Solved extensions register unregister

Status
Not open for further replies.

Dan Nica

Free User
Joined
Jan 7, 2019
Messages
22
Reaction score
1
So my challanges with 3CX continues :) since a while I noticed that the extensions that have phones (Grandstream different models) are unregistering/registering after a while, and obviolusly I can not call on that extension, any hints on this behaviour?

Code:
01/28/2019 12:14:39 PM - Endpoint Extn:100 has removed contact <sip:100@IP_ADDRESS:1047>
01/28/2019 12:14:39 PM - [CM504002]: Endpoint Extn:100: a contact is unregistered. Contact(s): []
01/28/2019 12:13:39 PM - Endpoint Extn:100 has removed contact <sip:100@IP_ADDRESS:5065>
01/28/2019 12:13:39 PM - [CM504002]: Endpoint Extn:100: a contact is unregistered. Contact(s): [sip:100@IP_ADDRESS:1047 / 100]
01/28/2019 12:12:38 PM - Extn:100: Synch devices to dndOFF
01/28/2019 12:12:37 PM - Extn:100: Starting DND change transaction to dndOFF requested from a device
01/28/2019 12:12:37 PM - Device's (<sip:100@IP_ADDRESS:1047>) DND state is dndOFF for Extn:100
01/28/2019 12:12:37 PM - Endpoint Extn:100 has added contact <sip:100@IP_ADDRESS:1047>
01/28/2019 12:12:37 PM - [CM504001]: Endpoint Extn:100: new contact is registered. Contact(s): [sip:100@IP_ADDRESS:1047 / 100,sip:100@IP_ADDRESS:5065 / 100]

Thanks,
Dan
 
Hi Dan,

The fragment you have sent simply outline as register/deregister as well as showing your using enabled DND on/off - this however would not trigger a de-register.

Run Wireshark from the PBX and see what it says about the register messages - it should give you a reason also: https://www.3cx.com/docs/capture-network-traffic/

Can you also outline where these extensions are in regards to the PBX ? Local, remote (and method).
 
my extensions are behind a NAT network on a different than the PBX which is using a public IP, so no nat on the PBX side.

I will try to use debug en extenion
 
yes, I am using STUN, actually I have more than 8 SIP phones behind the nat router, so you are suggesting to set uniq port for RTP and SIP, where can I set that?
 
Although this could work, I find STUN with more than a couple of phones a little sporadic (if on the same site) and would probably recommend you move across to using the 3CX SBC on Raspberry Pi (which can take upto 20 handsets currently) if possible.

As per: https://www.3cx.com/docs/manual/configuring-ip-phones/

"STUN - 3CX in the Cloud - for sites with 1 or 2 IP phones where a local SBC is overkill."
 
ok, so if I just bypass the NAT router and reach directly to pbx I should be able to avoid this issue, correct?
 
Personally, I wouldn't deploy more than a single phone without an SBC.
 
yes, I am using STUN, actually I have more than 8 SIP phones behind the nat router, so you are suggesting to set uniq port for RTP and SIP, where can I set that?
The ports are set under the extension settings / Phone provisioning tab. After you set the ports you will need to forward these ports to the local IPs of the phones and reprovision the phones so the apply the new ports.
As mentioned however you could also use an SBC that eliminates the need for firewall configuration.
https://www.3cx.com/3cxacademy/videos/intermediate/configuring-remote-extensions/
 
So I managed to "stabilize" the phones, since we own the network, I was able to transport a separate voip only vlan directly into the PBX,

thanks for suggestions
 
Glad to see the issue has been resolved and thank you for updating the thread with your solution.
 
Status
Not open for further replies.

Forum statistics

Threads
111,913
Messages
589,706
Members
164,782
Latest member
David853