Quote:
Originally Posted by gotalkvoip
So wondering about other settings like,..
Use DNS SRV:yes
DNS SRV Auto Prefix: yes
Proxy: au.voxalot.com
Outbound Proxy: au.voxalot.com
Register Expires:240
Proxy Fallback Intvl: 250
Register: YES
Make Call Without Reg: YES
Ans Call Without Reg: NO
Use Outbound Proxy: GUESSING 'YES'
Use OB Proxy In Dialog: GUESSING 'YES'
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Some of these settings are not very important, however I would suggest a few changes to what you listed above.
"Register Expires:" would typically be 3600 seconds, but you might want it at 600 seconds in order to respond more quickly to a failed proxy.
"Proxy Fallback Intvl:" 3600
"Ans Call Without Reg:" yes
If your Linksys/Sipura SIP device is behind one or more NAT routers, set the following:
(under SIP tab)
SIP_T1: 1
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 (if your phone does not always ring on incoming calls try 119 or 59, use highest number that works)
(under Line_1 and Line_2 (or PSTN_Line) tabs)
NAT_Mapping_Enable: yes
NAT_Keep_Alive_Enable: yes
NAT_Keep_Alive_Msg: 0000
NAT_Keep_Alive_Dest: $PROXY
Register Expires: 3600
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Optional settings, steps 1 - 4:
Sometimes its preferable to operate the ATA independently of a STUN server. The benefit is that phone service can continue no matter if the STUN server is working. See steps 1 to 4 below.
1. Forward the SIP ports and the RTP port range from the router to the ATA.
2. Set "STUN Enable:" no
3. Set "NAT Keep Alive Enable:" no
4. The ATA must learn it's public IP address, either through normal SIP registration, or place your public IP address in the 'EXT IP:' field).