ATL Telecom IP300S - Manual (Page 96)

IP SIP Phone v2 User’s Guide Mar. 2005
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Appendix B – Trouble shooting
1. 『 』 To verify your network, please go to Advanced , then type a domain name to ping its
reachability and/or aliveness, like “yahoo.com”, “iptel.org” or “fwd.pulver.com”. If the
response is “Host unreachable”:
i. Check your network link, make sure it works normally (RJ-45 jack plugs into the right
hole and the LAN indication LED should be on.
ii. 『 』 Check your IP, DNS, and gateway settings on Network setting page. Note: if you
『 』 do not statically assign domain name server by picking Static DNS Server , you
must enable e
『 』 『 』 ither DHCP or PPPoE ; otherwise, it will use the static DNS IP
『 』 『 』 configured into the Primary DNS Server and Alternate DNS server item.
iii. If you reside on LAN without gateway, you should specify “0.0.0.0” as your gateway IP
to disable gateway routing rather than assign a non-existent or an invalid IP; otherwise
the network packets may not be routed correctly (which may result in no voice packets
could be sent from this phone)! This constrain applies to DHCP and PPPoE as well:
DHCP and PPPoE server should not designate a non-existent or an invalid gateway.
iv. 『 Go to IP SIP Phone』 『 』 page to make sure that those Active Network Status
matches those configured. Specifically, if the active DNS is 0.0.0.0, then you may have
wrongly configured to 『Static DNS
』 Server without setting a valid DNS IP in either
『 』 『 』 Primary DNS server or Alternate DNS Server (see point-2 for detail).
2. To verify SIP settings:
i. 『 Go to SIP Settings』 page:
a.
『 』 『 』 Your Transport setting should include UDP. Verify the network settings to
see whether it works normally.
b. 『 Your SIP Listen P 』 ort setting must be less than 65536 and greater than 0. We
suggest a value greater than 5000 to avoid accidentally conflicting with system
service ports. System default is 5060.
ii. 『 』 『 Go to Network / RTP S 』 ettings :
a. Your 『 』 RTP Port Base should be an even number ranges between 2 and 65534.
b. 『 』 Your RTP Port Range setting should be an even number and larger than or
equal to 2. At least 4 ports should be configured for IP SIP Phone since it has a
maximum capacity of two concurrent calls and each call consumes two consecutive
UDP port pair (one for RTP and the other for RTCP). We suggest a range value of 6
c. 『 』 『 』 The sum of RTP Port Base and RTP Port Range must be less than 65536,
『 and must not overlap with SIP Settings』 『 / SIP List 』 en Port .
iii. 『 Go to N -th Domain』 and configure
a. Authentication
b. SIP Address-of-Record