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Yealink T4XG phones will not autoprovision over HTTPS with FreePBX 14

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i created a certbot certificate in addition to the certman cert and it works perfectly.

sudo certbot certonly --agree-tos --non-interactive --standalone --email testpbx15@bundystl.com --no-eff-email --preferred-challenges http --domains tpbx15.bundystl.com --pre-hook "systemctl stop httpd" --post-hook "systemctl start httpd"

sudo cp /etc/letsencrypt/live/tpbx15.bundystl.com/chain.pem /etc/asterisk/keys/tpbx15.bundystl.com-ca-bundle.crt
sudo cp /etc/letsencrypt/live/tpbx15.bundystl.com/privkey.pem /etc/asterisk/keys/tpbx15.bundystl.com.key
sudo cp /etc/letsencrypt/live/tpbx15.bundystl.com/cert.pem /etc/asterisk/keys/tpbx15.bundystl.com.crt

sudo fwconwole certificates --import

Then I went into SysAdmin and set the tpbx15 cert active.
image

Then I updated my DHCP and rebooted the phone.

You can clearly see the phone request, get a 401 and then request with the creds and get a 200 response. XXXXXXX = masked username from the Provisioning protocols.


[jbusch@freepbx ~]$ sudo tail /var/log/httpd/access_log -f
64.53.199.99 - - [17/Jan/2020:04:53:00 +0000] "GET /001565649346.boot HTTP/1.1" 401 381 "-" "Yealink SIP-T46G 28.83.0.120 00:15:65:64:93:46"
64.53.199.99 - XXXXXXXX [17/Jan/2020:04:53:00 +0000] "GET /001565649346.boot HTTP/1.1" 404 215 "-" "Yealink SIP-T46G 28.83.0.120 00:15:65:64:93:46"
64.53.199.99 - - [17/Jan/2020:04:53:04 +0000] "GET /y000000000000.boot HTTP/1.1" 401 381 "-" "Yealink SIP-T46G 28.83.0.120 00:15:65:64:93:46"
64.53.199.99 - XXXXXXXX [17/Jan/2020:04:53:04 +0000] "GET /y000000000000.boot HTTP/1.1" 404 216 "-" "Yealink SIP-T46G 28.83.0.120 00:15:65:64:93:46"
64.53.199.99 - - [17/Jan/2020:04:53:08 +0000] "GET /y000000000028.cfg HTTP/1.1" 401 381 "-" "Yealink SIP-T46G 28.83.0.120 00:15:65:64:93:46"
64.53.199.99 - XXXXXXXX [17/Jan/2020:04:53:08 +0000] "GET /y000000000028.cfg HTTP/1.1" 200 7974 "-" "Yealink SIP-T46G 28.83.0.120 00:15:65:64:93:46"


System Recordings not working

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You where advised to restore the directory that you deleted , that meant to install the directory and all its contained files. From bash :-

find /var/lib/asterisk/sounds/ |grep custom

would be helpful to see if you have anything you can use.

(‘ll’ is usually an alias for ‘ls -lah’ , it might not be available on your box, not sure what ‘li’ is an alias for , just use the unaliased ‘ls’ builtin command )

Sip end user account as sip trunk

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This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.

System Recordings not working

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The command find /var/lib/asterisk/sounds/ | grep custom shows the following results:

/var/lib/asterisk/sounds/at-customers-request.g729
/var/lib/asterisk/sounds/customer-service.alaw
/var/lib/asterisk/sounds/customer-service.g722
/var/lib/asterisk/sounds/ja/custom
/var/lib/asterisk/sounds/en/custom
/var/lib/asterisk/sounds/customer-relations.alaw
/var/lib/asterisk/sounds/at-customers-request.ulaw
/var/lib/asterisk/sounds/there-is-no-customer-support.g729
/var/lib/asterisk/sounds/there-is-no-customer-support.g722
/var/lib/asterisk/sounds/customer-relations.g722
/var/lib/asterisk/sounds/lines-complaining-customers.g729
/var/lib/asterisk/sounds/customer-accounts.g729
/var/lib/asterisk/sounds/there-is-no-customer-support.ulaw
/var/lib/asterisk/sounds/there-is-no-customer-support.alaw
/var/lib/asterisk/sounds/en_US/custom
/var/lib/asterisk/sounds/en_GB/custom
/var/lib/asterisk/sounds/customer-relations.ulaw
/var/lib/asterisk/sounds/customer-accounts.alaw
/var/lib/asterisk/sounds/custom
/var/lib/asterisk/sounds/customer-service.ulaw
/var/lib/asterisk/sounds/customer-service.g729
/var/lib/asterisk/sounds/at-customers-request.alaw
/var/lib/asterisk/sounds/customer-accounts.ulaw
/var/lib/asterisk/sounds/lines-complaining-customers.g722
/var/lib/asterisk/sounds/lines-complaining-customers.ulaw
/var/lib/asterisk/sounds/at-customers-request.g722
/var/lib/asterisk/sounds/lines-complaining-customers.alaw
/var/lib/asterisk/sounds/customer-accounts.g722
/var/lib/asterisk/sounds/customer-relations.g729

