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Verizon Branded T46G repurposed to FreePBX?

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Hello,

Does anybody have experience/knowledge with using a Verizon Branded T46G with FreePBX? There are several for sale right now on eBay but I cannot figure out via googling if they are firmware locked to Verizon.

Do these have a custom firmware? If so, can they be flashed to the normal T46G firmware?

The price is right and the phones are in great shape, but If I cannot use them on FreePBX (like any other T46G) I would waste money on them.

Thank you

3CDaemon or TFTPD32 as a TFTP server

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Can someone here teach me on how to use 3CDaemon or TFTPD32 as a TFTP server.

I Want to use it with my yealink phones and 3CX Phone system.


Thank you very much in advance!

Managing\Enabling 4 simultaneous calls W56P

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I need some assistance please.

We recently purchased a W56P Handset and Base combo and our client required both SLA and the ability to manage multiple calls, up to 4 simultaneously from one SIP account on the one device as advertised. Answer a call, put it on hold, answer another call, put it on hold, answer a third call, put it on hold and then answer a fourth call, put that call on hold and then go back and retrieve the second call I had on hold already (example), all the while I have the other three calls still on hold. I am currently doing it with T48G\S phones. I need it (W56P) to behave exactly the same way. I currently assigned all incoming and outgoing calls to be handled by Handset 1.

All of this was managed under Account->Number assignment.

It appears that this is not working. My client insists that he is still only able to answer 2 calls & put on them hold and no more. How do you set up the handset to handle and put on hold 4 simultaneous calls, please?

Thank you in advance!

RE: T41S: severe audio interruptions

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Hi Kevin,

Thanks for following up the issue.

Answering your questions:

1. The audio issues happen in any type of communications: A>BPX, A>B, A<B, Opus/G722 (A/B/AB), MOH/speech/music. We use music to perceive better the interruptions as it's a continuous reproduction and playing MOH directly from the BPX excludes any quality degradations caused by the other endpoint, but human speech with another phone has the same problems.

After some extensive tests simulating different environment conditions (and rebooting the device multiple times) we were not able to consistently reproduce the more severe interruptions. It looks like there were 2 issues: some strange interruptions that could last for 3-4 seconds or more and repeat very frequently (and sporadically, like in the first post where a second call had no such issues) and small interruptions due to environment issues (network delays, packets reordering, etc.).

We came to a conclusion that the small interruptions issue is due to deviations of the packets order/timing from the absolute perfect condition. It looks like our Yealink phone is extremely sensitive to any deviation in order or latency, it doesn't compensate for this imperfections with adaptive jitter buffer (was configured with "adaptive,20,200,20"). At the same time, Linphone was able to "eat" most of the environment issues without audible defects or with light sound distortions.

At the same time, for unknown reason, Yealink phone was always behind Linphone in audio reproduction, sometimes with a delay of up to 300-500ms, most of the time around 80-100ms (subjective judgment). One could guess that this could be an extremely large jitter buffer, but the small interruptions were happening with this delay too.

When enabled Settings > Voice monitoring > Display Report options on phone, there were strange statistics. JitterBufferMax, Packets lost and RoundTripDelay were always at 0, but Jitter had strange numbers: it could stay at 0 when there really was jitter, after one reboot it stayed constantly at 950-960 without dropping to normal values though there was no such jitter (pictures attached). At other times it was varying from 0 to 400, but not consistently, i.e. when there was jitter, the value sometimes was staying without change, at other times it was following the real jitter pattern. At the same time Linphone was changing its jitter buffer from 0 to 60ms, staying most of the time at 40ms.

We got an impression that there was no jitter buffer enabled at all or it was discarding buffered data too quickly as the phone was not able to compensate for most deviations. I'm not sure if this is a standard behavior or we got a defective unit, or there are some options we're not aware of that deal with this sort of issues.


2. We don't have another Yealink or any other (non-Cisco) hardphone at hand, but were using multiple smartphones (iOS/Android) and PCs (Win/Mac/Linux) with Linphone, WiFi and cable connections. Everything works well in any combination, except the Yealink. Even tried disconnecting Yealink's cable and connecting it to a PC with Linphone.

