similar to: [OT] Yealink T26/28/38 and Open-VPN

Hi Guys We have a strange a strange issue at a client they have 3 panels on their phone and every so often the panels reboot themselves yet the phone stays on. We decided to replace the T26 for a T28 to see if it fixes the issue and still have the exact same issue. Has anyone seen this before? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi! > > We have a strange a strange issue at a client they have 3 panels on their phone and every so > often the panels reboot themselves yet the phone stays on. > > We decided to replace the T26 for a T28 to see if it fixes the issue and still have the exact > same issue. > > Has anyone seen this before? > I frequently use the newer T48G and T46G phones with the EXP40

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi List, I have come across the above handset a few times in the UK, They are quite cheap over here (~?80) Not the best handset in the world but works well enough. I have been asked to setup a central config server for a large collection of these handsets. I know they can do Auto provisioning via FTP/HTTP/TFTP I have got an example of the generic

I've just started deploying these (well the T28P model) after years of Snom issues and they look pretty good (although the documentation is execrable; if you thought the Snom stuff was obtuse Yealink have got them knocked into a cocked hat!). Anyway, for provisioning I use HTTP with a DHCP entry like:- # # Yealink Phones # group { #

Hello all, After numerous issues with Snom phones (360/370/870) potentially looking to migrate too Yealink as their product range looks very promising indeed. Are any of you using them with Asterisk ? How do they perform ? Do you use mass deployment at all ? Would be very interested to hear from you. -- Thanks, Phil -------------- next part -------------- An HTML attachment was

Hi, I would like to test the following COLP use case : Alice and Bob are both using a SIP phone registered on a Asterisk 10 server. Alice dials Bob's extension. While Bob's phone is ringing, Asterisk updates Alice phone screen with Bob's name, so that at a glance, