Quick update. The issue seems to have been fixed after doing the following.
I setup the Grandstream for PPPoE and it seems to have worked. However, I am not clear on why PPPoE is needed? I understand my DSL modem needs it for authentication and for pushing down the SLAs, but not able to figure out why the ATA needs a PPPoE? Kindly, if you can clarify it that would be great.
Here is a summary of the issue incase some others ran into it.
after setting up the Granstream with freephoneline, incoming calls become not possible after 30-60mins, however outgoing calls continues to work all the time.
VOIP box Grandstream HT812
ISP = Teksavvy, using DSL modem/router Smart/RG Model # SR515ac
The following needs to be done on the DSL modem. need to check the box beside "Bridge PPPoE Frames Between Wan and Local Ports" (this is under advance setup--Wan Service)
on the Voip HT812 Grandstream box, you need to do the following.
Under "Basic setting"
click the radio botton beside "use PPPoE"
filling in the following 3 fields
- PPPoE Account ID (your service provide needs to provide it to you, make sure this is a different value than your DSL modem PPPoE account)
- PPPoE password (your service provide needs to provide it to you, make sure this is a different value than your DSL modem PPPoE account)
PPPoE service Name (just make up a name)
Don't forget to scroll to the bottom of the screen and hit update/apply
from the top menu, Click on the “FXS Ports” tab
SIP User ID = your phone # , example 1613-555-5555
Authenticate ID = your phone#, example 1613-555-5555
Password = need to obtain this by visiting this link
https://www.freephoneline.ca/showSipSettingsDon't forget to scroll to the bottom of the screen and hit update/apply
Under “Profile 1” tab
Enter the following
Primary SIP Server: voip.freephoneline.ca
◦ • NAT Traversal: Keep-Alive
• Outgoing Call without Registration: No
• SIP Registration Failure Retry Wait Time: 120 (there are two entries for sip registration failure, only populate the first one, leave the other one with the default value)
• Enable SIP Options Keep Alive: Yes
• SIP OPTIONS Keep Alive Interval: 20
• Use Random SIP Port: Yes
• Use Random RTP Port: Yes
• Transfer on Conference Hangup: Yes
• Allow Incoming SIP Messages from SIP Proxy Only: Yes
• SIP REGISTER Contact Header Use: WAN Address
• Preferred DTMF method: (in listing order) ◦ Priority 1: RFC2833 ◦ Priority 2: In-audio ◦ Priority 3: SIP INFO
• Enable Call Features: No
• No Key Entry Timeout: 4
• Preferred Vocoder: (in listed order): ◦ Choice 1: PCMU ◦ Choice 2: G729 ◦ Choice 3: PCMU ◦ Choice 4: PCMU ◦ Choice 5: PCMU ◦ Choice 6: PCMU
- Dial Plan: copy/paste to the following string
{*xx|[2-9]xxxxxxxxx|1[2-9]xxxxxxxxx|0xxx+} Click update/Apply at the bottom.
You might not be prompted to reboot. In this case, after you applied changes: • Go back to the FXS Port tab and click Reboot at the bottom.
Phone should work
incase you need to upgrade the firmware of the ATA, mine was one release behind, however, I did upgrade it.
to do so, make sure you disable PPPoE on your ATA, have the WAN on your ATA connect to your router (in my case dsl modem)
from the top menu click on “Advance settings”
select HTTP which is located beside the "firmware Upgrade and Provisioning"
in the "Firmware server Path" type "firmware.grandstream.com
in the "Config Server Path" type "firmware.granstream.com
scroll to the bottom, hit update/apply then reboot.
it will take a minute or two to come up, however, after it comes up, give it approx 5 mins to upgrade.
this worked for me, hopefully it will help anyone that is experiencing the same issue.
Cheers,
Rafi