HELP !!! can't receive calls using Linksys PAP2T

rikili

Hello All,

I am a n00b to VoIP, so this might be an easy one to answer.
I have a free account with freespeech.ie, and another one with FreeCall.com.
I also have a DID number, which I transfer to either of those accounts while testing.

Now, when I setup X-Lite softphone, I can make calls, receive calls, and receive calls from my DID number. no issues at all.

When I use my PAP2T, I can only make calls!!! I can't receive any calls. I tried each account (FreeSpeech and FreeCall), but always the same! The device shows that's it's registered. and making calls to any landline (Ireland or International) works absolutely fine. the issue is receiving calls. I tried every possible combination, forwarding all the ports from the router to the ATA device, enabling NAT settings on the ATA. nothing worked. When I ring my DID number, it rings on the caller side, but I don't receive the call. The call doesn't even get forwarded to the vmail after the elapsed time!!! I guess this is not an issue with my router, as the X-Lite softphone works 100%.

Am I missing something in my config? I am still scratching my head for the second week, and still no joy.

By the way, I have NTL broadband (10 Meg + phone line) could that be upsetting things?

Any help is very much appreciated.
Thanks in advance.

Johnboy1951

Kill the softphone.
Reboot both the PC and the Linksys and see if it makes any difference.
What OS are you using?

rikili

thanks Johnboy,

Of course I killed the softphone before plugging the ATA and vice versa, I'm not that thick :-)

I don't see what the OS has to do with the ATA!!! it's the ATA that's not working not the softphone! The softfphone is working 100% making and receiving calls.

it's the ATA that's unable to receive calls, eventhough it's making calls without a problem.

malachy47

rikili wrote: »

I also have a DID number, which I transfer to either of those accounts while testing.

I don't know what this means Where did this DID come from, who issued it?

Secondly, would you indulge us and get a free account at Blueface? It would truly assist as their service is a known quantity and it would help eliminate many variables in regards to your setup.

I'm still not sure what's meant by transferring a DID number however.

rikili

Hello All,

I figured it all out, and now it's working like a charm....

First to answer Malachy47, a DID number is a PSTN number, that's forwarded to you SIP account. I got this one for free for testing purpuses from TollFreeForwarding.com. They give you a free Dublin number for 10 days so you can test their services.

Now I don't want a free Blueface account, because that defeats the purpose for me! Freecall offers me free calls to all European landlines, and very cheap mobile and international calls. Also, I could get a DID Dublin number for €3.50 a month only...

Now the issue was with 3 main things (A skype phone, the router, the phone itself!):

1/ I have a Skype phone (PC-Less phone), and that was causing troubles. So as soon as I disconnected it I was able to see, on the info page, the last caller ID, but still the phone didn't ring, and I couldn't get the connection going.

2/ from there, I knew that the router wasn't allowing connection to the ATA device, so first thing to do is put the ATA in a DMZ.

3/ From now, I was seeing on the info page, that the call was coming, and the status was "Ringing"... but still the phone doesn't ring...

4/ I changed the phone (a cheap Argos one) with a DECT phone... and lo and behold... it rings....

5/ When I thought every thing was honkey dory, it turned out that there was more issues!!!! When I pick up the phone, I don't hear anything, and the calling phone still getting the ring signal, and on the info page the call status was "Invalid".

6/ few Google searches later, and it turned out that the ATA couldn't establish the call because it's behind a NAT router, even though it's on a DMZ.

7/ To sort that out.... Set up a STUN server as follow (The STUN server helps the adapter figure out its port number and external ip address.):

On the Line tab:
NAT Mapping Enable: yes
NAT Keep Alive Enable: yes

On the Sip tab:
Substitute VIA Addr:Yes
Send Resp To Src Port:Yes
STUN Enable: Yes
Stun Server: stun.freecall.com

et Voila... every thing is spot on!

Hope this helps anyone having the same issues as me...
N.B. (I have broadband with UPC, so the router is a CISCO EPC2425).

malachy47

Good grief Rikili, thanks quite a setup and I'm happy it's working out for you.

I notice freecall.com is one of those Betamax outfits, which, if they remain true to form, sooner or later will disappoint you, like many of the subscribers to this forum will attest. But if they have mended their ways do please come back and tell us

I do know what a DID number is, and as you have demonstrated by your solution, played no part in the problem you were trying to solve. That's why I questioned why it was part of your description of the symptoms.

Finally I'd be curious to know what your expected call rates will be in your setup, after all freecall.com are not free :-)

stephendevlin

Im having an issue with my Router (netopia 2247-62) I also have a pap2t connected and cant seem to receive calls.. I ring from my mobile to my sip number and it (obviously) just rings out the phone doesnt even flinch. I also use x-lite on the pc and it also doesnt flinch so Im guessing its the firewall in the router (which I have set at low and is behind NAT) I have opened the UDP ports from 5060 -5070 but to no avail... I did turn the firewall off and I managed to get through to the x-lite (only once and then it doesnt work after that first time) but not passed the ata to the phone. I have tried 2 different phones and neither of them work. not sure what to do at this stage as Im not sure how a "stun" server works. I can make calls out and the quality is perfect but the calls coming in .. jus aint workin

sk8gr8

on the pap2 line's configuration pages, enable 'nat mapping' and nat keep alive.
Also , you can set a stun server ( any stun server address)
Nat activation has the same effect as stun server, so don't use both, is useless..

stephendevlin

sk8gr8 wrote: »

on the pap2 line's configuration pages, enable 'nat mapping' and nat keep alive.
Also , you can set a stun server ( any stun server address)
Nat activation has the same effect as stun server, so don't use both, is useless..

So you mean turn NAT off? and just use the stun server? Can you be specific as this is driving me mental

sk8gr8

solution 1
enable nat mapping and nat keep alive
enable nat on the router
no need for stun server

or

solution 2
enable stun server
no meed for nat mapping

also coud be a port issue: if you have both pap2 and xlite on port 5060, than the system could be confuse.( could try to put port 5062, 5063 on the pap2)

stephendevlin

sk8gr8 wrote: »

solution 1
enable nat mapping and nat keep alive
enable nat on the router
no need for stun server

or

solution 2
enable stun server
no meed for nat mapping

also coud be a port issue: if you have both pap2 and xlite on port 5060, than the system could be confuse.( could try to put port 5062, 5063 on the pap2)

I need nat on the router for PSN. I might try a few things tonight. My phone also has Fring ... I may delete it as I never use it. Could be a conflicting error

stephendevlin

Bloody figured that problem out!!!!!! it was the ATA settings... there wasnt enough power getting to the phone from the ATA there was a voltage setting on it. once changed to 90 it worked perfectly from there on in!

Incase anyone else having this issue!

fonadain

It seems like you are having issues receiving calls to your Linksys PAP2T ATA device. This can be frustrating but there are a few things we can try to resolve it.

First, make sure your Analog Telephone Adapter (ATA) is connected and configured properly per the installation guide. Validate SIP settings match what your VoIP provider gave you. Also check if firmware needs any updates.

If the configuration checks out, the problem could be that you don't have Direct Inward Dialing (DID) enabled for the connected phone number. DIDs allow external callers to reach you directly without going through a receptionist.

Confirm if your VoIP provider assigned you a DID number. Log into your account portal and enable the DID feature if required. You typically need this for direct incoming calling. There could be a small monthly fee per DID line activated.

Additionally, some firewalls block internet calls. You may need to open UDP ports 5060 and 5061 on your router for connectivity. Feel free to reach out and I'd be glad to troubleshoot further if issues persist! Let me know how it goes or where exactly you are stuck.