Is there a way to test FPL before buying the unlock hey?

[jumbo]

Hi All,

I have the LinkSys PAP2-NT adapter. I would like to know if there is a way to use test SIP settings in order to make sure I will be able to use FPL over my internet connection before choosing to buy the unlock key.

Thank you very much.

[Jake]

They don't offer a test account, but if you want to check if your connection is OK, run the VOIP test on this page http://myspeed.visualware.com/servers/n ... 24%20Kbps) That should give you a result at the end of the test saying if you are good to go or not.

What sort of connection do you have?

[jumbo]

Thanks Jake for the reply.

I have a 1Mbps DSL connection, but I'm currently in Lebanon not Canada. I'm concerned my ISP might be blocking some ports.

When I run the connection test that comes with the FPL software, it say that "SIP traffic is filtered for your internet connection. If so, please consult your ISP to allow the SIP traffic to your computer". It also tells me to make sure that ports 6060 and 13000 are not blocked.
Btw I'm able to make calls through the software, and voice is OK in both ways. However the call gets dropped after about 40 seconds. I noticed many people are complaining of similar problems with the software.

Thanks for the link though. I'll run this test and get back with the result.

[jumbo]

Hello again Jake,

here's the result for the VoIP test using G.711 codec: http://myspeed.pathcom.com/myspeed/db/report?id=45312

However, I get this on the graph (see pic attached): SIP response times: Register (no response), Invite (no response), Bye (no response).
Are these relevant?

[Jake]

It looks like you might have a few problems then. Your connection is strong enough to use voip, but it could be that some ports are being blocked as you mentioned.

Try the test in this page http://myspeed.visualware.com/servers/n ... alware.com

[jumbo]

It looks like you might have a few problems then. Your connection is strong enough to use voip, but it could be that some ports are being blocked as you mentioned.

Try the test in this page http://myspeed.visualware.com/servers/n ... alware.com

Thanks Jake.
Connection is successful for this test (ports 20010-20014).
How can i test for ports 5060-5061, 6060 and 13000?

[Jake]

You can always change the PAP2T to work over those RTP ports. These are the ports that carry the audio.

If you want to check calls, setup a couple of accounts on https://www.getonsip.com/ and put one on your ATA and the other on something like X-lite http://www.counterpath.com/x-lite.html and call each other. It should give you a good idea if things are working or not.

[jumbo]

Ok, I must admit I'm dumb. I've been connecting the phone to Line2 instead of line1 !

Thanks Jake for the amazing help! I've been able to configure both the PAP2 and XLite with 2 distinct getonsip.com accounts. Calling the phone from XLite has been successful. Voice is working in both directions.

[Jake]

No, that SIP port should be 5060. Change it back, and make sure you have done the other settings like mine. I have tested them and they work fine.

Proxy: getonsip.com
Use Outbound Proxy: YES
Outbound Proxy: sip.onsip.com
Display Name: Whatever you like
User ID: Username (from your email)
Password: Sip Password (from email)
Use Auth ID: YES
Auth ID: getonsip_(your username)

When the ATA reboots you should get sent to the home page and there it should say that the line is registered.

[Jake]

Ok, I must admit I'm dumb. I've been connecting the phone to Line2 instead of line1 !

Thanks Jake for the amazing help! I've been able to configure both the PAP2 and XLite with 2 distinct getonsip.com accounts. Calling the phone from XLite has been successful. Voice is working in both directions.

You keep changing your posts, I can't keep up!

Anyway, there is a good chance that you will be OK. You never really know until you try it proper though.

[jumbo]

Ok, I must admit I'm dumb. I've been connecting the phone to Line2 instead of line1 !

Thanks Jake for the amazing help! I've been able to configure both the PAP2 and XLite with 2 distinct getonsip.com accounts. Calling the phone from XLite has been successful. Voice is working in both directions.

You keep changing your posts, I can't keep up!

Anyway, there is a good chance that you will be OK. You never really know until you try it proper though.

Yes, sorry I deleted the previous posts coz they make no sense anymore

I will go ahead and buy the unlock key. But what about RTP ports you told me about earlier? The ones that tested successfully, ports 20010-20014?

[Jake]

if those tested properly then the other probably will also.

If you are looking where to change it on your PAP2T then login as Admin -> Advanced view, then go to the SIP tab and you should see the min and max RTP port numbers. I would leave it as is unless you get a problem.

[jumbo]

Here's the good news: it's working!

Thanks again Jake, I appreciate it. I hope the instructions you gave in this post will serve as a test guideline for others.

Regards.

[Jake]

Thanks for letting us know jumbo. Glad it worked out for you.