linksys PAP2T one way audio

[Patrice]

Hi
I'm a registered member to FPL since a month. I buy my own ata PAP2T adapter on a website. The purpose of using FPL for me is to have a second phone line. I have a phone line also with Distributel with an SPA 2102 adapter, it also the main router of my LAN. Wheter if place or receive a call, the audio goes one way after arrond 5 to 10 minutes.

My lan is like this:
- Modem (distributel)
- Main Router/Ata distributel : SPA 2102 (192.168.1.1)
- Second router/ wireless acces point : Linksys WRT54GL with DDWRT (192.168.1.2)
- ATA adapter for freephoneline : PAP2T (192.168.1.200) wich is conncected in my WRT54GL

I did the configuration with this reference:
http://www.voipfan.net/ATAs/PAP2_freephoneline.html
I thought that the problem was the the port forwarding so I did configure it on the SPA 2102 (main router). The confiration for port fowarding is this : ( 5060,5061) and (13000,13001) and (6060,6061) to 192.168.1.200.
I also insert a STUN server stun.xten.com an I enabled it.

If i did a mistake with my configuration, i would please to have comments on it.

Sorry for my english since I'm a french canadian.

Regards

PAt

[bridonca]

I suspect your problem is that you have not forwarded any RTP ports to the ATA. It can be any unused port. Port 12000 is a good choice. If you can forward a range like 12000-12009 to the ATA, that should be good. Sometimes you only need just one RTP port, it depends on the ATA. Just make sure the ATA is programmed to accept the same ports you got your router to forward. If you prefer to use the ATA's default RTP ports, that works also. You can get the router to forward those instead of 12000. Just make sure nothing else is using the ports you have selected.

Try not to forward any SIP ports, because you have 2 ATA units. I would go as far as removing ( 5060,5061) and (13000,13001) and (6060,6061) from the forwarding list. The default SIP port is is 5060. The other SIP port choices are there if for some reason port 5060 is blocked by the ISP. It seems unlikely that you have that as a problem.

The reason I say you do not need to forward any SIP ports is because the Freephoneline SIP proxy should already be able to handle the routing of SIP. (useless for RTP, as you noticed) Completely disable the STUN settings. It is of not much use if you already have a SIP proxy.

One other thing. Set the registration period, or keep alive on your ATA to 30 seconds. You do that, and the ATA will tell your router it still has a use for that port. It might be redundant if you got the RTP settings correctly configured, but it won't hurt.

[Patrice]

Hi bridonca,

I made your configurations suggestions and it works. Thanks a lot, it's very appreciated