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Gigaset C530 IP / LUG - USA en / A31008-XXXXX-XXXX-X-XX / appendix.fm / 25.03.2015
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Information on operating Gigaset VoIP telephones with routers
with Network Address Translation (NAT)
In general no special telephone or router configuration is required when operating a Gigaset
VoIP phone with NAT router. The configuration settings described in this section are only
necessary if you encounter one of the following problems.
Typical problems caused by NAT
u
No incoming calls are possible via VoIP. Calls to your VoIP phone number are not put through.
u
Outgoing calls via VoIP are not connected.
u
A connection is established with the other participant, but you cannot hear them and/or
they cannot hear you.
Possible solution
1
Change the port numbers of the communication ports (SIP and RTP ports) on your phone
(
£
"1. Changing the port numbers for SIP and RTP on your VoIP phone").
2
In some cases, you must also define port forwarding for the phone's communication ports
on the router (
£
"2. Setting port forwarding on the router").
1. Changing the port numbers for SIP and RTP on your VoIP phone
On your VoIP phone
, define different (local) port numbers for the SIP and RTP ports (between
1024 and 49152).
u
These numbers must not be used by any other application or host in the LAN and
u
Must be considerably higher or lower than the SIP and RTP port numbers that you usually use
(and are preset on the phone).
This procedure is particularly useful if additional VoIP phones are connected to the router.
To change the SIP and RTP port numbers on your VoIP phone, proceed as follows:
¤
Connect your PC's browser to the Web configurator of the phone and log in.
¤
Open the web page
Settings
¤
Advanced VoIP Settings
and change the
settings for the SIP and RTP ports (
To help you remember the new port numbers (e.g. for router configuration), you can choose
port numbers that are very similar to the standard settings, e.g.:
¤
Save the changes on your phone.
¤
Wait for the active VoIP connections to be re-registered. To do so, switch to the web page
Settings
¤
Telephony
¤
Connections
to see the
Status
of your VoIP connections.
¤
Check to see whether the problem persists. If it does, perform step 2.
SIP port
49060
instead of 5060
RTP port
49004 to 49010
instead of 5004 to 5010
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