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Transfer calls T26P

September 13, 2018, 6:42 pm
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Hi,

I have three T26P and sometimes transfer calls don't working. I make captures in the pbx(asterisk), and i see the invite packet from the phone but don't send the packet to the pbx.

Someone with the same problem?
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Yealink RPS upgrade

September 13, 2018, 11:06 pm
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Dear Yealink Customer,

This Email is very important, Please pay your attention.
Please also informed all your partner/reseller.
[Scheduled Maintenance]
Please be informed that Yealink RPS (Redirection & Provisioning Service) system will be taken offline for a two-hour scheduled maintenance on September 14th, 2018. The maintenance involves system upgrade, data migration and security enhancements on RPS core servers to improve overall service reliability. Meanwhile, new domain names will be used for RPS web portal and API access, as an effort to streamline Yealink’s cloud services. Details as follows.

Date & Time
Friday, September 14th, 2018
- 00:00 – 02:00 GMT/UTC +0
- 01:00 – 03:00 CET/UTC +1
- 20:00 – 22:00 EDT/UTC -4, September 13th
- 17:00 – 19:00 PDT/UTC -7, September 13th

System/Service Affected
Downtime: Two hours (as specified above)
During the downtime, RPS servers will be taken offline and all RPS services will be unavailable.

Important Notes
1. After the maintenance, the access URL of RPS web portal and API domain name will be changed to:
- Web portal: https://dm.yealink.com
- RPS API: https://api-dm.yealink.com:8443/xmlrpc
Please modify the server if integration with RPS API.
NOTE： The Domain name for phone to request is not changed.

2. To avoid any service disruption, please whitelist the following domains and/or IP addresses prior to the maintenance:
https://dm.yealink.com
https://api-dm.yealink.com
https://rps.yealink.com
52.71.103.102
35.156.148.166
106.15.89.161
47.75.58.202
47.89.187.0
If you have any concern or question regarding this maintenance, please create a ticket at https://ticket.yealink.com
Thank you for your attention.

BR
Yealink technical team
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Pricing model

September 14, 2018, 2:36 am
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So the YDMP (which we are still testing) says pricing will be released in January or that it will no longer be free. What will that pricing be exactly? We'd like to prepare for such a ramp-up in cost if possible.
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rps.yealink.com has been down since last week

September 17, 2018, 12:44 am
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Delete handset button missed with FW 77.83.0.10

September 17, 2018, 4:33 pm
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Dear Support,
I have registered 8 handset W56H on W60 with new FW 77.83.0.10.
As you ca see in the attached screen shot , the "Delete_handset_button" is missed, in the web interface menù Status >> Handset&VoIP!!!
Why???
Is this a new specification or a bug??
Best regards
Walter

.jpg  Delete_handset_button_missed.JPG (Size: 32.92 KB / Downloads: 0)
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Yealink Dialplan

September 19, 2018, 2:10 am
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Hi Yealink,

In my dialplan I've
dialplan.digitmap.interdigit_short_timer = 4
[1-8]xx.T

If I dial 514 after 4 seconds the timer end and dial automatically. Can we have a way to set them with no timer or the only way is to increase the short timer.

Aastra phone example

[1-8]xx# no timer they wait until pounds key is press ?
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Using a department phonebook managed bij phones

September 19, 2018, 7:37 pm
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Hello we are currently testing multiple phones because our company has multiple departments who need to maintain their own department phonebooks.

What the users need is that they can add the number of the callee to an phonebook which is used for the whole department.

Now the users are using an old fashioned phone which wirelessly transfers every contact update to the connected phones but logically these are limiting the users in other ways.

With the new V84 firmwares we are able to connect an Google phonebook, and this works great! Downside is that the users are unable to add contacts from their phone. I know they can add it through the google website, but that involves multiple steps the users don't have to do in the current situation.

Is there any way we can edit an shared phonebook through the handset itself?

The phones we are testing now are the T48S and T58V phones.

Regards,
Johan de Zwaan
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Transfer with T2XG and call back

September 19, 2018, 10:59 pm
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Hi, I have several T2XG or P connected with Asterisk. For transfer I'm using TRANFER + EXT or directly press key from exp.

Now I need to use that when I transfer a call if the destination doesn't answer, the calls back to the original extension.

With Asterisk there are a feature code to use it, I use # + ext.

This code works If I press # + ext, but if a press # + key(blf in my exp), it doesn't works.

