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Autoprovisionong T4xS v83 error

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Hello,

does anyone know if there was any change in auto provisioning for firmware version 66.83.x.xx for T4xS phones, because it stops working after upgrading.
I searched documentation and forum, but couldn't find anything.


I have phones T42S, T46S and T48S.

In my y0000000000xx.cfg files I tell them to upgrade firmware:
firmware.url = tftp://172.16.xxx.xxx/T4xS-66.83.0.10.rom

It all worked fine, but after upgrading to 66.83.0.10 it stopped working for all phones.
On phone I get message: "Unable to complete. Will try again now!"
On TFTP server I get message: "TIMEOUT waiting for Ack block #1"

I tried to upgrade to 83.0.10, 83.0.20, 83.0.35 and 84.0.10. It all works from v82, but after upgrade to any of these firmwares, it stops working.

I can still do manual firmware upgrade and downgrade via web browser, but it doesn't work via TFTP autoprovisioning any more.

Does anyone else have this problem?
Any help would be appreciated.


Regards, Gasper

OpenVPN tunnel disconnects frequently

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Hi team,

we are having an issue with T4x phones on V82 with OpenVPN.

We are using the pfSense in-built OpenVPN server and the Client Export wizard T38(2) option. VPN tarball loads fine and phones connect and work fine.
Sites with just one T4x phone, the tunnel is stable over weeks.
Sites with multiple T4x phones often re-establish the tunnel, in general under 24 hours. Randomly this re-connect fails all together and the tunnel is down. Only way to fix this is to restart the phone.
We have run syslog with verb 6, however I can't see anything that indicates the problem.

Has anyone come across this issue?

OpenVPN config is as per Yealink whitepaper.
Quote:dev ovpns3
verb 6
dev-type tun
tun-ipv6
dev-node /dev/tun3
writepid /var/run/openvpn_server3.pid
#user nobody
#group nobody
script-security 3
daemon
keepalive 10 60
ping-timer-rem
persist-tun
persist-key
proto udp
cipher BF-CBC
auth SHA1
up /usr/local/sbin/ovpn-linkup
down /usr/local/sbin/ovpn-linkdown
local 203.174.129.186
tls-server
server 192.168.55.0 255.255.255.0
client-config-dir /var/etc/openvpn-csc/server3
tls-verify "/usr/local/sbin/ovpn_auth_verify tls 'TQMS-OpenVPNServer' 1"
lport 4194
management /var/etc/openvpn/server3.sock unix
max-clients 500
push "route 192.168.254.0 255.255.255.0"
push "dhcp-option DOMAIN tqms.net.au"
ca /var/etc/openvpn/server3.ca
cert /var/etc/openvpn/server3.cert
key /var/etc/openvpn/server3.key
dh /etc/dh-parameters.1024
persist-remote-ip
float
topology net30
push "route 192.168.11.0 255.255.255.0"

Thanks for any pointers that let us fix this.

license of PJSIP

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Does any body concern about the legality risk of PJSIP usage of Yealink phones?
I found it is listed in the link of Yealink official web https://www.yealink.com/onepage_83.html
It is licensed under GPLv2.
PJSIP is a library as described in it official web
https://www.pjsip.org/
Does this mean Yealink does not buy commercial license and they should publish all their linked source code which I cannot find on their official web.

BR

W60B Shared Directory

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How frequently does the shared directory file get read and cached by the base station? Is this configutable?

Headset Prior (Yealink T21P)

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Hello,

I'm having a huge issue at one of my offices; we have three Yealink T21P phones with Jabra headsets (these work great).

We purchased a Jabra remote handset lifter in order to be able to answer phone calls without having to push the "headset" button on the phone itself. I made sure to set "headset prior" to "true" in the web interface (of each phone); I can see that it's working because the icon changes from a phone to a headset on the phone's display (showing headset prior is active).

Everything is working, except when the lifter picks up the handset, it auto switches to the handset and you still have to push "headset" on the phone itself in order to talk via the headset (even if headset prior is active).

Am I doing something wrong? Can this issue be fixed?

Thank you in advance,
Ray.

