ATL Telecom IP300S - Manual (Page 99)

IP SIP Phone v2 User’s Guide Mar. 2005
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failure cause is due to the mutual exclusion of both parties’ CODEC capabilities. For example,
if you specify explicitly to use only G.723.1 for voice stream whereas the peer is only capable
of G.711, then the conversation cannot proceed.
To ensure the phone will gracefully fall back to G.711, either party of the call should not
disable G.711 CODEC. To check CODEC settings on IP SIP Phone, please go to
『
Advanced』 『 / CODEC』 『 page to check for Preference』 section. If you prefer to use
G.729 or G.723.1 for voice compression, consider lower the priority of G.711 rather than
disabling them.
10. I can call others, but some of them or all others can not ring me.
Possible cause:
i. You are in All Call Forward mode (You could hear stuttering dial tone while picking
up handset).
ii. Both of you are under on the same NAT, and either one of you 『 employs Enable
STUN』 or 『Static NAT IP/UDP Map』 to traverse NAT. This is largely because
some NATs will not loop packets back if the source and destination are on the same
NAT.
iii. You use 『Static NAT IP/UDP Map』 to traverse NAT but the assigned static NAT IP
『 』 on NAT & Firewall page does not match the real IP of your NAT. (This is
possible especially when your NAT does not come with a fixed IP but employ PPPoE
or DHCP to gain access to WAN instead). By default, IP SIP Phone will auto-detect
『 』 the NAT IP (if the STUN server configured into NAT & Firewall page is viable)
『 』 『 』 and notify you whenever you hit the Network / NAT & Firewall page if it
detects a mismatch between the static NAT IP you configured and the one it
auto-discovered by STUN protocol. Alternatively, you enable the option –『Auto
』 Update Static NAT IP by STUN to auto-refresh the newly changed NAT IP
whenever it detects an inconsistency.