Solved Yealink T58W Not Ringing

[Gilroy]

• 3CX Version, 18.0 (Build 450) Standard Annual
• Server OS, Debian 9
• Is the 3CX Server Hosted and where? AWS Lightsail
• IP Phone Make/Model/Firmware version, Yealink T58W 150.86.0.8
• Provisioning Method: STUN
• Trunk Provider or VoIP Gateway Make/Mode: Symbio
• Has the Firewall Checker passed: YES
• Are custom Phone Templates being used: No

Hi 3CX Community!

We're experiencing a strange issue after I've provisioned three new phones on the site. So all phones are working except one which is Yealink T58W. It does not ring when called both from internally or externally however, it can call an extension and establish two-way audio. When calling this Yealink phone, it stuck at Dialing without any beep audio for a few seconds and finally going to the extension's voicemail message.

All the phones were provisioned using the convenient auto-provisioning link inserted on the Yealink phone and started working except this one.
Here is the configuration for the T58W on the 3CX management console


I found a thread that has the same issue and followed it through, in his/her case, the solution is port forwarding the SIP and RTP ports to the Phone's Local IP. We then applied the port forward rule:

• port 5073 UDP/TCP to 192.168.54.X
• port 14160 - 14179 UDP to 192.168.54.X

However, the issue still persists even after reprovisioning and rebooting the phone. Re-created an extension just to make sure, however, the issue still occurred.
I tried replicating the issue to capture the call flow sequence and found the result below.


From what it looks like it was unable to get further information after sending an INVITE SDP.
I've run out of ideas about what to do next and hoping you could help me.

Kindly see the Activity log that I've got below.
NOTE:

• EXT 100 - Support Test Extension
• EXT 152 - Yealink T58W
• Replaced some information such as the actual IP and FQDN

03/09/2022 7:33:01 PM - Route to L:22.2[Extn:152] sends Invite-OUT Send Req INVITE from 0.0.0.0:0 tid=e15d750bbc1ff40d Call-ID=hQoAj3DKT1kMgvSAXO2h3A..:
INVITE sip:[email protected]:5073 SIP/2.0
Via: SIP/2.0/ ;branch=z9hG4bK-524287-1---e15d750bbc1ff40d;rport
Max-Forwards: 70
Contact: <sip:[email protected]:5060>
To: <sip:[email protected]>
From: "Support"<sip:[email protected]:5060>;tag=1252254d
Call-ID: hQoAj3DKT1kMgvSAXO2h3A..
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE
Content-Type: application/sdp
Supported: replaces, timer
Content-Length: 445

v=0
o=3cxPS 35426011562639360 1405170776801281 IN IP4 54.6X.XXX.XXX
s=3cxPS Audio call
c=IN IP4 54.6X.XXX.XXX
t=0 0
m=audio 9042 RTP/AVP 18 9 8 0 112 102 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:112 opus/48000/2
a=fmtp:112 maxplaybackrate=16000;sprop-maxcapturerate=16000
a=rtpmap:102 telephone-event/48000
a=rtpmap:101 telephone-event/8000
a=sendrecv
03/09/2022 7:33:01 PM - Created new call info (connection): CstaConn{pcid=,caller=100,C:22.2;connst=}
03/09/2022 7:33:01 PM - SLA is globally disabled
03/09/2022 7:33:01 PM - Added leg L:22.2[Extn:152] <-> L:22.1[Extn:100]
03/09/2022 7:33:01 PM - [Flow] Call(C:22): making call from L:22.1[Extn:100] to T:Extn:152@[Dev:sip:[email protected]:5073]
03/09/2022 7:33:01 PM - [CM503027]: Call(C:22): From: Extn:100 ("Support" <sip:[email protected]:0>)  to  T:Extn:152@[Dev:sip:[email protected]:5073]
03/09/2022 7:33:01 PM - [CM503004]: Call(C:22): Route 1: from L:22.1[Extn:100] to T:Extn:152@[Dev:sip:[email protected]:5073]
03/09/2022 7:33:01 PM - [Flow] Endpoint Extn:152 has no forwarding rule on reason 'All calls'
03/09/2022 7:33:01 PM - [Flow] Call(C:22): has built target endpoint: Extn:152 for call from L:22.1[Extn:100]
03/09/2022 7:33:01 PM - [Flow] Target endpoint for 152 is Extn:152
03/09/2022 7:33:01 PM - [Flow] Building target endpoint to 152 from "Support" <sip:[email protected]:0>
03/09/2022 7:33:01 PM - [CM503010]: Call(C:22): Making route(s) from Extn:100 to <sip:[email protected]:0/UDP>
03/09/2022 7:33:01 PM - Remote SDP is set for leg L:22.1[Extn:100]
03/09/2022 7:33:01 PM - [CM505001]: Endpoint Extn:100: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CXPhone for Windows 16.3.0.264] PBX contact: [sip:[email protected]:5060]
03/09/2022 7:33:01 PM - Inbound DID: ''; Phonebook Name: ''
03/09/2022 7:33:01 PM - [CM500002]: Call(C:22): Info on incoming INVITE from Extn:100:
Invite-IN Recv Req INVITE from 127.0.0.1:5080 tid=Pj9ee62ceb77de4889950b2082573a9113 Call-ID=852b5f64ac8848f49bcc0c79709aa831:
INVITE sip:[email protected]:5060;transport=UDP;tag3cx=bbf1829e037a48079bd555de041a9a15 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK-524287-2---Pj9ee62ceb77de4889950b2082573a9113;rport=5080
Via: SIP/2.0/UDP [::ffff:120.29.111.209]:13172;branch=z9hG4bK-524287-1---tunneltid;rport;tnlid=clnt.1-47f9a0d7047c4679a2dc4b79bef99bc4
Via: SIP/2.0/TCP 192.168.100.21:50195;rport;branch=z9hG4bKPj9ee62ceb77de4889950b2082573a9113;alias
Max-Forwards: 69
Record-Route: <sip:[email protected]:5080;user=proxy;uri=clnt.1-47f9a0d7047c4679a2dc4b79bef99bc4>
Contact: "Support" <sip:[email protected]:5060;rinstance=1-47f9a0d7047c4679a2dc4b79bef99bc4>
To: <sip:[email protected];tag3cx=bbf1829e037a48079bd555de041a9a15>
From: "Support"<sip:[email protected]>;tag=0925fffb1c9a412f886f79b469f84864
Call-ID: 852b5f64ac8848f49bcc0c79709aa831
CSeq: 15749 INVITE
Session-Expires: 1800
Min-SE: 90
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Type: application/sdp
Proxy-Authorization: Digest username="q2WmxnT8tY",realm="3CXPhoneSystem",nonce="414d53596228906d04:bc03ae9f85e252b68de541fcb8d6972f",uri="sip:[email protected];transport=tcp;tag3cx=bbf1829e037a48079bd555de041a9a15",response="aa5acb7f1771ba59039345d043286119",algorithm=MD5
Supported: replaces, 100rel, timer, norefersub
User-Agent: 3CXPhone for Windows 16.3.0.264
Content-Length: 463

