SIP URI possible with FPL?

[voip]

Can I call out to a SIP URI with FPL?

Does FPL support incoming SIP URI type calls? If so, what is the addressing format?

[FONGO_kris]

Yes of course you can, and yes it does.

The current SIP format is - 1519xxxxxxx@voip.freephoneline.ca

Make sure all the settings are still set to what they are in the Config file, and just so you're aware, that 1519xxxxxxx@voip.freephoneline.ca is only if you are configuring your own SIP phone, you don't need to dial that for outbound calls.

-- The SIP URI calling feature only works with the ATA, depending on which one you get (If it's an $8 ATA from China it may not have this feature because they are generally "unlocked" from another service which can corrupt the core files in the ATA)

Thanks

[ececon]

Yes of course you can, and yes it does.

The current SIP format is - 1519xxxxxxx@voip.freephoneline.ca

I've created a DID at Voip.ms setting it up to forward my calls to the SIP URI of my FPL account above (1xxxxxxxxxx@freephoneline.ca) and it doesn't work (I get a busy signal).

I got a reply from FPL support saying that my scenario is not supported. Did I misunderstood admin's reply above?

[FONGO_kris]

Yes of course you can, and yes it does.

The current SIP format is - 1519xxxxxxx@voip.freephoneline.ca

I've created a DID at Voip.ms setting it up to forward my calls to the SIP URI of my FPL account above (1xxxxxxxxxx@freephoneline.ca) and it doesn't work (I get a busy signal).

I got a reply from FPL support saying that my scenario is not supported. Did I misunderstood admin's reply above?

SIP Protocol has changed in the past while, at least in our system. Try to use the IP you get from pinging voip.freephoneline.ca

-- 1xxxxxxxxxx@208.65.240.142

[ececon]

Hi, thanks for the quick reply!

I've just tried changed the URI to 1xxxxxxxxxx@208.65.240.142, and it still doesn't work... I still get a busy signal.

[FONGO_kris]

Hmm, I'll have to test this at the office tomorrow. Theoretically this should work since our server uses SIP URI protocol, we just trunk on the CID (caller id) so it can route through the PSTN (analog phone line) network.

<-- That was the "basic" version, it IS accurate for those users who "Just want to use the phone!" and aren't particularly interested in the architecture of our service. .

[ececon]

Thanks! I'll be happy if you can provide an update on that later!

In case this helps, I've just tested a SIP URI call from my computer using sjphone and the result of calling 1xxxxxxxxxx@208.65.240.142 is: "Call rejected: 403 Rejected: empty username is not allowed".

[FONGO_kris]

Okay I'll look up the SIP logs right now if you PM or email me your FPL number..

PS:

SIP URI = sip:x@y:Port
Where x=Username and y=host (domain or IP)

Examples:
sip:joe.bloggs@212.123.1.213
sip:support@phonesystem.ca
sip:22444032@phonesystem.ca

It's hard to dial SIP URI to a device or software with a dynamic IP....

[johoja]

I can't get this to work either?! Is there any update to this?

[FONGO_kris]

Not that I can see, I would imagine you would need a static IP for this to work if you're having this much trouble. What happens when you dial out?

[james_c]

Not that I can see, I would imagine you would need a static IP for this to work if you're having this much trouble. What happens when you dial out?

Kris, are you sure FPL do accept SIP URI call in?

I tried to point my ipkall calls to FPL by SIP URI and never work. And Sipbroker doesn't work with FPL either, I think they are using SIP URI like method to connect calls.

[mjbad2]

I Think FPL has to be clear about this issue. Don't dodge it. It doesn't work.
So better find out why and how it should work.

[FONGO_kris]

I'm not sure why you guys are so upset. It works for me here on my Snom m3 DECT phone, I have a static IP set to my phone which is ALWAYS plugged in, through my Thomson 546i ADSL2+ router, on an al1-fastpath ADSL2+ dry-loop line.

It also works on SipSorcery which though it is under construction still works. I'll try and find an easier way for you guys to do it, but my recommendation is - Use an ATA.

[james_c]

I'm not sure why you guys are so upset. It works for me here on my Snom m3 DECT phone, I have a static IP set to my phone which is ALWAYS plugged in, through my Thomson 546i ADSL2+ router, on an al1-fastpath ADSL2+ dry-loop line.

It also works on SipSorcery which though it is under construction still works. I'll try and find an easier way for you guys to do it, but my recommendation is - Use an ATA.