Click to dial from web browser

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Hi,

I need help with setting up click to dial from Chrome browser. Found few extensions but with not good explanation how to setup, or at least not noob friendly guide. I've set " Action URI Allow IP List" to "any". Tried different things with different extensions but with no luck. Using T27G with latest firmware.

Anyone who can help with this?

Thank you!

SIP-T27G no Refresh Display BLF Monitoring When Receiving subsequence Notify

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The Yealink IP Phone SIP-T27G does not refresh the Device Display when receiving most recently NOTIFY messages, on BLF feature.

Following show packet capture:

NOTIFY sip:5991@phoneipaddr:5060 SIP/2.0 <------ First Notify received
v: SIP/2.0/UDP asteriskserveripaddr:5060;branch=z9hG4bK3ceecd02
Max-Forwards: 70
f: <sip:202@asteriskserveripaddr>;tag=as75cd87a9
t: "User Name 2" <sip:5991@asteriskserveripaddr:5060>;tag=190612704
m: <sip:202@asteriskserveripaddr:5060>
i: 0_472292147@phoneipaddr
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX
Subscription-State: active
o: dialog
c: application/dialog-info+xml
l: 558

<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="1" state="full" entity="sip:202@asteriskserveripaddr">
<dialog id="202" call-id="pickup-0_472292147@phoneipaddr" local-tag="190612704" remote-tag="as75cd87a9" direction="recipient">
<remote>
<identity display="202">sip:202@asteriskserveripaddr</identity>
<target uri="sip:202@asteriskserveripaddr"/>
</remote>
<local>
<identity display="202">sip:202@asteriskserveripaddr</identity>
<target uri="sip:202@asteriskserveripaddr"/>
</local>
<state>early</state>
</dialog>
</dialog-info>


SIP/2.0 200 OK
Via: SIP/2.0/UDP asteriskserveripaddr:5060;branch=z9hG4bK3ceecd02
From: <sip:202@asteriskserveripaddr>;tag=as75cd87a9
To: "User Name 2" <sip:5991@asteriskserveripaddr:5060>;tag=190612704
Call-ID: 0_472292147@phoneipaddr
CSeq: 103 NOTIFY
Contact: <sip:5991@phoneipaddr:5060>
User-Agent: Yealink SIP-T27G 69.84.0.10
Content-Length: 0


NOTIFY sip:5991@phoneipaddr:5060 SIP/2.0 <------ Second Notify received refresh one previous
v: SIP/2.0/UDP asteriskserveripaddr:5060;branch=z9hG4bK7cd1503c
Max-Forwards: 70
f: <sip:202@asteriskserveripaddr>;tag=as75cd87a9
t: "User Name 2" <sip:5991@asteriskserveripaddr:5060>;tag=190612704
m: <sip:202@asteriskserveripaddr:5060>
i: 0_472292147@phoneipaddr
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX
Subscription-State: active
o: dialog
c: application/dialog-info+xml
l: 572

<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="2" state="full" entity="sip:202@asteriskserveripaddr">
<dialog id="202" call-id="pickup-0_472292147@phoneipaddr" local-tag="190612704" remote-tag="as75cd87a9" direction="recipient">
<remote>
<identity display="User Name Extern">sip:22225555@asteriskserveripaddr</identity>
<target uri="sip:22225555@asteriskserveripaddr"/>
</remote>
<local>
<identity display="User Name 1">sip:5988@asteriskserveripaddr</identity>
<target uri="sip:5988@asteriskserveripaddr"/>
</local>
<state>early</state>
</dialog>
</dialog-info>

Code:
[b]<----  In this point, when receiving, a IPhone1's visual screen on outhers devices of
<----  monitoring group must display visual screen 5988 <- 22225555, but it is not occur,
<----  but instead take place 202 <- 22225555[/b]

SIP/2.0 200 OK
Via: SIP/2.0/UDP asteriskserveripaddr:5060;branch=z9hG4bK7cd1503c
From: <sip:202@asteriskserveripaddr>;tag=as75cd87a9
To: "User Name 2" <sip:5991@asteriskserveripaddr:5060>;tag=190612704
Call-ID: 0_472292147@phoneipaddr
CSeq: 104 NOTIFY
Contact: <sip:5991@phoneipaddr:5060>
User-Agent: Yealink SIP-T27G 69.84.0.10
Content-Length: 0


