Gransdstream HT502 with VMedia ISP (Hitron and Eero)

[louclou]

Hello,
I change my ISP for Vmedia with Hitron cable modem CODA56 and Eero N010001 routeur, since I did this change, I am unable to send or receive calls.
Someone call help me with that ?
(Before I was with cable internet also but with Harris modem TM822G and routeur Nighthawk AX6 AX4300)
Thank you

[Liptonbrisk]

If the only changes made were the CODA56 and Eero 6 devices, put CODA56 in bridge mode (contact Vmedia for help), and connect your ATA to Eero 6 router. Supposedly, Eero routers don't have SIP ALG. If that's true, you should be good to go afterwards. Regardless, ask Vmedia if Eero 6 has a SIP ALG function, and if there's a way to disable it.

You can also try voip4.freephoneline.ca:6060 for Primary SIP Server in the ATA.

This is what you want:

CODA56 in bridge mode--->Eero 6 (with SIP ALG disabled, if possible)--->HT-502 (using voip4.freephoneline,ca:6060 for primary SIP server, particularly if SIP ALG exists and can't be disabled)

Otherwise, click viewtopic.php?t=20539, and follow the steps.

[Liptonbrisk]

Since I can't find VMedia bridge mode instructions for the CODA56, here's the general process based on similar Hitron modems/router combos:

a) Log into the CODA56, which is a modem/router combo, at 192.168.0.1 (in router/gateway mode) or 192.168.100.1 (bridge mode)

Username might be cusadmin

Password might be "password" (without the quotation marks) or CODA56's default WiFi password shown on the unit.

b) Navigate to Basic settings

c) Look for "Gateway Function" or "Residential Gateway Function"

d) Change from "Enable" to "Disable", or, preferably, select "Bridge Mode" and enable that.

e) Save changes, and reboot CODA56.

Note that bridge mode disables WiFi and DHCP in the CODA56, requiring your Eero router to handle all network functions. So, you will need to connect all devices to your Eero 6 either via network cables or Wi-Fi.

If Eero 6 contains a SIP ALG function that can't be disabled (I'm told that Eero does not have SIP ALG at all though), then you should use voip4.freephoneline.ca:6060 for primary proxy server in your ATA.


This is what you want:

CODA56 in bridge mode--->Eero 6 (with SIP ALG disabled, if possible)--->HT-502 (using voip4.freephoneline,ca:6060 for primary SIP server, particularly if SIP ALG exists and can't be disabled)

[Liptonbrisk]

Otherwise, click viewtopic.php?t=20539, and follow the steps.

By the way, your ATA firmware version should be 1.0.16.2: https://firmware.grandstream.com/Releas ... 0.16.2.zip
https://www.grandstream.com/support/firmware
I accept no responsibility for failed firmware updates.


1.
a) Dial ***
b) Then dial 02
c) Enter the IP address you hear into a web browser.
d) Login to your ATA.

Default login password is admin.


2. For HT-502, navigate to the FXS Port tab.

A. If you could be dealing with a SIP ALG problem (ISP's hub or gateway that’s been provided has SIP ALG on with no way for the user to disable it, or if you can't figure out how to disable SIP ALG in your own router), navigate to “Primary SIP Server”.


Enter "voip4.freephoneline.ca:6060" (without the quotation marks) to help avoid any potential SIP ALG bug. You should be testing with voip4.freephoneline.ca:6060 if you're getting 1-way audio problems (one side hears audio, and the other side doesn't), or if calls are failing. Anyone can use voip4.freephoneline.ca:6060, even people who don't use Rogers. If you have 1-way audio problems, use it.


B. “NAT Traversal” should be set to “Keep-Alive” (newer ATA models) or “Yes” with the STUN server field blank (Grandstream HT-286/HT-287). Don't use STUN (when the STUN server drops so will your Freephoneline service).

C. Enable SIP OPTIONS/NOTIFY Keep Alive

Select “Notify”

or Ensure “NAT Traversal” is set to “Keep-Alive”



D. Set SIP OPTIONS/NOTIFY Keep Alive Interval to 20 seconds.


If you have the option to enter the keep-alive interval, set it to 20 seconds.

The user manual suggests the option to change keep-alive interval exists: https://www.grandstream.com/hubfs/Produ ... manual.pdf

"Keep-alive interval
This parameter specifies how often the HT502 sends a blank UDP packet to the
SIP server in order to keep the “hole” on the NAT open. Default is 20 seconds.
Minimum value is 20 seconds."
(page 35)




E. Register Expiration needs to be 3600 seconds or 60 minutes.

Please take note of whether your ATA’s firmware designates minutes or seconds for this setting.

The value needs to be either 60 minutes or 3600 seconds (1 hour).
You can’t enter 60 seconds without getting your Freephoneline account blocked. Be careful here.

HT-502/HT-503/701/704/8xx series ATAs appear to use minutes for this setting. If so, the correct value is 60 minutes. Double check.

F. i) Set “Use Random SIP Port” to “Yes”.
ii) Set “Random RTP Port” to “Yes”.


G. Set “SIP Registration Failure Retry Wait Time” to 120 seconds


H. Refer to page 4 (bottom) of the PDF Guide located at viewtopic.php?p=64438#p64438. The PDF guide link is located at the bottom of the first post.

a) The preferred Vocoder (in listed order) should be PCMU (first choice), which is the G.711u audio codec. G729 can be the second choice, but it sounds worse. If you have any other codec listed, change it to PCMU. Freephoneline only supports PCMU (G.711u) and G.729a. If you use anything else, calls can fail.


3. Update/Apply (save) ATA settings if changes were made. Turn off modem, router, and ATA. Turn on modem. Wait for it to be fully up and running first. Turn on router. Wait for router to be fully up and transmitting data first. Lastly, turn on ATA after everything else is up and running. That's always the proper device boot order. Your ATA should always be booted last in the device chain. 1. Modem (wait) -->2. Router (wait)-->3. ATA

4. Login at https://www.freephoneline.ca/showSipSettings.
SIP Status needs to indicate "Connected", and SIP User Agent should reflect the device you're using.

5. Login to your ATA. Ensure your FPL account is registered with your ATA.

For HT-503/701/704/8xx series ATAs, navigate to Status–>Port Status–>FXS–>Registration (also ensure DND shows “No”).

[Note that unless you own two VoIP unlock keys, you can't register two Grandstream FXS ports with your FPL account simultaneously.

When there are multiple devices/softphones/FXS ports (on an ATA) using the same FPL account and SIP server, only the most recent registration is valid. The previously registered device (or FXS port, in this case) will lose registration, and, consequently, incoming calls will not ring on it. This is especially important to consider if someone else is using your SIP credentials (username and password) that are found after logging in at https://www.freephoneline.ca/showSipSettings or if you're trying to register your FPL account with a smartphone SIP app (or FPL desktop app), with another device, or more than one ATA line. Registration is required for incoming calls. It is not required for outgoing calls. Only one registration per FPL account is allowed at any time. A single FXS Port on a Grandstream ATA counts as one registration. A SIP app is another.]


6. You can try calling the 250 or 416 number at https://thetestcall.blogspot.com/. If it doesn't work, check the dial plan in the ATA.

Try using this:

Code: Select all

 {[2345689]11|*98|[2-9]xxxxxxxxx|1[2-9]xxxxxxxxx|011xxxxxxxxxxxx.}

Official VoIP Unlock Key settings are found at https://support.freephoneline.ca/hc/art ... redentials.