Not sure if we are talking on the same thing. I guess you are thinking we want to SIP URI FPL calls to our ATA or Softphone? That's why you keep saying static ip etc?

What I'm trying to do is forward my other DID calls to FPL by SIP URI, my ATA or Sipsocery or Pbxes is registered with FPL, so i'm expecting my phone will ring once these calls come in. But it doesn't. If Iforward my DID calls to Sipsocery or Pbxes by SIP URI, they rings on my phone. So my conclusion is FPL doesn't accept SIP URI call in from other DIDs.

[FONGO_kris]

I forwarded my Asterisk DID to my Freephoneline number which is plugged into an ATA:

1519804xxxx@xx.38.192.176:22517

I also called this number from another softphone on another computer and it worked perfectly.

What is happening when you dial it?

[james_c]

I forwarded my Asterisk DID to my Freephoneline number which is plugged into an ATA:

1519804xxxx@xx.38.192.176:22517

I also called this number from another softphone on another computer and it worked perfectly.

What is happening when you dial it?

Alright, what you are doing is you forward your call to your ATA directly with SIP URI....you can try to dial XXX@xx.38.192.176:22517, i bet your phone will ring as well.

What I'm trying to do is forward the call to FPL, so my dial from other DID is 1647XXXXXXX@voip.freephoneline.ca or 1647XXXXXXX@208.65.240.142 (which is the voip.freephoneline.ca ip). Currently i'm getting busy tone in the other end, and my phone never ring. And I don't see the call shows in the FPL call log.

[FONGO_kris]

My SIP URI:
1519804xxx@xx.xx.192.176:22517

What it means:
1519804xxxx = Pretty easy to guess, your 11 digit phone number
@ = If you need clarification, you shouldn't be allowed to own a computer
xx.xx.192.176 = IP Address of your ATA
22517 = Port number of your ATA

I think I did this earlier, if I didn't, sorry but it's here now!

[james_c]

My SIP URI:
1519804xxx@xx.xx.192.176:22517

What it means:
1519804xxxx = Pretty easy to guess, your 11 digit phone number
@ = If you need clarification, you shouldn't be allowed to own a computer
xx.xx.192.176 = IP Address of your ATA
22517 = Port number of your ATA

I think I did this earlier, if I didn't, sorry but it's here now!

What i'm trying to say is you don't need to use your FPL number at all for the above, you can just use any dummy number before the @...as you are dialing to your ATA and nothing to do with FPL.

Only you can see your calls in FPL call logs means it went through FPL....do we agree on this??

[FONGO_kris]

I agree, yes.

Any objections?

I am doing some testing to see if we can do it the other way as well:

1519804xxxx@voip.freephoneline.ca and then from the NUMBER it will find the IP of your ATA and ping it

[james_c]

I agree, yes.

Any objections?

I am doing some testing to see if we can do it the other way as well:

1519804xxxx@voip.freephoneline.ca and then from the NUMBER it will find the IP of your ATA and ping it

Ok, now we are back to the right channel. If the 1519804xxxx@voip.freephoneline.ca works, I'm expecting the Sipbroker access route will work on FPL as well, that will be a big enhancement for FPL.

[FONGO_kris]

It would yes, give me some time though... It's a headache.

[james_c]

It would yes, give me some time though... It's a headache.

no problem...i'm using some work around to get the feature works...if FPL fix it then will simplify my settings...

[johoja]

It would yes, give me some time though... It's a headache.

no problem...i'm using some work around to get the feature works...if FPL fix it then will simplify my settings...

What's your workaround ?

[james_c]

It would yes, give me some time though... It's a headache.

no problem...i'm using some work around to get the feature works...if FPL fix it then will simplify my settings...

What's your workaround ?

I use Voxalot to get SIP URI calls and Sipbroker calls. Set it up as a line on your ATA or as a trunk if you use pbxes

[nelsoon]

Sorry to revive a old thread but I wanted to know how SIP-2-SIP calling is going at FPL?

I Know that outgoing call from FPL to Sip PBX / Phone is working but the opposite is not. When my friend try to call me on SIP protocol, voip.freephoneline.ca reject his call. Is it because i didn't set parameters correctly or it's another thing?

Why i want to do calls between a SIP phone, FPL Pbx and my SIP phone? Because i like talking with G722 quality, because theses days no phone connection is secure except perhaps ZRTP but again i'm not so sure and because my phone is Acrobits Softphone Business/1.6 on an Bell iPhone that is behind a NAT so it's impossible to call it directly, it need the PBX to do it.

Thanks in advance!