Linksys SPA3102 and FWD.

[dibun]

I have bought 2 Linksys SPA3102 and configured them with FWD accounts with the help of Voxilla config tools. Both SPA3102 are registered and I can call each other with FWD account but I cannot hear any voice on any of them. I tried the FWD echo test by dialing 613 but I cannot hear anything.

I am attaching a screeshot of the Voice info page.

Any help will be appreciated.

[Gasman]

Have you forwarded ports on your router as I mention in this thread.?

I presume you are behind routers.?

[dibun]

Originally Posted by Gasman

Have you forwarded ports on your router as I mention in this thread.?

I presume you are behind routers.?

I have forwarded the RTP ports and I can sucessfuly carried out the FWD echo tests. The RTP ports on both boxes are different so there wont be any conflict. But still I cannot hear anything when I call from one box to another. Both boxes are behind the Speedtouch adsl router. I can see both the boxes are not receiving any RTP packets when I call each other but they receive rtp packets when I do a echo test.

So I am bit lost here now

[Gasman]

So I am bit lost here now

I would be as well, sorry. I have forwarded the ports 5060 and 5061 as well.

You could try the linksys forum as well.

[dibun]

After much testing now one unit works. It seems I cannot use two unit behind the same internet connection. Anyway I have found two different internet connection to test.

One more question how do I configure VOIP to PSTN? Can I use same FWD account or do I need to create a new one for it?

[boatman]

Both of your SPA3102 units should work behind the same router, however it may be prudent to move the SIP and RTP ports of one unit to avoid overlap with the other unit.

It's also easy to configure the ATAs so that no port forwarding is required. This is done by setting NAT Keep Alive Enable to yes, and set NAT Keep Alive Intvl to 60, and all the NAT Support Parameters to be as shown here:

<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>stun01.sipphone.com:3478

Depending on how long your router will remember the NAT mapping, you can increase NAT Keep Alive Intvl, up to 300 for some routers.

The settings that will work behind a NAT router without need for port forwarding are in this post HERE. Just change the Proxy, User ID, and Password to match your FWD account.

At the above link, you would select one of the Basic Settings Files labeled "For ATAs on a LAN behind a NAT router". If it doesn't load according to the instructions there, you can enter the settings by hand.

You will probably want to improve your dial plan. In order to call PSTN numbers you will need to buy outbound minutes from someplace, except for calling toll free numbers. The dial plan HERE will get you started, or at least give you some ideas about making your own dial plan.

[dibun]

Originally Posted by boatman

Both of your SPA3102 units should work behind the same router, however it may be prudent to move the SIP and RTP ports of one unit to avoid overlap with the other unit.

It's also easy to configure the ATAs so that no port forwarding is required. This is done by setting NAT Keep Alive Enable to yes, and set NAT Keep Alive Intvl to 60, and all the NAT Support Parameters to be as shown here:

<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>stun01.sipphone.com:3478

Depending on how long your router will remember the NAT mapping, you can increase NAT Keep Alive Intvl, up to 300 for some routers.

The settings that will work behind a NAT router without need for port forwarding are in this post HERE. Just change the Proxy, User ID, and Password to match your FWD account.

At the above link, you would select one of the Basic Settings Files labeled "For ATAs on a LAN behind a NAT router". If it doesn't load according to the instructions there, you can enter the settings by hand.

You will probably want to improve your dial plan. In order to call PSTN numbers you will need to buy outbound minutes from someplace, except for calling toll free numbers. The dial plan HERE will get you started, or at least give you some ideas about making your own dial plan.

Thx for the reply. Regarding VOIP to PSTN I was thinking in a different way and I will try to explain it below

SPA3102----------voip---->SPA3102----phone line---->local PSTN
Location A Location B Local phone line

Location B will have a local phone line connected to SPA3102. So can I from location A call the location B box first and from there make local phone calls from Location B to local PSTN?