v=0
o=- 3855843180 3855843180 IN IP4 192.168.100.21
s=pjmedia
b=AS:117
t=0 0
a=X-nat:0
m=audio 10020 RTP/AVP 8 0 9 18 120 96
c=IN IP4 192.168.100.21
b=TIAS:96000
b=AS:117
a=rtcp:10021
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:120 opus/48000/2
a=fmtp:120 useinbandfec=1
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
m=video 0 RTP/AVP 31
c=IN IP4 127.0.0.1
03/09/2022 7:33:01 PM - [CM503001]: Call(C:22): Incoming call from Extn:100 to <sip:[email protected]:0>
03/09/2022 7:33:01 PM - IncomingCall: C:22 from <sip:[email protected]:0/UDP> to <sip:[email protected]:5060/UDP>
03/09/2022 7:33:01 PM - Added leg L:C:22.1[No endpoint yet]
03/09/2022 7:33:01 PM - Call from "Support"<sip:[email protected]>;tag=0925fffb1c9a412f886f79b469f84864 to <sip:[email protected];tag3cx=bbf1829e037a48079bd555de041a9a15>;tag=1fae0c09

Thank you very much in advance!

 

[Gilroy]

Hi @JohnS_3CX,

Thank you for getting back to me. So far I'm pretty sure I've done the following.

3CX Server Configuration:

• Ports opened as per https://www.3cx.com/docs/ports/
• The firewall passed all the test
• Disabled "Disallow use of extension outside the LAN (Remote extensions using Direct SIP or STUN will be blocked)"
• Enabled "PBX Delivers Audio"
• Unique SIP and RTP Ports

Remote Location Configuration:

• The phone has a static IP address
• Port forward the unique SIP and RTP Ports as mentioned in the previous post.
• SIP ALG has been disabled.

Thank you for looking at this issue which I really appreciate.

 

[JohnS_3CX]

Hi Gilroy,

Since you have a capture showing the PBX sending an INVITE to the T58W, I assume that it can at least register (even though incoming calls fail). However, we do not know that the T58W actually received the INVITE. You would also have to run a capture on the phone, and see what the phone received to get the full picture.


Are you able to install an SBC instead of relying on STUN? It would really reduce the need to go all through this trouble for a single phone (and tomorrow it could be another phone, and another and so on).

 

[Gilroy]

Hi @JohnS_3CX!

Thank you for getting back to this thread.

I have an update from the customer that the issue came from their network setup. Basically, they haven't mentioned to us that they added a new router to the network. The problem is the router facing the phones isn't directly connected to the internet, it just gets packets from the other router. I believe this means the port forwarding and such aren't working. They have plugged the T58W into the network that has the router facing the world internet and that resolved the issue. The only odd thing here is T53 and T4 phones are working without issue.

Appreciate your response as always 3CX folks are awesome.

 

[JohnS_3CX]

Glad to hear it was resolved!

As for the others that were working, I think it was just a lucky coincidence. STUN phones can be unpredictable (ie. they may appear to work and then stop, or they may not work at all) when the network environment is not under your control.

An SBC largely solves this issue, and if they had one installed on-site, it would still continue to work even after they had added the new router that broke the forwards, and you wouldn't have had a call from the customer in the first place