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Unread 04-04-2009, 06:16 PM   #4
boatman
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The following settings should provide the correct NAT settings:

(under SIP tab)
Handle_VIA_received: yes
Handle_VIA_rport: yes
Insert_VIA_received: yes
Insert_VIA_rport: yes
Substitute_VIA_Addr: yes
Send_Resp_To_Src_Port: yes
STUN_Enable: yes
STUN_Test_Enable: yes
STUN_Server: stun.voxalot.com:3478 (or any STUN server such as 'stun01.sipphone.com:3478' or 'stun.sipgate.net:10000')
NAT_Keep_Alive_Intvl: 179

(The NAT_Keep_Alive_Intvl value 179 was chosen to be less than the NAT session expiry time of most routers.)

(under Line_1 and Line_2 (or PSTN_Line) tabs)
NAT_Mapping_Enable: yes
NAT_Keep_Alive_Enable: yes
NAT_Keep_Alive_Msg: (this setting should be left blank)
NAT_Keep_Alive_Dest: $PROXY
Register Expires: 3600

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At this point do another audio test. If the problem persists make the changes given below.
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1. In the routers on each end; forward the SIP ports and the RTP port range.
2. Set STUN Enable: no
3. Set NAT_Keep_Alive_Enable: no
4. In order for the ATA/phone to know it's public IP address, make sure the ATA/phone is registered with at least one SIP registrar, or enter your public IP address into the appropriate setting in the ATA (in Linksys ATAs this setting is EXT_IP).

If the problem persists at this point try using the Symmetric NAT option setting in your Voxalot account.
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