ZyXEL Communications P-2002 - Manual (Page 98)

P-2002 Series User’s Guide
98 Chapter 12 Troubleshooting
12.7 Problems with Telephone or Telephone Port
12.8 Problems with Voice Service
Table 39 Troubleshooting Telephone
PROBLEM CORRECTIVE ACTION
There is no dial
tone or I can’t make
or receive calls.
or
There is beeping
instead of the dial
tone.
Check the telephone connections and telephone wire.
Beeping means that there is not a SIP account registered for the phone to use.
You can check the Prestige’s IP addresses and VoIP status in the Maintenance
Status screen.
Make sure you have the VoIP screen properly configured. If you configured a SIP
account to receive calls on only one of the phone ports, make sure your phone is
connected to that port.
Make sure you have the Phone Port screen properly configured. If you
configured a phone port to only use one of the SIP accounts for outgoing calls,
make sure that SIP account is properly configured and active (see the VoIP and
Maintenance Status screens).
There is a beep
before the dial tone.
A single beep before the dial tone indicates that there is a voice message for SIP
account 1.
Two beeps before the dial tone indicate that there is a voice message for SIP
account 2.
Use your voice service provider’s instructions to check your voice messages.
Table 40 Troubleshooting Voice Service
PROBLEM CORRECTIVE ACTION
After the VoIP is
configured and
working, others are
unable to call you
or you lose your
connection during a
call. There is a NAT
router between the
Prestige and the
SIP server.
This could be caused by a short NAT UDP session timeout on the NAT router.
When the SIP session’s entry in the NAT table times out, the NAT router does not
have any record to use for forwarding VoIP traffic to the Prestige.
If possible, set the NAT router to use a longer NAT UDP session timeout.
Otherwise, try one of the following:
•
Shorten the registration expiration period (see the Expiration Duration field
in the VoIP Advanced screen) in order to cause the Prestige to re-register
with the SIP register server more frequently. Note that this will not help if the
SIP register server enforces a long registration expiration period (since the
Prestige will also use the period set by the SIP register server).
• Use STUN. If your VoIP service provider does not have a STUN server, you
can still enable STUN and enter the IP address and port number of the SIP
server in the STUN server fields. This causes the Prestige to send STUN
requests to the SIP server. While this will not make STUN work (since there
won’t be any responses to the STUN requests), it should keep the NAT UDP
session in the NAT router.