**Also note that this is the result of several contradicting directions, found here, here, and here.
These are the options I changed. If something isn't mentioned, the default setting should work.
Router
-> WAN Setup Tab
[Same as initial configuration]-> LAN Setup Tab
Networking Service: Bridge-> Application Tab
Enable DHCP Server: no
Enable DMZ: no
Voice
->SIP tab
->SIP Parameters
SIP TCP Port Min: 5060
SP TCP Port Max: 5060
->RTP Parameters
RTP Port Min: 16384
RTP Port Max: 16390
-> NAT Support Parameters->PSTN Line tab
STUN Enable: yes
STUN Test Enable: yes
STUN Server: stun.ekiga.net
EXT IP: [IP address of the box]
-> Line Enable: yes
-> NAT Settings
NAT Mapping Enable: no
-> SIP Settings
SIP Port: 5061
EXT SIP Port: 5061
SIP Debug Option: full
-> Proxy and Registration
Proxy: [IP address of Asterisk Server]
Register: yes
Make Call Without Reg: no
Answer Call Without Reg: yes
-> Subscriber Information
Display Name: PSTN
User ID: pstn
Password: [whatever you specify]
-> Dial Plans
Dial Plan 1: S0<:1000@[IP addr of Asterisk Server]>
-> VoIP-To-PSTN Gateway Setup
VoIP-To-PSTN Gateway Enable: yes
Line 1 VoIP Caller DP: 1
VoIP Caller Default DP: 1
-> PSTN-To-VoIP Gateway Setup
PSTN-To-VoIP Gateway Enable: yes
PSTN Ring Thru Line 1: no
PSTN CID for VoIP CID: yes
PSTN Caller Default DP: 1
-> FXO Timer Values (sec)
VoIP Answer Delay: 0
PSTN Answer Delay: 2
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