NOTIFY sip:5991@phoneipaddr:5060 SIP/2.0 <------ last Notify received refresh state=terminated
v: SIP/2.0/UDP asteriskserveripaddr:5060;branch=z9hG4bK4ece7870
Max-Forwards: 70
f: <sip:202@asteriskserveripaddr>;tag=as75cd87a9
t: "User Name 2" <sip:5991@asteriskserveripaddr:5060>;tag=190612704
m: <sip:202@asteriskserveripaddr:5060>
i: 0_472292147@phoneipaddr
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX
Subscription-State: active
o: dialog
c: application/dialog-info+xml
l: 210

<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="3" state="full" entity="sip:202@asteriskserveripaddr">
<dialog id="202">
<state>terminated</state>
</dialog>
</dialog-info>


SIP/2.0 200 OK
Via: SIP/2.0/UDP asteriskserveripaddr:5060;branch=z9hG4bK4ece7870
From: <sip:202@asteriskserveripaddr>;tag=as75cd87a9
To: "User Name 2" <sip:5991@asteriskserveripaddr:5060>;tag=190612704
Call-ID: 0_472292147@phoneipaddr
CSeq: 105 NOTIFY
Contact: <sip:5991@phoneipaddr:5060>
User-Agent: Yealink SIP-T27G 69.84.0.10
Content-Length: 0



Yealink's software engineering should correct this behavior, so that the device displays the correct group user information, as soon as possible.

Voice Mail displayed on T48S?

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Is it possible to see a list of all messages on a voice mailbox and select/delete/listen to individual calls on the voice mailbox?
Instead of listening to the voice of the mailbox and select the messages on the box by using cryptic number keys??

Thanks for help.

Dutch translation

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Good morning,

I have 14 Yealink T48S devices with the 66.81.x.x firmware, but I want to upgrade to the latest. The importer ( Lydis ) here in the Netherlands only has the 66.81.x.x translated. The latest seems to be the 66.84.x.x.

The next firmware they translated will be the 66.83.x.x, and not yet released.

Now I have 2 devices upgraded perfectly, and running smooth. I used the provisioning language file to make our devices Dutch again ( work in progress, still some missing translation).

Anyway, I have no problem letting our intern do some translation work and sharing it on bitbucket. But; is that allowed by Yealink? Or is the only one that may do the translation and share them Lydis?

What is the correct procedure to report a BUG?

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I believe I've found a bug what is the correct procedure for reporting this and what attachments/logs are best to include?

BUG - inbound hold/unhold/hangup not work via remote control for some accounts

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Via remote control I am experiencing a bug whereby the phone will NOT hangup/hold/unhold an INBOUND call for an account connected to a 3CX voip system.

Yet for another voip provider (Gradwell) remote control works fine.
And for the 3CX account it will hangup/hold/unhold an OUTBOUND call via remote control.
And for the 3CX account it will Answer an inbound call via remote control.

We are seeing this bug on xx.83.0.35, xx.83.0.50 and xx.84.0.10
We are seeing this bug on T46S and T23

Please advise how to get this resolved and what logs are needed.

Remote phonebook - T56A - Failed to download

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I used the following template for testing remote phonebook
-----------------------------------------
<YealinkIPPhoneDirectory>
<DirectoryEntry>
<Name>Test1</Name>
<Telephone>23000</Telephone>
</DirectoryEntry>
<DirectoryEntry>
<Name>Test2</Name>
<Telephone>303</Telephone>
<Telephone>915980830849</Telephone>
</DirectoryEntry>

<DirectoryEntry>
<Name>Test3</Name>
<Telephone>6650</Telephone>
<Telephone>915980830849</Telephone>
</DirectoryEntry>

</YealinkIPPhoneDirectory>
------------------------
and saved it in my PC which is connected to same LAN as T56A phone.
IP address of
T56A 192.168.1.13
My PC 192.168.1.14

I saved the template "Department.xml" at two locations C:/Department.xml and D:/Department.xml

i tried both remote URL
http://192.168.1.14:8080/Department.xml
and
http://192.168.1.14/D/Department.xml
to download the test contact to T56A phone.
It shows error " Fail to download remote phonebook".

