Liptonbrisk wrote:Latest firmware for your ATA can be found at http://www.grandstream.com/support/firmware
Follow the steps below, step by step, down the list.
1. Are you using a Telus Wi-Fi Hub? NO
For Telus Wi-Fi Hubs, put LAN port 1 on the Hub in bridge mode (I've had to go through this process before for a family member). Then connect pfSense router/gateway to LAN port 1.
Alternatively, contact Telus and ask them how to enable bridge mode.
2. Make sure you're not using Siproxd, which is similar to SIP ALG, in pfSense. It wouldn't make sense that you are using Siproxd, but you might want to double check to ensure it's not installed. Or make sure that it's disabled. NO
A number of people using Asus routers have had to disable SIP Passthrough, which is the setting for SIP ALG in Asus routers, to get incoming calls from Rogers working:
http://forum.fongo.com/viewtopic.php?f=8&t=20211#p79016
viewtopic.php?f=8&t=20182#p78916 (Fido is Rogers)
I realize this isn't a Rogers issue, but it could be SIP ALG related (in either the Telus Hub or with Siproxd).
3. Ensure the ATA is not running behind a VPN. NO
4. In your ATA, set Primary SIP Server to "voip4.freephoneline.ca:6060" without the quotation marks. The purpose of voip4.freephoneline.ca:6060 is to help circumvent buggy SIP ALGs (which is why bridge mode in Telus Hub should be enabled and Siproxd not installed).
Apply changes.That is what I have
5. Ensure that (in your ATA)
a)SIP REGISTER Contact Header Uses "WAN Address" Just changed from LAN to WAN
b) NAT Traversal: "Keep-Alive" YES
c) Register Expiration: 60 minutes YES
d) SIP Registration Failure Retry Wait Time: 120 seconds Changed from 20 to 120
e) OPTIONS/NOTIFY Keep Alive is set to anything other than NO
f) SIP OPTIONS/NOTIFY Keep Alive Interval is 20 seconds Changed from 30 to 20
g) Use Random SIP Port: YES Do you really need this, I have already port forwarding enabled for a specific one?
h) Use Random RTP Port: YES Do you really need this, I have already port forwarding enabled for a specific one?
Apply changes.
6. Reboot Modem/Telus Hub. Wait for it to be fully up and running. Reboot pfSense device. Wait for it to be fully up and running. Reboot ATA
This is always proper device reboot order.
7. a. Login at https://www.freephoneline.ca/showSipSettings
b. Ensure that SIP Status shows "connected"
c. SIP User Agent reflects the ATA you're using
SIP Status: connected
SIP User Agent: Grandstream HT802 1.0.11.6
8. Navigate to https://www.freephoneline.ca/voicemailSettings
a. Ensure Rings Before Voicemail is not set to 1.
b. Change Rings before voicemail to 12. Click "Update".
c. Change Rings before voicemail to 5 (or something reasonable). Click "Update".
Just changed 6
9. Dial *79 to disable Do Not Disturb just in case. Confirmed
10. Test again with incoming call from Telus number
11. If 10 doesn't work, try testing with the ATA connected directly to the Telus Hub LAN port 1 to rule out pfSense being the issue. (will try that too... don't think that should be issue as all other calls comes fine)
By the way, kiekar believed all incoming calls weren't working, but incoming calls from Bell carrier numbers worked while others (unspecified carrier) didn't. After reinstalling a fresh version of pfSense (not a backup of the config file), all incoming calls worked: viewtopic.php?f=38&t=20221&p=79059#p79060.
12. Do incoming calls from other Telus numbers from that same area code work? If so, then submit a ticket: https://support.fongo.com/hc/en-us/requests/new
Thanks will try that last
Provide the phone number that doesn't work in the ticket and mention that it's a Telus carrier number. You will likely receive an automated response, but hopefully someone will take a look without automatically closing the ticket.
Jas63 wrote:
1. Are you using a Telus Wi-Fi Hub? NO
e) OPTIONS/NOTIFY Keep Alive is set to anything other than NO
g) Use Random SIP Port: YES Do you really need this, I have already port forwarding enabled for a specific one?
h) Use Random RTP Port: YES Do you really need this, I have already port forwarding enabled for a specific one?
Grandstream HT802 1.0.11.6
Liptonbrisk wrote:By the way, kiekar believed all incoming calls weren't working, but incoming calls from Bell carrier numbers worked while others (unspecified carrier) didn't. After reinstalling a fresh version of pfSense (not a backup of the config file), all incoming calls worked: viewtopic.php?f=38&t=20221&p=79059#p79060.