3. Syslog and PCAP files attached: yealink.pcap is a pcap generated by the phone for about a minute of a call to PBX with MOH, fs.pcap is a pcap generated at the Linux machine running FreeSWITCH BPX almost at the same time as yealink.pcap (was started some seconds before and was turned off some seconds after). We generated varying CPU load at the PBX machine during this recording to cause some jitter and got a lot of small interruptions on Yealink. During this recording the same channel was reproducing on Linphone without interruptions (there were 2-3 very light sound distortions for about half a second).

4. Depending on the demand from our clients, we expect that we could purchase some 20-30 phones this year and more in the future. We're currently using Cisco 7900 + chan_sccp_b + Asterisk with G722 and would like to setup new deployments with Yealink T4S + FreeSWITCH using Opus (and, according to our clients' decision, possibly migrate existing deployments).

We really like the new T4S line for the features, ease of configuration (compared to the Cisco phones) and its look&feel, but we can't put it in production with this performance.

Please let me know if you need additional details.

Regards,
Anatoli

BLF Key conflicts

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Hello,

We T48G's keeps getting conflicts on the BLF Keys each day. When I reboot the devises the problem is gone and the BLF keys works properly on every devise. Then after a while (not even a day) some BLF's:

- show users "talking" when they are not? eventually after a while it goes to idle.
- doesn't show when users are "ringing or talking" at all while they are.
- cant puck up other phones with BLF while that phone is ringing (probably because of my second point)

This is starting to get a bit annoying. Can Someone plz direct us to a possible answer?
Or tell me what to supply.

Creating a user account is nearly impossible!

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I tried to create an account for this forum, and it was nearly impossible! This is really a poorly made forum software!

Here a screen video from my try:


LDAP list is empty; misconfiguration?

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Hi,

I am using Active Directory (LDAP) of an Apple server.
I made a test account with following data:

User: test
password: test
Server 10.0.1.99

I have one entry in the LDAP directory:
Max Mustermann
private number: 040 xxxxxxx
iPhone number: 0451 xxxxxxxx

I have setup this LDAP account in a Yealink T46G, but the list is empty. Max Mustermann is not visible. So I think I made a mistake, but I do not know what.

.png  Screenshot 2017-08-23 19.10.24.png (Size: 71.48 KB / Downloads: 1)

Yealink Meeting Server

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Hi,

How can i configure yealink Meeting server to register my yealink VC Mobile from Outside(4G Network)

i dont have a public IP For Meeting server.
i have dyndns that is configured in my router.

T46S Holding OK button allows factory reset

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Hello,

Just discovered that no matter what level of lock down choosen on the phone T41S/T42S/T46S users can reboot or factory reset phone.

The "X" button if held will allow user to reboot phone, not really an issue but was unknown, hold time is 5secs.

The "OK" button if held will allow user to factory reset the phone, not acceptable behaviour, hold time is 5secs.

What can we do to remove/block/prevent users from doing this. This type of functionality should not be hard coded in to a buttons functionality.

Cheers,
Scott

t58v disabling apps

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Hello, I was just looking into if there is a way to disable apps on a t58v. Specifically, we'd like to do the following:
Disable access to "Settings" except for admin user
Disable Email and Calendar apps
Disable camera (not just flip shutter?)
Disable recorder

Please let me know if there is any way to do this. I did not see any items in the web interface for these, which is typically how we apply preferences to a template for other phone such as the t46 and t48.

T58V Issues using Number Pad

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Anyone else have an issue if the phone is asleep (black screen) when you go to dial a number the first number you dial is missed because it wakes up the phone. Is there a way to prevent this?

I have not been able to find a screensaver function so the phone does not go to sleep. I can set the phones backlight time to ALWAYS but i want to prevent screen burn over time.

This is very frustrating!!! We are using 3CX so it does not provision the phone yet.

