Forwarding to FPL from another SIP

[yyzyyz]

What's the SIP uri to forward calls to FPL from another SIP account?
1xxxxxxxxxx@voip.freephoneline.ca doesn't seem to work. Is this even supported?

Thanks!

[FONGO_kris]

The system itself supports it but our service currently does not, no. We are very, VERY close to getting this bothersome online payment processor online, all we're doing now is working on that and minor bugs, then doing a revamp of the website (and hopefully software phone to C++!!).

[yyzyyz]

Does that mean that once you get around to activating this service, it would be possible for the other SIP users to make purely SIP-to-SIP calls to FPL phones by using URI 1xxxxxxxxxx@voip.freephoneline.ca (that is, without getting PSTN involved at all)?

[FONGO_kris]

What are you using to dial out using a SIP URI with?

[yyzyyz]

Gizmo5. It lets you add other SIP users as contacts and make free SIP-to-SIP calls.
For example, I can call my CallCentric # by adding 1777xxxxxxx@in.callcentric.com as a SIP contact.

It's even possible to dial out SIP URIs straight from the ATA device by using services like SipSorcery to create a custom dial plan and assigning numeric codes to the SIP URIs.

The obvious benefits are free long-distance/international calls to FPL phones.

[FONGO_kris]

Could you try dialing a freephoneline SIP URI? 1xxxxxxxxxx@voip.freephoneline.ca and PM me your error logs if any?

[yyzyyz]

Okay, so here's what happens when I call SIP URI 1xxxxxxxxxx@voip.freephoneline.ca:

The SIP call initiation request does get across to voip.freephoneline.ca, but the problem is that Sippy wants these incoming SIP INVITE messages authenticated! Instead of passing these requests over to the registered user agent for this account, it sends back 401 Unauthorized to the caller!

This confuses the client, which still tries to authenticate the INVITE message based on the client credentials on the caller side. Sippy gets this new INVITE message with auth parameters and obviously cannot identify the user (since no such username is registered with FPL), so it responds with "This account number is not valid" streamed over an audio channel and the call never terminates...

That's not the way it works - the INVITE messages originate from a different domain and should never have to be authorized.

[FONGO_kris]

Hmmm, could you PM me your number and the number you called from, as well as the time so I can pull up your SIP Logs?

I think I may know how to fix this but I'll have to look at your logs first.

[yyzyyz]

Sent you a PM with the details...

[yyzyyz]

Did the information I provided help nail down this problem at all?
If you need me to make more sip-to-sip test calls, please let me know.

Thanks!

[notchy]

I'm curious if this has been resolved yet?
Please post the reply so we all can see the corect setting

[FONGO_kris]

As I previously thought, this is not a "bug" it is just currently something we do not allow at this time.

For more info please veiw: http://forum.freephoneline.ca/viewtopic ... 968&p=4853

Thanks!

[yyzyyz]

I think there are two things worth considering here:

Incoming SIP calls would really be a very small percentage of the total call volumes. The reason being, it's not that straight forward for an average user to use this feature. To do this, you'd either have to create complex dial plans for use with your ATA device or use a software client that supports dialling SIP URIs (for example, eyeBeam).

Secondly, it's not a fair assumption that taking away this ability from these SIP callers would force them to call the FPL PSTN number instead. They may alternatively call on a non-FPL number or not call at all and use other means to communicate (especially, the international callers).

In my opinion, the revenue loss on account of missed PSTN terminations is pretty insignificant in this case and easily outweighed by the gain in popularity if this is enabled.

Cheers!

[FONGO_steve]

We are taking those considerations into mind - I know this idea will still be discussed in further detail after we get a few other things cleared up this year.

One thing to consider here:

Incoming SIP calls would really be a very small percentage of the total call volumes. The reason being, it's not that straight forward for an average user to use this feature. To do this, you'd either have to create complex dial plans for use with your ATA device or use a software client that supports dialling SIP URIs (for example, eyeBeam).

If user A receives an incoming SIP call, we loose the termination fee on our DID. If he calls his mom, we have to pay a termination fee.
If user A receives an incoming call on his DID, we earn the termination. If he calls his mom, we pay the termination. <---- buddy's calling is better balanced and paid for this way. I know it's a small fraction of users that would do this, but we also must consider the whole picture in consideration for why we are a free service!

[Jake]

I think I tried this before, but can't remember the specifics, but if you have a static IP you should be able to do something like this.

dial something@yourip:5060 (eg me@123.123.123.123:5060) with your sip phone, it should go to your ATA.

The other way would be to set up an account that does allow sip dialling and set up an account on sipsorcery.com to deal with the multiple accounts, or use the second line on your ATA.

There is work arounds, it is just not something that FPL offers it would seem. I can understand their point of view, but I don't think it would make all that much difference. I am having troubles making my mum understand I don't actually have a land line any more, let alone how to dial a SIP number.