Ports 5060, 5061 are blocked!

[zo3bi]

Hey,
I'm going to Dubai on chrismas break and I want to take my freephoneline number with me. I purchased the ata device + config ( the one for 99.99) and I want to plug it in when I get to Dubai . The problem is my friend told me that ports 5060 and 5061 are blocked.... So I was wondering if there was any other ports that I could use to bypass the blocked connection. I don't want to use sippp.com but their idea is great, does freephoneline have something similiar?
Btw: if i have to I am willing to pay....

Thanks ,
Moe

[bridonca]

That is one of the nice things about freephoneline, the service has ample ports to choose from. Try one of the sip ports from 5063 to 6060 and see what happens.

[laurent]

The way SIP works is that when your ATA/phone registers, it tells the server on which port to reach back. (Actually, it's a pair of ports, on even and one odd.)

In the configuration of your ATA, just try any port in the 5000 or 6000 range, but be sure to use an EVEN number, ending in 0-2-4-6 or 8. Then make sure you forward that power in your router, AND the next one. If you pick 5086 for example, also forward 5087.

[zo3bi]

Thanks for the replies will try it out in a week and keep you updated

[cu2o2o2]

The way SIP works is that when your ATA/phone registers, it tells the server on which port to reach back. (Actually, it's a pair of ports, on even and one odd.)

In the configuration of your ATA, just try any port in the 5000 or 6000 range, but be sure to use an EVEN number, ending in 0-2-4-6 or 8. Then make sure you forward that power in your router, AND the next one. If you pick 5086 for example, also forward 5087.

@laurent

You need only one sip port open/forwarded for each phone line. However, you will need to open/forward at least a PAIR of RTP ports (one odd, one even) for each sip port, if you want two-way communication.

[laurent]

Yes and no. SIP needs the two ports *usually* in order to receive encrypted or non-encrypted signalling.

[cu2o2o2]

For FPL purposes, we need only one SIP port for each number, AFAIK.

If SIP *usually* needs two ports to operate, then why does my FPL phone work ALL the time (not just *usually*), with just one SIP port open (with required RTP ports open)?

[laurent]

By "usually", I mean all the SIP providers in the world. Perhaps FPL doesn't need the encrypted port right now. Maybe it will in the future, or maybe it will never use it. I certainly don't know that.

One thing that I do know, is that SIP is an established standard that defines itself as requiring two ports for proper use, and I will setup those two ports on all devices that I come across because I want to make sure it works now, and in the future too. Should FPL one day decide to turn on encryption of SIP data, for example, my devices will be ready, because I followed the recommended standard settings, not just "hey this works, so I'll stop here".

I generally prefer to go with the safe approach, and make sure I follow the guidelines and established standards

Additionally, FPL themselves state that both ports are required to be open and forwarded, so as long as they claim it's needed, I'll stick to that.

[cu2o2o2]

Okay, getting future-proof is not bad!

For this one, I'll cross the bridge when I get there!


I missed the part where FPL states that both ports are required to be open and forwarded. Please be kind to point me to that statement, thank you.

Additionally, FPL themselves state that both ports are required to be open and forwarded, so as long as they claim it's needed, I'll stick to that.

[Jake]

Okay, getting future-proof is not bad!

For this one, I'll cross the bridge when I get there!


I missed the part where FPL states that both ports are required to be open and forwarded. Please be kind to point me to that statement, thank you.

Additionally, FPL themselves state that both ports are required to be open and forwarded, so as long as they claim it's needed, I'll stick to that.

They mention the ports in the FAQs for the softphone. I would assume the same applies for the ATA. http://www.freephoneline.ca/faq#30

I opened up and forwarded these ports on both my and my friends systems and neither has had problems. I don't see it as bad to open them, or is it if I don't really need them?

[zo3bi]

Dosn't work
All ports suggested are blocked!
Worst of all, I can't even access the device to change the settings, when i press *** it gives me an error!

UAE = NO VOIP, another factor that makes UAE a bad place to live in!

Any suggestions? I was reccommed to use vpn4sip.com , what do you think?

[laurent]

Yes you'll need to ues a VPN service to tunnel all these ports to a host outside UAE which will then make the final link to FreePhoneLine. Be advised though that there's two issues to take care of:
1. It's possible that VPN might not be allowed either. If the country wishes to monitor all traffic, they might block all encrypted traffic, which a VPN is.
2. Using a proxy or VPN introduces additional delay in the transmission time, which might make the VoIP call quality drop. You'd have to test to see how much of an effect it has, because that varies.

[curriegrad2004]

You can take my approach is that what I do before leaving the country is I make sure I have all my PPTP VPNs and OpenVPN VPNs functioning properly before I leave.

If anything like this happens, I can easily bypass them all by connecting back to my own VPN back at home. Quality wise, if the VPN is properly configured on both ends with QoS, then the voice quality shouldn't drop that badly. But if the latency is around the neighborhood of 300ms or so, then don't expect the call to complete properly

[abuzuhair]

does using a tunnel makes the calling limired to 35 sec Exactly... i am using a tunnel as i have 2 devices during my transition period. i am thinking of stopping my vonage for Freephoneline connection, but the only problem is any call i am making is limited to 35 Seconds excatly. if anyone call me, it goes unlimited

around 1 second delay is notice on Freephoneline while vonage has unnoticable delay... without any additional configuration change.

i tried it on 2 devices, 1st is my SPA3102 and 2nd Nokia SIP phone and they have the same case