Time Group Question

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Hello,
yes i enabled BLF in the GUI, everything is working when i manually change the Database entry, so i presume it is a wrong INSERT statement. I will report this at issues.freepbx.org.
Thank you!

Using analogue line for Trunk

Yealink T4XG phones will not autoprovision over HTTPS with FreePBX 14

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So I did not find this until I tested with the certbot cert. in the Yealink’s syslog (level 3 dump), you eventually see this in the log. Looks like it does not trust the X3 cert authority from LE, but as previously shown, it still connects to the web server. Of note, it will not register in this state if you enable TLS for registration in PJSIP.

<131>Jan 16 23:18:50 lbt [1267]: BADT<3+error > Can not find adapter, please plug up.
<131>Jan 16 23:18:55 sua [1286]: NET <3+error > [255] verify error:num=20:unable to get local issuer certificate:depth=0:/CN=tpbx15.bundystl.com
<131>Jan 16 23:18:55 sua [1286]: NET <3+error > [255] X509_V_ERR_UNABLE_TO_GET_ISSUER_CERT_LOCALLY issuer= /C=US/O=Let's Encrypt/CN=Let's Encrypt Authority X3
<131>Jan 16 23:18:55 sua [1286]: NET <3+error > [255] verify error:num=27:certificate not trusted:depth=0:/CN=tpbx15.bundystl.com
<131>Jan 16 23:18:55 sua [1286]: NET <3+error > [255] X509_V_ERR_CERT_UNTRUSTED issuer= /C=US/O=Let's Encrypt/CN=Let's Encrypt Authority X3
<131>Jan 16 23:18:55 sua [1286]: NET <3+error > [255] verify error:num=21:unable to verify the first certificate:depth=0:/CN=tpbx15.bundystl.com
<131>Jan 16 23:18:55 sua [1286]: NET <3+error > [255] X509_V_ERR_UNABLE_TO_VERIFY_LEAF_SIGNATURE issuer= /C=US/O=Let's Encrypt/CN=Let's Encrypt Authority X3
<131>Jan 16 23:18:56 sua [1286]: NET <3+error > [255] Failed to verify remote certificate
<131>Jan 16 23:18:56 GUI [1267:1267]: CUIT<3+error > 336.988.442:check passwd err
<131>Jan 16 23:18:57 GUI [1267:1303]: WIND<3+error > 337.180.857:[DCMN]Recode is 404, Request err.
<131>Jan 16 23:18:57 GUI [1267:1303]: WIND<3+error > 337.192.858:[DCMN]download common error, errcode:404.

But even this failure is not in the syslog for the certman generated certificate. Instead we get an error that the URL is empty. So obviously, something with the certificate is failing to be read, since it has no problem reading an LE cert from the X3 authority.

<131>Jan 17 13:37:49 lbt [1425]: BADT<3+error > Can not find adapter, please plug up.
<131>Jan 17 13:37:55 ATP [1438]: ATP <3+error > Get static config url is empty

Now, as mentioned, the phone pulls the config and other web related stuff but does not register. Wll let’s see if we can make it trust things. First if you disable cert checking it will register. This did not help with the certman cert though, and is not something I would ever recommend as a long term solution.

Turning that setting back on and then uploading the CA bundle into the phone made the error in the phone sys.log go away and let the phone register. Again, this made no difference to the certman cert. Only made the Certbot cert work fully.


