Fongo Community Forums

Check this link
http://forum.fongo.com/viewtopic.php?f=8&t=7348
Most of this is in the Sipura (Linksys manual)
If that is not enough the run STUN with a suitable stun server.

It does seem to work OK from my Android tablet although it takes a little while after the offering of English or French for the keypad to appear and work. If you are just looking for shipping info then it is best to use the internet.

Well, I hope that your service is not really as bad as you are finding due to poor service from FPL. Normally, these problems tend to be due to incorrect configuration at the user end. As things were working OK, try this 1. Power down the PAP2T and then the router. Disconnect any other devices conne...

I don't own a PAP2 so I have only configure router/pap2 over the internet.. You could post your router an PAP2 configuration or email me the configuration.

Sent you a pm.

What's DTMP? Do you mean DTMF? I can dial numbers starting by *70 or *67 and noticed also that I can't reach an FPL number (438-338-xxxx). My settings were not changed since 02/2012. I can call my DV home number from my cell or from Skype and the call shows 0:00 minutes for the numbers I can't dial...

It seems that your phone is not sending the correct DTMf tones. Go to your GS286 and check what tones are being sent. Chen the line in the web set up that looks like in-audio via RTP (RFC2833) via SIP INFO currently mine is set to "in audio" but it you have that setting as well then try SI...

I suspect that you need to go back to the beginning and reconfigure your router and ATA as it should work OK. The PAP2 is quite fussy about its configuration so one option would be to try it with a basic SIP provider that offers a free account such a antisip.com, ekiga.net, getonsip.com or ideasip.c...

I also get beeping when I call a couple of numbers in particular(both PSTN) and I am not even using FPL. Calling other number is usually OK. I have no idea what causes it but since the connection is not broken it has not really bothered me.

It might be best if you call awan for help as I will not be around for a while but it does look as if you need to go back a properly configure your system. STUN should get the external address (that you need) and replace the 192.168.100.1 address that it is getting at present with it and things may ...

I've tried everything listed and it is still the same. One thing just came to mind. My internal IP on my new modem is 192.168.100.1, but my internal network address (on my router) is different. Do I need to change my internal home network numbering to match the modem? Internet and everything else s...

1. Give your PAP2 a static IP address (your need to set it up on your router and on the ATA}. 2. Forward port 5061 and 5062 UDP from the router to the PAP2, set line 1 of the PAP2 to port 5061 for the sip port and line 2 of the PAP2 to port 5062 for its sip port. Leave the default external port as 5...

This is not a site for Vonage problems .... however
If you are able, one thing to do is to try running a stun server. It appers that the ATA is losing its record of the server IP address.
Whether or not you can do this depends on the ATA (voip box) that you are using.

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 so...

As far as I know router DMZ only makes those ports not already in use available so perhaps you have something else using ports that is clashing with you ATA. Lots of people have the same problem (or similar) to what you have. For the 3102 you probably need to run STUN, NAT keep alive, NAT mapping en...

Unable to receive calls as of this morning- caller gets no ring, just a busy signal. No problems without outbound calls That seems like a configuration problem. For now try running STUN (eg. server stun.callwithus.com) as well as NAT Keep alive and NAT Keep Alive. Nope I disagree. Any Bell/Telus li...

Read on a different forum of someone having the same problem, a user recommended "NAT Mapping Enable" & "NAT Keep Alive Enable" to be turned on - tried it, still no progress. Not a call today, problem continues :( You need a STUN server as well for your problem. Run STUN wit...

Try running STUN as per the SPA2102 manual.

crusader9 wrote:Unable to receive calls as of this morning- caller gets no ring, just a busy signal.

No problems without outbound calls

That seems like a configuration problem. For now try running STUN (eg. server stun.callwithus.com) as well as NAT Keep alive and NAT Keep Alive.

This is probably happening because your Router/ATA is not full configured. The router IP may have changed but the ATA has not been updated so it does not see incoming messages and cannot send them out. Check your keep-alive interval (mine is set to 20 sec) and the Register Expiration time (mine is s...

I think Jake is correct. The only real way is do some tests ... send a FAX or two and try to receive some. Codec G711U should be OK if you do not have T38. and, yes, use the slowest Fax rate that you can.
I cannot remember the last time that I sent or received a FAX!

i have linksys wrtp54g http://homesupport.cisco.com/en-ca/supp ... rs/WRTP54G my computer : windows 7 So, what are you using to send the FAX. You need to used a high quality Codec and if you look in the manual for the WRT54G there should be some settings there that you need to modify. SIP is genera...

You need to give more information about how you are trying to send/receive fax (Computer, ATA.configuration/codec,,, phone service).

You need to configure each SPA2102 with a separate static IP and forward suitable ports to each for SIP. Have you done this?

Again, the Linksys ATA of then need you to run STUN with a server of your choice (eg. stun.callwithus.com), Nat Keepalive and Nat mapping enable. It may or may not cure the problem.

You probably need to properly configure your ATA and router. Forward the required ports etc. but to really help we need to know more about your set up (router, ATA, phone etc).