How can I use it? Or are there any other way to do that?

Thanks
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T41P Multiple 'static' forwarders via DSSKEY

September 19, 2018, 11:20 pm
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Hey there,

situation is as followed:
Company has 3 employees that rotate the after hours shift.
A previous admin had it set up so that on the phone, a DSSKey could be pressed, and all calls would be forwarded to the mobile nr. that was on call for that day.

After we took over the IT for the company, the forwarding options were no longer present on the phone (maybe deleted?).

The customer wants this same situation back.

i've tried
Type = Forward
Value = Mobile number
Label = The Dude

This seemingly does nothing

i've tried
Type = XML Browser
Value = path to xml
Label = The Dude

XML:
Code:
<YealinkIPPhoneExecute Beep="Yes">
<ExecuteItem URI="Key:HANDFREE"/>
<ExecuteItem URI="Dial:*72#mobilenr#" interrupetcall="no"/>
</YealinkIPPhoneExecute>

I found somewhere that *72#nr# is used for Yealink, even though my country uses *21*nr#
When trying to manually forward with *21*nr# on the phone, this indeed does nothing (or isn't recognized).


Work arounds via Always Forward on the portal page of the phone is a no-go.
The customer wants to use the DSSKey to forward to 1 of 3 people via 3 buttons.
A 4th button can be used to remove the forward.

Further info:
There is no dial plan set up
The firmware version = 36.80.188.11
Have tried using Forward and Transfer in DSSKey types. These do not provide a static forward.


Please let me know any info you require to help me work towards the situation the customer desires.
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Issues with variable substitution on action URLs

September 20, 2018, 12:18 am
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I'm not sure if I'm trying to use $active_user wrong, but here's what I have in my config file:

Code:
action_url.registered = http://sipserver/test.php?ext=$active_user&ip=$ip
action_url.on_hook = http://sipserver/test.php?ext=$active_user&ip=$ip
action_url.call_terminated = http://sipserver/test.php?ext=$active_user&ip=$ip
action_url.idle_to_busy = http://sipserver/test.php?ext=$active_user&ip=$ip
action_url.busy_to_idle = http://sipserver/test.php?ext=$active_user&ip=$ip
action_url.setup_autop_finish = http://sipserver/test.php?ext=$active_user&ip=$ip

What I want is for the phone to substitute $active_user with the extension number, so for example if I'm in extension 24 I want these actions to get the URL "http://sipserver/test.php?ext=24&ip=10.10.95.201"

I've found that sometimes $active_user is replaced with nothing instead of the extension number, but the $ip substitution is always correctly replaced with the IP address of the phone:

action_url.registered: $active_user and $ip work
action_url.on_hook: only $ip works
action_url.call_terminated = $active_user and $ip work
action_url.idle_to_busy = $active_user and $ip work
action_url.busy_to_idle = only $ip works
action_url.setup_autop_finish = only $ip works

Is this a bug or is there a more correct way that I should be getting the extension number? I'm testing on a T46G running 28.83.0.57.

Thank you!
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Problem after update firmware t46g

September 20, 2018, 6:35 pm
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Hello

i make update last firmware T46G change password but now i can make login with a new one appears these error :

You are not authorized to access the web interface.

Please contact your support team or try again 3 minutes later.

HTTP 403



Best Regards,

Paulo
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DTMF for carrier pre-selection

September 21, 2018, 4:16 am
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I apologise for using the wrong terminology but not sure how exactly to search for this problem:
I bought a T21P E2, which I have connected to a fritz.box 5490 together with a few more DECT phones.
On the DECT phones if I call "*121#" this allows me do VOIP provider pre-selection on the fritz.box and receive the dial tone to then being able to enter the number going out through that provider.
I'm trying to use that in the IP phone (ultimately to associate with a DSS key), but can't get it working.
When I send "*121#", I get immediately "user busy" and it seems to me as if that particular code is sent straight-through to the VOIP provider, since if I send "*121#1234566" it works as expected calling 1234566 using the selected provider.
Does anyone know what settings would allow me to have the DSS key working to do this pre-selection, and receive the dial tone to enter phone number as a second step ?

Thanks
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T46S + EHS36 + Jabra 920 = Long delay

September 23, 2018, 9:33 pm
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Hi folks,

we are using a combination of T46S, EHS36 and Jabra 920. Everything is working ok, but there is a long delay (3-5 seconds) to end a call by pressing the designated switch on the Jabra 920. We almost set the call control mode fix to DHSG but that didn't changed anything. With an old Alcatel telephone the same headset is running very well.

