Your Phone/ATA Not working, check this first

[awan]

Hi

I am a FPL with SIP key user like any one of you.

I was having these problems , phone one way, not registering.

Here is the solution I am posting after my research as there seems to be no support for the purchased SIP keys.

Solution 1.

Place your ATA under DMZ in the router, reboot the router and ATA. should work fine

Solution 2 ( more safer)

Do the Port Range Triggering in your router for the following ports. this option is under Applications and Gaming on my linksys router.

Triggered Range Forwarded Range enabled
13000 to 13001 13000 to 13001 yes
5060 to 6061 5060 to 6061 yes
Save
Reboot
Reboot the ATA

my Audio settings
preferred codec is G711U
Second prefered codec G729a ( i have tried flipping and seems not to make any diff to me.)
Use pref Codec Only: no

I hope this will help many,

Can moderators make this sticky at the top of Support

[awan]

Please note that I am not using STUN as some folks have mentioned to use. I have tested and my ATA is working with the port range triggering .

I think it is the 13000-13001 port range that the router blocks.

[Truexile]

IMO 5060 and 5061 are your main concern. I have heard before to forward ports 10000 to 20000 which I thought was a bit ridiculous. If you log onto the SIP settings on your ATA device and forward your RTP Port Min (mine is 16384) to your RTP Port Max (mine is 16482) seems like a better solution because thats the ports the device is using. Anyways thats my 2 cents.

[dibsmft]

The part that makes setting up SIP voip is usually the router. Every router that I have used has been different in exactly what it does and often how it is done. Sometimes some of the main features do not even work (eg. SIP helpers). I have not had a router so far that I could not get to work but some can be very uncooperative! Usually I do not forward port 5060 anywhere (a few devices will insist on it so it is best to leave it alone). So I am using ports 5061 - 5068 UDP (SIP messaging a different port for each device). For SIP RTP packets I have usually used port 8000 UDP up with a few ports for each device. I only have one provider at present where I have to use STUN and If I don't I get the results that many members are having with incoming calls (no sound in or out).

[FONGO_steve]

I'm going to advise against option #1 as putting the ATA in the DMZ exposes the entire device, including the web GUI access ports, to external access. This would make it quite easy for a device to become compromised and thus a hacker could gain access to one's device or account.