Freephoneline.ca



  • Hello everyone!  Has anyone been able to get freephoneline.ca to work through pfsense.  Nothing I try seems to work!



  • tried siproxd?



  • Did you get pfsense working with freephoneline?  I'm having the same issue, can't get any calls in, although the portal shows me as registered.  I have other SIP providers behind the firewall, and they work fine.  FPL also works fine when using a DD-WRT box.  This is really hurting the WAF.  And no, SIPROXD didn't solve anything.



  • try to get some extra info from provider.

    It's hard to help without information.

    Some providers needs sip ports other then 5060.

    Try to tcpdump via console on wan and other console on lan to see where it stops.



  • When I tried (and I didn't try to hard) to get my ATA to connect to any other port than 5060 on Siproxd it did not work. I see that the pfSense log reports that Siproxd was bound to the other ports when I tried but Siproxd was not working. Once I got the ATA to connect to the providers server on 5060 all was good…  The port your ATA allows connection on is not a factor...

    But-  Look at your "States" for the ATA...  Paste em here.  (Dont use your real public IP.)

    SIP providers can work in different ways...  Vonage uses 10000 as a SIP port.  My provider uses there own SIP servers (5060-5080) but audio (in most cases) comes directly from the carrier server. Others do this too. Think of how your firewall would and should react in this situation...  RTP ends up coming from a strange new source suddenly...

    My provider recommends that users port forward to their ATA on your network. This can throw a monkey wrench into the operation of your other providers equipment though...



  • From their FAQ-

    Q.What do I need to do to get the softphone functioning properly after installing it?
    A.Our softphone operates on UDP ports 13000, 13001, 5060, 5061, 6060 and 6061. Please make sure these ports are opened in any firewall, internet security suites and hardware routers you may have in your home.

    Q.When I call somebody, only one party can hear (one way audio)
    A.Please make sure UDP ports 13000, 13001, 5060, 5061, 6060 and 6061 are opened in your firewall and internet security suite. If you have a router, you will also need to forward these ports to the computer with the softphone on.



  • Im trying to sign up to help understand their service better but their website is very intermittent.

    I believe that they use ports 6060 and 6061 for alternate sip ports…  But 5060 should work just fine.



  • Yes, the website has been flaky since yesterday, seems better today.  Currently I'm forwarding my FPL number to one of my other providers as a workaround.

    Do I did capture the internal and external interfaces and I do see the Invite arrive on the public interface, on port 5060.  Nothing appears to get passed to the internal side.

    In my NAT rules I have voip.freephoneline.ca allowed (last ditch attempt), port 5060-5061 & RTP 10000-20000 forwarded to my Asterisk server, this works fine with DD-WRT.  I'll get some fresh captures in a few minutes.

    @chpalmer:

    Im trying to sign up to help understand their service better but their website is very intermittent.

    I believe that they use ports 6060 and 6061 for alternate sip ports…  But 5060 should work just fine.



  • Their "free" service is the use of their own softphone, which uses 6060, 6061 & 13000/1.  SIP costs you $50 one time fee to use.

    @chpalmer:

    Im trying to sign up to help understand their service better but their website is very intermittent.



  • @tbrummell:

    Their "free" service is the use of their own softphone, which uses 6060, 6061 & 13000/1.  SIP costs you $50 one time fee to use.

    @chpalmer:

    Im trying to sign up to help understand their service better but their website is very intermittent.

    Ah- well that might put a damper in it eh!?

    You might need to do static port for that particular ATA also…



  • OK, finally back at this, I've had a workaround in place since I originally posted so it was put to the back-burner to simmer.  :)

    Here is a complete, un-sanitized capture of TWO calls.  The first call, placed via my working provider (Voip.ms).  The second via FPL.  The capture is a result of merging a tcpdump at the firewall, and on the Asterisk server.  On the FPL call, you see the call hit the firewall, but never make it to the Asterisk server.  5060 is forwarded to Asterisk, it should be there.

    Any idea's?

    http://brummell.net/CompleteCapture.zip



  • Well, I got it working.  I found this http://www.freepbx.org/forum/freepbx/tips-and-tricks/asterisknow-freepbx-and-pfsense post that helped me.  It's totally backwards thinking to how I'd normally do it, but my FPL line is now up and working.  We'll see how long it lasts…


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