So do you have any ideas to get rid of this annoying delay?

T46S Firmware Version: 66.83.0.35
EHS36 Hardware Version: 48.16.0.0
EHS36 Firmware Version: 4.16.0.0
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T58 voice strange dropouts

September 23, 2018, 11:19 pm
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I have a strange problem which occurs most of the time with my T58 phones:

During the call when I am talking (using the handset), and the remote party starts talking while I am still talking, about the first letter the remote party speaks is dropped. In the other direction there a no such problems. My partner always hears everything correct.
I also have a W60B set. With this phone, there is no such problem.

I already tried tweaking settings like echo cancellation, noise suppression. No changes.

Is this something common? Where could I look for a solution? Firmware and Phone system is current: 3cx 15.5sp6, phone 58.80.0.65

TIA
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Changing the certificates

September 24, 2018, 1:36 am
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I am trying to replace the default SSL certificate with a valid cert for our organization. I see that there are two keystores in the YDMP Tomcat folder: dm,jks and temp.jks.

dm.jks works as expected with the typical warnings, temp.jks will not work at all and I can't even add an exception in Firefox to allow me to use it. Since I would prefer to not have any certificate warnings at all I would like to replace the cert with a wildcard for our domain.

When I try and follow the instructions in the manual:

Openssl pkcs12 –export –in dm.drt –inkey dm.key –out dm.p12 –name dm

I cannot find the files dm.drt and dm.key on my system.

If I build a brand new keystore using our wildcard certificate Tomcat will not load afterwards.

If I add our wildcard cert to the existing keystore and then activate it I get a error about the server not having any protocols to communicate with which tells me that simply replacing the localhost alias cert in the keystore with my own either missed a step or its incompatible with the current tomcat subsystem.

Any guidance would be appreciated.
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Let’s add FEC and DTx to Opus!

September 24, 2018, 3:14 am
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Adding the opus codec was an excellent feature, but now time to take it to the next level, with FEC and DTx.

What is FEC?

Glad you asked! FEC, or Forward Error Correction, allows the RTP (audio) to contain a partial “backup” of previously sent audio. With this, in the event of packet loss on the network, the phone (decoder) is able to reconstruct any lost audio.

With this feature, a phone call could withstand up to 30% packet loss and you’d hear the other person clear as a bell.

DTx saves bandwidth by sending null audio packets during silence.
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Help DialPlan

September 24, 2018, 4:32 am
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How to configure, so that the phone does not apply dialplan from the call history?

Or

How to make the phone save the original numbers to the history, not the numbers modified by the dialplan.



Firmware Version 44.84.0.10
Model: T23G
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Wav.Play playback on handset

September 24, 2018, 8:37 am
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So I have been able to push an XML file to a T46 phone to playback a wav file. The problem I'm having is that the wav file is always played back over the phone speaker, I want to have the option to playback over the speaker or handset.

If I set LockIn to yes on my XML app then I can pick up the handset while the wav file is being played back and I will hear the wav file in the handset, which is what I want. But if I play another wav file while the handset is still picked up it plays through the speaker again.

Is there a way make this work the way I want it to? The only option I've been able to come up with is to have the phone make a call and playback the file in the call, but that's not a great user experience.
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How to Turn Off Activation Lock on Apple Device?

September 24, 2018, 6:53 pm
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Activation lock keeps your Apple device as well as information safe in case if it gets lost or stolen. However it is always advised to keep it on but still, if you want to turn it off then you have to go through a procedure. You can do so by entering the correct Apple ID and password.
If you erased your device accidently then you will have to enter the credentials to set up the device. Or if you bought the device from others then ask the owner for the activation code. Get Help for Fix to Unlock Apple Device.
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Change disolay name

September 12, 2018, 12:16 am
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Hi,

in our company we use Yealink T41P phones and the phone software Placetel. This automatically sets up the connection to the phones once the Mac address is given. In particular, it sets the VoIP-name as the display name. However, when I change the VoIP-name, the display name does not change. But for other reasons, actually, the display name should be different from the VoIP-name.

So my question is just if it is possible to change the display name of the phones.

Just to be clear what I mean: The display has (in the middle) a clock, underneath it the date, and underneath it the display name of the user.
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