T54W issues with STUN

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jshdigital

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Hi there,

I have a few handsets which connect to my 3CX server which is in Azure.

Web client calls are fine, both receiving and calling out.

I setup 3 Yealink T54W handsets which use STUN. They auto provision fine, they can contact the server and pull down the config.

2 issues:

when calls come in, the phone can't answer them. They appear "answered" but it doesn't answer the call and the other two phones keep ringing.

Making calls out sits there without a dial tone until a invite request error appears on the phone.

Using wireshark I can see the phone is talking with the 3cx server fine until the invite is made, and then the 3cx server does not respond to the handset. Does anyone have any ideas?
 

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Seems I also get

01/11/2020 4:12:30 PM - [CM102001]: Authentication failed for AuthFail Recv Req REGISTER from 5.62.62.159:1150 tid=d360c2e8-4bfc-44b3-a1c4-9597c93c40ed Call-ID=vyyhmapoipvodeuqiluhtyvjgpjquvosvyrwctvqdskrjnauuo: REGISTER sip:51.140.6.164 SIP/2.0 Via: SIP/2.0/UDP 100.120.0.18:6815;branch=z9hG4bKd360c2e8-4bfc-44b3-a1c4-9597c93c40ed;rport=1150;received=5.62.62.159 Max-Forwards: 70 Contact: <sip:[email protected]:6815;rinstance=fa0e6f5eb782f143> To: "274"<sip:[email protected]> From: "274"<sip:[email protected]>;tag=ogfhmvdp Call-ID: vyyhmapoipvodeuqiluhtyvjgpjquvosvyrwctvqdskrjnauuo CSeq: 2 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE Proxy-Authorization: Digest username="274",realm="3CXPhoneSystem",nonce="414d53595e19f3ee30:a8334886ab2085950c58b0ec8bcc5a75",response="a3dd51c5fcb34416adf1324912fa051d",uri="sip:51.140.6.164",algorithm=MD5 Supported: 100rel User-Agent: Avaya Content-Length: 0 ; Reason: Credentials don't match, check that authorization-ID and password match the ones in extension settings


I should note, I have turned off the "prevent use outside LAN" so it's not that.
 
Before going further, please check if each phone use its own port, e.g.:
Phone 1: SIP-5062 RTP 14000-14019
Phone 2: SIP-5063 RTP 14020-14039
Phone 3: SIP-5064 RTP 14040-14059
 
each phone is using 5065
 
This shows in the activity log...

01/11/2020 4:48:05 PM - Exception: ParseException ../../rutil/ParseBuffer.hxx:230, Parse failed unexpected eof in context: Voice Mail ^ @ ../../rutil/ParseBuffer.hxx:230
 
Hi thanks for your help

I'm unsure what this means

"and then port forwarding on the phone said of said ports to each phone"

On the firewall right?
 
If you arent familiar with the setup requirement and configuration on the firewall your best option would most probably be to use the 3CX SBC instead of STUN.

If you are and just need a bit of guidance then the following applies:

  • Each endpoint needs a static (or statically assigned via DHCP) IP address.
  • Their own incremented sets of ports for SIP and RTP - Which do not clash
  • Port forwarding for each of these sets through the firewall to each phones static IP address.
See the attached I drew to outline this using a hosted 3CX and Draytek.
 

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thanks guys, will revert when tried
 
Hi @jshdigital

Also make sure you have the option "PBX Delivers Audio" on the extensions that use STUN.

Once the ports are assigned as per the recommendations above, you will need to reset and then reprovision the phones to ensure they accepted the changes.

Then those ports need to be forwarded at the phone side router.

This can get tricky if you have more than 2-3 phones there, at which point a SBC begins to makes sense. You can install one on a small PC or even a Raspberry Pi 3B+ and should make provisioning easier, eliminate the need to open and maintain ports, and can keep traffic between those phones local. It also provides more security as the 3CX tunnel protocol encrypts calls going to and from the internet.
 
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