Can someone help please?

Provisioning multiple servers - tips

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You make the Provision Job, you can utilize the Import Device highlight to relate a produced PC name with every gadget's MAC address when you import the gadgets to the BMC Server Automation Console. For data about how to do this, see Importing a rundown of gadgets. When you select those gadgets for provisioning on the Provision Job Devices board, you see a parameterized property estimation like ??LOCAL_COMPUTER_NAME?? in the Computer Name field.

In the event that this parameterized property estimation results in novel PC names for all gadgets, the wizard leaves the names as determined. In any case do my homework online, if the wizard discovers copy names, it consequently attaches succession numbers to the copy names to make them special (NAME_1, NAME_2, et cetera).

On the off chance that the parameterized property estimation results in a few gadgets not having names, you can give names on the Provision Job Devices board.

W56H Wishlist

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We actually use Gigaset VoIP DECT phones - the config is really great including windows client to access the mobile over USB and upload own .wav ringtones.
The ringtones than can be configured to the sip accounts (if an incoming call to account 1 is coming, play ringtone A, if to account 2 play ringtone B).
This all work as ist should... the only (but important) thing is, that support for OPUS is missing...
Therefore we have bought a W60B with two W56H.
I no had to note, that it seems not to be possible to upload own ringtones Angry although the "wired" models, link the T46S (that we also use) support it fully.
We use the T47S in parallel to the DECT and want to use the same (custom) ringtone for one of the lines.
Please add as sonn as possible the ability to uploade own ringtones (.wav) to the W60B /W56H.

Question:
- Is there really now way (some "hack") to load an own ringtone?

Thanks for a feedback
Fredy

BTOE usb driver init failed

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Seen this error else where but no solution.

When trying to pair the T42S with SFB, this fails with USB driver init fail.
Try to install driver which shows successful but still gives same error when trying to pair again.

USB call recording on T58v

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Hello,

I need help by setting up the T58v for automatic USB call recording for in- and outgoing calls.

Right now I have to push each time a sofwarebutton on the phone to record a call :-(

Thanks

3CX v15 + Yealink T4x - Customer Complaining about Call Quality

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We've just rolled out a 3CX v15 for Linux install for a client (16 license, professional), with Patton SN4114/JO trunked to 4x PSTN lines. They are not using VoIP, however they have Yealink T42G, Yealink T46G and Yealink T48G handsets.

The customer has advised that their previous NEC 'digital' phone system was better than the new system we've put in. Apparently they feel the clarity of calls was better. They are hearing crackling on the call.

Originally we had to bump up the output gain on the Patton to around 7 db, otherwise their customers could not hear them.

Can anyone suggest things we can do to improve call quality? It's rather disheartening to hear a customer who you've just sold 30x handsets tell you they don't think it's as good as the piece of junk you pulled out.

Dial from Windows

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I wish to program my MS Access client database system to dial out with a button, either directly using and API or by shelling a command line containing the phone number. Is this possible?
We use Access 2017 and Win7.
Thanks for any help!

T42S/G Recovery Firmware

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I have the recovery firmware for our T42G and T42S phones. I have recovered many phones in the past. Currently I have several phones that will update the firmware with TFTP, but when the phone reboots it stays at Initializing. I do get the message that says firmware update complete before reboot.

custom ringtone with new BLF feature

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Chaps,
this is related to the feature request for BLF audio alert I raised as a feature request that was fulfilled in firmware 66.84.0.10 - it works really well, BUT.

I;ve tried to attach a custom ring tone (actually a doorbell sound) and this does not work. The wav file for the custom ring works as a direct ring on on another phone at a lower software, so I think my wav file is ok.
This works ok with Ring8.wav
features.pickup.blf_audio_enable = 1
features.pickup.blf_visual.list =
features.pickup.blf_audio.list = 2001

features.blf.ring_type = Ring8.wav

But when i substitute the custom ring file
fetures.blf.ring_type = Doorbell3.wav

No sound emerges - I;ve checked the phone to see that the custom file is loaded.

I have a feeling it should be
fetures.blf.ring_type = XXXXXX: Doorbell3.wav
but cant find anything in the admin guide.
chris
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