I am running PBXact (and I am new to the cloud) host on Virtual server. I have gotten all my phones to work such as Yealink and Fanvil phones manually no issue BLFs and call park work perfect. I can not get the Grandstream 2160 to work at all or any grandstream phones please help. Like I said i am manual configuring them
Does it try to register at all? What do you see in the logs?
Manually configuring Grandstream’s is pretty straightforward, there’s nothing special about it. If you can configure Yealinks and Fanvil’s then you shouldn’t have a problem with Grandstream.
Configure Account 1, input your SIP Server, SIP Username & Password, apply settings. Click on the status option at the top of the web GUI/view your logs to confirm registration.
Does your phone register?
The phone did not register at all. I have used the grandstreanm 2160 on other pbx’s I know how manual to do it. I am not blond… (lol) I am blond… That is why driving me crazy. Also I am new to Freepbx so please bare with me.
As asked earlier: what do you see in the FreePBX logs?
Is there a possibility that your phone got banned by fail2ban?
I did not see in the fail2ban, i am not sure how to check the FREEPBX logs? Someone had mention that the TCP stacks has to be reloaded which I am not sure how to do that.
to view logs just type in asterisk -rvvvvvvv (more v’s for more verbosity) from the CLI… You should see registration attempts from your Grandstream.
I finally got to register… setting on the phone for the firewall was wrong, now I have one way audio. I can hear the call the out but no audio on the Freepbx side. I did configure anther freepbx with same setting but I upgraded that account PBXact account no issues with audio. So I am wondering what is the difference on the trunk settings
Inbound
type=friend
insecure=very
host=64.136.173.31
dtmf=auto
disallow=all
context=from-trunk
canreinvite=no
caninvite=no
allow=ulaw&alaw
outbound
type=friend
insecure=very
host=64.136.174.30
dtmf=auto
disallow=all
context=from-trunk
canreinvite=no
caninvite=no
allow=ulaw&alaw
These are for voipinnovations
I saw that the codec for the call PCMU on the phone
Thank you guys for the help Grandstream GPX2160… It is working … Now to tackle one way audio, Free Version freepbx.
One way audio issues are going to NAT related… Ensure the correct ports on your firewall are forwarded, check Settings > Asterisk SIP Settings to ensure your External IP is correct, set local networks, check your extension settings > NAT Mode = Yes if your phone isn’t on the same LAN as your PBX, No if your phone is on the same LAN as your PBX, etc.
That worked thank you so much. It was the setting on the extension side of the pbx. Nat was off…
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