Dial Plan Replace Rule

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My phone system requires a 9 before the number being called to dial out. I'm trying to create a replace rule for 911 under Settings>Dial Plan. First I created a replace rule prefix 911, replace 9911. When I dial 911 it does not add the extra 9. Then I created replace rule prefix ([0-9]xx), replace 9$1. This also does not add the 9 in front of 911, but when I dial 511 it will add the 9 and in the history it shows i dialed 9511. Any ideas as to why the phone will not add the 9 in front of 911? I am using T48G's with 35.81.0.110 firmware.


Thanks

T58v using third part app

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Hi i want to install three apk on my phone
Ipbell : to control door phone
Domowidget: to control my home with widget on dashboard
And hangout.

Only the first one thims to be installed, but when i lauch ip ell i have a messagebox that they the application need administrator permission.

what is the solution to use third part app on this phone?

one way audio issue with T41P

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Hello,

I've some troubles with a T41P (the only Yealink phone one my LAN network). When I answer to an external call, the caller can hear me (T41P-> external : audio ok) but I can't hear him (external -> T41P : no audio).

If I answer the call with an other SIP phone (which is ringing at the same time that T41P) there is no issue.

The SIP server and the phones are on the same network (i.e. no nat and no routing).

The strange thing is that in the PCAP traces every thing looks good and furthermore the RTP voice stream from the external caller is actually received by the T41P !
(I was able to play the both stream - T41P->external and external->T41P - with Wireshark).

After some tests, it looks like that the issue happens if the RTP stream is first established by the SIP server and secondly by the T41P. In the opposite case (RTP first established by the T41P) every thing works.
When I get this issue it is possible to get the audio working by putting the call on hold and resuming it : after the call has been resumed, the caller can be heard.

If I call an external number there is no issue (audio works fine in the two ways - in this case, RTP stream is first established by the T41P).

For information :
SIP server : Asterisk 14 (IP 192.168.101.27)
T41P : version 36.81.0.110 (IP 192.168.101.30)
in attachment : the PCAP file
.pcap  36.81.0.110_7_23_9.pcap (Size: 476.78 KB / Downloads: 0) and the .bin configuration file
.bin  config.bin (Size: 209.52 KB / Downloads: 0) (I can't attach the allconfig.tgz file :/)

Someone have some ideas ?

Let me know if you want more informations.

Best regards,
Jonathan

T23G Auto Provisioning Issues

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I have been trying to set up a provisioning server for our Yealink phones and have had some success but I can't get the T23G phone I have to provisioning. I've managed to get a T26P, T21P E2, and T29G phone to connect to the server and download their respective configuration files, but while the T23G can reach the server it can't download any config files.

They all have the same auto provisioning configuration: IP Address, User, Password, Certificate, Certificate settings, and the T23G has the latest firmware but for some reason it does not want to provision. This is the first time I've set up anything like this and it's been quite challenging building it all from scratch so any help would be appreciated.

I've attached a pcap and level 6 log file, as well as a few screen shots of my configuration. Also I noticed on my server logs (IIS 10.0) that the same HTTP error keeps appearing (500 - Internal server error) and the same Windows error (64 - The specified network name is no longer available)


Thank you


.png  screen1.png (Size: 231.82 KB / Downloads: 0)

.png  screen2.png (Size: 206.2 KB / Downloads: 0)

.png  screen3.png (Size: 296.41 KB / Downloads: 0)

.pcap  44.81.0.110_5_31_2.pcap (Size: 394.8 KB / Downloads: 0)

.bin  config.bin (Size: 167.52 KB / Downloads: 0)

Local Contacts Backup Not Working

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I cannot seem to get this feature to work.. Using 3CX 15.5 SP1, T46G on 28.81.0.110, and the following relevant provisioning file settings:

Code:
static.auto_provision.local_contact.backup.enable = 1
static.auto_provision.local_contact.backup.path = %%PROVLINK%%/LocalContacts/%%mac_address%%-contacts.xml

Any tips?

Cannot connect socket node

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I am using T4xS phones with Firmware 66.81.0.110, the phones often loose registration with my server where the phone syslog shows "NET <6+info > [255] Cannot connect socket node / select timeout (0 ms)" after the gray out period ends the phones come back to life however the active call drops, the blf keys gray out.

