Faq; Appendix B - Planet VIP-281GS User Manual

H.323/sip voip gsm gateway
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Appendix B

FAQ

Q1: What is the default administrator password to login to the gateway?
A: By default, your default username is "admin"; default password is "123" to login to the
router. For security, you should modify the password to protect your gateway against hacker
attacks.
Note: Default guest login username/password: guest/guest
Q2: I forgot the administrator password. What should I do?
A: Press the Reset button on the rear panel for over 5 seconds to reset all settings to default
values.
Q3: What is the default IP address of the router?
A: The default WAN IP address is 172.16.0.1 with subnet mask 255.255.0.0.
Q4: What is different [set to default] and [Factory set to default]?
A: Factory set to default, you must push RST button until 5 second, and gateway will clear
all your setting, and let gateway Wan port become the factory default (172.16.0.1). When
you use setting to default by Web or telnet, it will clear all your setting, but the wan port
setting will be saved. If you remote the gateway, after set to default, you can login gateway
again. No reset the gateway wan port again.
Q5: Why can I call out when the gateway under the NAT?
A: VoIP product almost has NAT Pass through problem. By SIP, there are many NAT Pass
through Function can solve 80% NAT Problem. You can choose STUN/Outbound Proxy/
Symmetric RTP to Pass through NAT, you don't set any other setting (DMZ/Virtual Server)
by router side. If you use STUN/Outbound Proxy, you must have a STUN/Outbound Proxy
Server to support. If they can't pass NAT, please open the DMZ/Virtual Server by
Router/NAT/Firewall.
Q6: Why does the one way talk happen?
A: Generally, one way talk happen when use the different codec between VoIP devices
make call. Please check and setting the same codec, most one way talk will be solved.
Q7: Why can I call out by Gateway?
A: Please chick your Gateway is registered SIP Proxy Server (ITSP), and chink your
Internet works fine. Gateway can't make a call without Internet or SIP Account that from
ITSP supply. You must have a SIP account or know the other Gateway IP/Domain Name,
and then you can make a VoIP call.
Q8: Why I use asterisk by G.729 sometimes disconnect happen?
A: In asterisk setting VAD must disable, if you open Silence Compression (VAD), it will
make call disconnect happen, please disable the option when you use the asterisk.

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