Calls don't ring to remote Yealink extension

FreePBX Endpoints
1

We create a backup of an hosted version of freepbx instance and created a new install of freepbx 16 in another data center and restored from backup. It’s mostly used for paging schedules and they paging phones all registered fine. But we had one Yealink dect phone on a W60b sip dect base at one site that I couldn’t log into to update sip server so I factory reset it. I got it registered again without much problem. But calls won’t ring to it and I can’t make outbound calls. Outbound just gives a beeping / busy signal. Inbound calls just go to VM which is the unavailable destination. I had also registered a new cordless phone at the same time.
I had also updated the firmware on the yealink base. I ended up rolling back to the old firmware and then factory resetting it and I’m still not getting the cordless to ring. It’s set to go to an IVR and if you press 0 it rings the extension, but it just ends up at voicemail, which is the failover destination.

I turned on logging on the router where the remote phone is and I’m not seeing traffic hitting the phone like I would expect.
Here’s the sip debug log for extension 161

<— SIP read from UDP:64.224.83.XX:5060 —>

<------------->
share*CLI> sip set debug peer 161
SIP Debugging Enabled for IP: 64.224.83.XX
Reliably Transmitting (NAT) to 64.224.83.XX:5060:
OPTIONS sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 162.244.177.XXX:5060;branch=z9hG4bK34db5b17;rport
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as696fcbbd
To: sip:[email protected]:5060
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-16.0.40.11(18.20.2)
Date: Wed, 26 Feb 2025 18:43:43 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:64.224.83.XX:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 162.244.177.XXX:5060;branch=z9hG4bK34db5b17;rport=5060
From: “Unknown” sip:[email protected];tag=as696fcbbd
To: sip:[email protected]:5060;tag=2547525447
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: Yealink W60B 77.85.0.25
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]:5060’ Method: OPTIONS

Here the extension shows registered under Asterisk Info:
161/161 64.224.83.XX D Yes Yes A 5060 OK (11 ms)

I have restarted the freepbx.
I have STUN server enable on the phone base.

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What am I missing? I’ve spent all day on this :frowning:

2

The log shows no failures and no calls. Please provide a log with a failing call.

Also, chan_sip is no longer supported by the official project.

3

That’s the only log I get when I made a call.
I did some more digging. I changed the inbound route to ring Ext 161 directly instead of the IVR and hit apply config. Called in again and I still hit the IVR greeting! I restarted the pbx and I’m still getting the IVR greeting, even though it’s still set to ring the Ext 161.

I just ran fwconsole ma upgradeall and yum update and everything was showing up to date.

4

Chansip has been working fine on other deployments, but I went ahead and created a new pjsip for extension for testing. But the yealink fails to register, it shows wrong password. I copied and pasted it, I changed the password, but it still says wrong password…

[2025-02-26 16:49:43] NOTICE[2375]: chan_sip.c:29058 handle_request_register: Registration from ‘“GG Demo” sip:[email protected]:5060’ failed for ‘64.224.83.XX:5060’ - Wrong password
Scheduling destruction of SIP dialog ‘[email protected]’ in 32000 ms (Method: REGISTER)
Really destroying SIP dialog ‘[email protected]’ Method: REGISTER

5

I changed the new test extension from pjsip to chansip and it promptly registered.
I changed incoming route to the new extension, and I still get the IVR greeting on incoming calls.

6

You are registering to chan_sip.

7

That log is the result of Asterisk, periodically testing connectivity.

Your fault lies outside Asterisk. Either the request is being sent to the wrong address, or is being blocked by a firewall.

1 Like
8

You were right on the money. I had taken over the project after someone else had started it, and I understood that the trunks were cut over to the new IP address. After I was making changes and still kept hitting the IVR, I wondered if that was the problem. Sure enough, it was. Updated the ip and the phone now rings.

1 Like
9

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