Liptonbrisk wrote:Jas63 wrote:
1. Are you using a Telus Wi-Fi Hub? NO
What is the brand and model modem being used? If it's a Telus modem/router combo or gateway, ensure that it's in bridge mode so that SIP ALG is not enabled in it and that its router features can not interfere in any manner.
Sometimes the devices that ISPs issue to customers can be the source of the problem.
TELUS fiber box Nokia terminates into pfSense which gets Public IP.e) OPTIONS/NOTIFY Keep Alive is set to anything other than NO
It's NOTIFY
I hope you have either NOTIFY or OPTIONS selected.g) Use Random SIP Port: YES Do you really need this, I have already port forwarding enabled for a specific one?
I want to mention a couple of points here:
1. Port forwarding is a security risk. Do not use UDP 5060 or 5061 for the local SIP port and forward it. You'll just be opening yourself up to SIP Scanners/hackers.
Those are well-known ports to target for attacks.
Thanks I moved to random high number.... although it should not have matter, as I set source DNS to voip4.freephoneline.ca.
If you don't want to use the random sip port setting, I recommend specifying a high, random, local SIP port in your ATA between 30000 and 60000. Just pick a UDP port number in that range (provided it's not being used by anything else on your LAN). Do not use the same random SIP port for any other Line. Always choose a different random local SIP port for each FPL account you're using.
Using a high random sip port may help avoid SIP ALG issues and sip scanners.
You'd have to specify the UDP local SIP port you want to use in the ATA, and then forward it, if you absolutely need to port forward. Port forwarding should
only be used if the service doesn't work without it.
2. Using random SIP port can help with one of the problems mentioned in point D from viewtopic.php?f=8&t=20199#p78976.
But no, it's not a requirement. Otherwise other ATAs and IP Phones that don't have the random SIP port setting wouldn't work.h) Use Random RTP Port: YES Do you really need this, I have already port forwarding enabled for a specific one?
Not absolutely required, but see above.
They're recommended: https://support.freephoneline.ca/hc/en- ... redentials
"Local SIP Port: RANDOM
Local RTP Port: RANDOM"Grandstream HT802 1.0.11.6
ATA firmware hasn't been updated to 1.0.33.4
https://firmware.grandstream.com/Releas ... 0.33.4.pdf
https://firmware.grandstream.com/Releas ... 0.33.4.zip
I will not be held responsible for failed firmware updates.
Surely, I will update itLiptonbrisk wrote:By the way, kiekar believed all incoming calls weren't working, but incoming calls from Bell carrier numbers worked while others (unspecified carrier) didn't. After reinstalling a fresh version of pfSense (not a backup of the config file), all incoming calls worked: viewtopic.php?f=38&t=20221&p=79059#p79060.
That link might be worth taking a look at. Again, try testing with the ATA connected directly to whatever modem (again, if it's a modem/router combo or gateway, ensure that it's in bridge mode so that its router features can't possibly interfere or block anything) Telus gave you to rule out whether pfsense is involved in the problem.
Jas63 wrote:TELUS fiber box Nokia terminates into pfSense which gets Public IP.
Jas63 wrote:
I first thought my cordless phone blocking it, but it does not have any such feather, then I check my VoIP device, same.
Jas63 wrote:I will try soft phone as well.
If that also same, that means not even related to ATA
Jas63 wrote: it just goes to VM
phone never rings or missed call on my phone display
Jas63 wrote:162.213.111.21
Does this belongs to Fongo?
Liptonbrisk wrote:
In your ATA, set Primary SIP Server to "voip4.freephoneline.ca:6060" without the quotation marks. The purpose of voip4.freephoneline.ca:6060 is to help circumvent buggy SIP ALGs (which is why bridge mode in Telus Hub should be enabled and Siproxd not installed). Apply changes.
Jas63 wrote:.That is what I have
Jas63 wrote:SIP Status: connected
Jas63 wrote:ok, finally found the solution....
in pfsense you have to allow incoming from these 2 IPs.
162.213.111.21
208.65.240.44
Seems to be IPs belonging to Fongo servers.
Its working now.
Likely these are load balancing, Fango incoming server IPs change based on source phone number, it seems.
Please don't trust the ** they give you, just allowing SIP servers using port forwarding is not enough.
When the request hits their server, that trigger, secondary IPs that makes RTP connections.
Liptonbrisk wrote:
10. If that doesn't work, also allow incoming from 208.85.218.149 and 208.85.218.150 (FPL's RTP IPs for the media/audio stream)
208.85.218.150 (unknown IP belongs to FIBERNETICS CORPORATION, SHADY STUFF)
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