP register server failed.
Table 134
RTP Logs
LOG MESSAGE
DESCRIPTION
Error, RTP init fail
The initialization of an RTP session failed.
Error, Call fail: RTP
connect fail
A VoIP phone call failed because the RTP session could not be
established.
Error, RTP connection
cannot close
The termination of an RTP session failed.
Table 135
FSM Logs: Caller Side
LOG MESSAGE
DESCRIPTION
VoIP Call Start Ph[Phone
Port Number] <- Outgoing
Call Number
Someone used a phone connected to the listed phone port to
initiate a VoIP call to
the listed destination.
VoIP Call Established
Ph[Phone Port] ->
Outgoing Call Number
Someone used a phone connected to the listed phone port to
make a VoIP call to the listed destination.
VoIP Call End Phone[Phone
Port]
A VoIP phone call made from a phone connected to the listed
phone port has terminated.
Table 136
FSM Logs: Callee Side
LOG MESSAGE
DESCRIPTION
VoIP Call Start from
SIP[SIP Port Number]
A VoIP phone call came to the ZyXEL Device from the listed SIP
number.
VoIP Call Established
Ph[Phone Port] <-
Outgoing Call Number
A VoIP phone call was set up from the listed SIP number to the
ZyXEL Device.
VoIP Call End Phone[Phone
Port]
A VoIP phone call that came into the ZyXEL Device has
terminated.
Table 137
PSTN Logs
LOG MESSAGE
DESCRIPTION
PSTN Call Start
A PSTN call has been initiated.
PSTN Call End
A PSTN call has terminated.
PSTN Call Established
A PSTN call has been set up.
Table 133
SIP Logs (continued)
LOG MESSAGE
DESCRIPTION