SIP URI?

[lysack]

Hello,

I'm wondering if FPL supports SIP URIs? I was reading some 3rd party forums and they suggested this URI format (1 + FPL number plus the domain):

1604NNNNNNN@voip.freephoneline.ca

I have purchased the SIP config file and use my account on my Linksys IP phone. When I try to call the above SIP URI from another VOIP provider, I get a busy signal.

Your thoughts on this?

Thanks!

[Jake]

They don't allow it. If you really need to do this there is a work around, but you need to use a 3rd party.

Sorry.

[dibsmft]

This is definitely not for beginners with voip/sip!

This method can be useful if you have only one sip/voip line at it is dedicated to a provider that does not allow calling directly to the SIP address(URI).

If you connect through an ATA box or use a sip client then you may be able to call the ATA directly to its IP address. Unless you have a static external IP address the best way to start is by getting a free account with dyndns.org which is a free service that stores your current IP address and allows you to use an address like user.dyndns.org. Once that is done you can call the ATA box by calling mybox@user.dyndns.org:portnumber where mybox is your login name used on the ATA and the portnumber is the port that is forwarded to the ATA. To call my GS286 I would call something like 5555555@myname.dyndns.org:5063 (or to a line registered to FPL 1604NNNNNNN@yourname.dyndns.org:portnumber) and if everything is correctly configured it will ring and thats it. This I know to work with the GC286, SPA3102 and the Yealink T22P IP phone.

[evilmonkey]

Please please please be careful doing what dibsmft suggests. You are opening your phone line to the entire world, meaning anyone with a SIP connection anywhere in the world can call you for free. I tried to open up my phone to SIP calls once, and ended up getting weird calls at 3:00AM, with no one on the other line. It was probably some hacker machine looking for a way to dial into a box and try to make long distance phone calls or something. If you're running a PBX, you should be even more careful, although I assume if you're running a PBX you should have some knowledge of what you're doing.

In short, there's a good reason why FPL doesn't allow SIP URI calls (well, that and SIP URI calls make them no money while wasting their bandwidth).

[dibsmft]

Well, I did say
"This is definitely not for beginners with voip/sip!"
Incoming:
If you have means of knowing a persons voip IP then you can indeed call them if you also know their username("phone number") and port but this is generally true of voip and more or less true of how Fongo sends call to you.
Outgoing:
Several ATA devices will allow calling a SIP address "over" the logged in account and some (eg. SPA3102) give good support to it. This can be very useful if you only have one voip line.

I would not expect profit making voip providers to support this type of calling and if you have a second voip line then you would use a "free" voip provider to make and receive SIP URI calls.

[Bloodsong]

There are some free and robust ways to deal with this that do not expose your ATA to the net.
For a while I used an online SIP-Switch (SipSorcery), it would connected to freephoneline for me, then people could POTS dial my FPL OR they could SIP URI me at bloodsong@sipsorcery.com

But I don't use that set-up anymore... I just haven't had much need for it, and of course, now I'm versed in running IP-PBXs of my own, and so will do that when I need the functionality.

P.S.

Using something like SIP Sorcery allows for multiple SIP providers/DIDs to be connected via your one account, resulting in multiple functioning SIP URIs and DIDs that ring your device which is registered with SIP Sorcery.

(I used an IP-Kall number for american telephony, FPL for Canadian, and ofcourse the SIP Sorcery URI.)
Now that Asterisk has support for GVoice as a trunk however, I can use that for American calls and FPL for Canadian weee.

[dibsmft]

With the Sipura/Linksys ATAs you can (depending on the model) make calls to outgoing SIP URIs by putting them in the in the dialplan or use a provider in in one of the internal Gateways. Incoming calls can be accepted as I described above or by using the phones second line if it has one. If you need to make a call to another person who uses SIP/voip you can often use a local Sipbroker number to make the call or a local iNUM number to call the other persons iNUM world number if they have one. As far as I can tell, most Fongo users just want a good working phone and to save a little money so in those circumstances it is probably best to keep things as simple as possible.