Very annoying to our users, please any help would be appreciated.

.txt  001565bd824b-sys.txt (Size: 1.92 MB / Downloads: 0)

T41P - one way audio issue

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Hello,

I've some troubles with a T41P (the only Yealink phone one my LAN network). When I answer to an external call, the caller can hear me (T41P-> external : audio ok) but I can't hear him (external -> T41P : no audio).

If I answer the call with an other SIP phone (which is ringing at the same time that T41P) there is no issue.

The SIP server and the phones are on the same network (i.e. no nat and no routing).

The strange thing is that in the PCAP traces every thing looks good and furthermore the RTP voice stream from the external caller is actually received by the T41P !
(I was able to play the both stream - T41P->external and external->T41P - with Wireshark).

After some tests, it looks like that the issue happens if the RTP stream is first established by the SIP server and secondly by the T41P. In the opposite case (RTP first established by the T41P) every thing works.
When I get this issue it is possible to get the audio working by putting the call on hold and resuming it : after the call has been resumed, the caller can be heard.

If I call an external number there is no issue (audio works fine in the two ways - in this case, RTP stream is first established by the T41P).

For information :
SIP server : Asterisk 14 (IP 192.168.101.27)
T41P : version 36.81.0.110 (IP 192.168.101.30)
in attachment : the PCAP file
.bin  config.bin (Size: 209.52 KB / Downloads: 0) and the .bin configuration file
.pcap  36.81.0.110_7_23_9.pcap (Size: 476.78 KB / Downloads: 0) (I can't attach the allconfig.tgz file :/)
the allconfig.tgz can be download at : https://filex.univ-paris1.fr/get?k=rNQqJ...0mB&auto=1

Someone have some ideas ?

Let me know if you want more informations.

Best regards,
Jonathan

one way audio issue with T41P

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Hello,

I've some troubles with a T41P (the only Yealink phone one my LAN network). When I answer to an external call, the caller can hear me (T41P-> external : audio ok) but I can't hear him (external -> T41P : no audio).

If I answer the call with an other SIP phone (which is ringing at the same time that T41P) there is no issue.

The SIP server and the phones are on the same network (i.e. no nat and no routing).

The strange thing is that in the PCAP traces every thing looks good and furthermore the RTP voice stream from the external caller is actually received by the T41P !
(I was able to play the both stream - T41P->external and external->T41P - with Wireshark).

After some tests, it looks like that the issue happens if the RTP stream is first established by the SIP server and secondly by the T41P. In the opposite case (RTP first established by the T41P) every thing works.
When I get this issue it is possible to get the audio working by putting the call on hold and resuming it : after the call has been resumed, the caller can be heard.

If I call an external number there is no issue (audio works fine in the two ways - in this case, RTP stream is first established by the T41P).

For information :
SIP server : Asterisk 14 (IP 192.168.101.27)
T41P : version 36.81.0.110 (IP 192.168.101.30)
in attachment : the PCAP file [attachment=4557] and the .bin configuration file [attachment=4558] (I can't attach the allconfig.tgz file :/)
the allconfig.tgz can be download at : https://filex.univ-paris1.fr/get?k=rNQqJ...0mB&auto=1

Someone have some ideas ?

Let me know if you want more informations.

Best regards,
Jonathan

T42G VLAN settings for PC VLAN

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Hi,
we are new to Yealink phones. We want to use a T42G connected to VLAN 41 und thru the phone a PC connected to VLAN 31.

The Switches are HPE / aruba HP 2530-48G-PoEP Switch (J9772A)

the phone: Firmware Version 29.80.150.2
Hardware Version 29.1.0.0.0.0.0

We tried:
switchport VLAN 41 untagged, VLAN 31 tagged
switchport VLAN 41 tagged, VLAN 31 tagged
Phone:
WAN port disabled / enabled, VID 41
PC Port diesbled / enabled, VID 31

In some cases the PC gets a IP address from VLAN 41, or apipa address

In no config it gets an IP from VLAN 31.

What is the correct configuration on the switch and the phone?

Thanks
Uwe
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