You can load it in the config like this. I could have put it in the tftpboot folder also, but I have that folder under version control with git so I didn’t want a cert that changes randomly in there. Instead I symlinked it from keys, (I already have stuff in custom).
sudo -u asterisk ln -s /etc/asterisk/keys/tpbx15.bundystl.com-ca-bundle.crt /var/www/html/custom/tpbx15.bundystl.com-ca-bundle.crt

Then I setup the config file for the phone to have this.

static.trusted_certificates.url = https://tpbx15.bundystl.com/custom/tpbx15.bundystl.com-ca-bundle.crt
static.security.trust_certificates = 1

After the phone reboots and gets the cert, it willreboot again.
But then the sys.log shows this.

<131>Jan 17 00:58:53 lbt [1277]: BADT<3+error > Can not find adapter, please plug up.
<131>Jan 17 00:58:58 sua [1296]: NET <3+error > [255] depth=2:/O=Digital Signature Trust Co./CN=DST Root CA X3
<131>Jan 17 00:58:58 sua [1296]: NET <3+error > [255] depth=1:/C=US/O=Let's Encrypt/CN=Let's Encrypt Authority X3
<131>Jan 17 00:58:58 sua [1296]: NET <3+error > [255] depth=0:/CN=tpbx15.bundystl.com
<131>Jan 17 00:58:59 GUI [1277:1315]: WIND<3+error > 339.658.335:[DCMN]Recode is 404, Request err.
<131>Jan 17 00:58:59 GUI [1277:1315]: WIND<3+error > 339.693.589:[DCMN]download common error, errcode:404.
<131>Jan 17 00:59:00 GUI [1277:1277]: CUIT<3+error > 340.061.047:check passwd err

Adjusting TX/RX volume on specific Trunk

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No… But, if you have a volume issue with an analog trunk using an FXO adapter, there is likely a problem with your impedence settings. Some FXOs have a testing mode that will automatically set the impedence correctly. The Grandstream GXW-410X series does.

Others don’t. If your doesn’t, you’ll either have to have someone test the impedence, or you’ll have to just try each setting until you find the one that works best for you.


Unable to receive calls

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Need way more information. Logs would help. Could be:
1 Trunk configured incorrectly.
2. Asterisk SIP Setings configured incorrectly.
3. Inbound route configured incorrectly.
4. Firewall configured incorrectly.
5. Service provider issue.

Does anyone know if we have a troubleshooting flowchart in the Wiki? If not, that’s something that should exist. :slight_smile:

Changing vm-login file from "Comedian Mail, Mailbox" to "Mailbox"

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It used to be that you could just download the replacement file and use scp or winscp to copy it over.

I fell in love with the Pat Fleet voice prompts many years ago. I grew up listening to her whenever I called time or dialed a wrong number. The prompts take care of the “Comedian Mail” issue, which I cannot believe is still a thing with Asterisk today.

Regrettably, there are now some new files (mostly added by FreePBX) that you’ll hear Allison’s voice on, but it’s rare.

You can download the Pat Fleet files here:

Disabling the Firewall

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This topic was automatically closed 31 days after the last reply. New replies are no longer allowed.

Where are queue and ring group

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This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.

System Recordings not working

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all your ./custom directories are empty. There are no files to play,

Yealink T4XG phones will not autoprovision over HTTPS with FreePBX 14

Adjusting TX/RX volume on specific Trunk

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if using dahdi, first do the fxotune thing then balance the hybrid for echo reduction with dahdi_monitor and a call.file calling a local milliwatt test number and connecting it to app_milliwatt, while adjusting RX/TX settings ( requires dahdi restarting)


Using analogue line for Trunk

System Recordings not working

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Would this stop the module from loading on the GUI?

Changing vm-login file from "Comedian Mail, Mailbox" to "Mailbox"

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another solution using what’s already available

cp   vm-login,wav vm-login-saved.wav
sox vm-login-saved,.wav vm-login.wav trim 1.2

you can then use asterisk’s built in

file convert   en/vm-login.wav en/vm-login.gsm (g722 etc.) 

for the other formats needed

System Recordings not working

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what module di you believe not being loaded? Asterisk certainly can’t play a non existent file though :slight_smile:

the “entries” in system recording module are just that, mysql pointers to a file with a name set in language/custom it has no knowledge if you have deleted the referenced file.

Using analogue line for Trunk

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they work well enough, keep the ADSL filter in line on the line port though. You can set it up so your old phone plugged into the phone port works in pass-through fashion

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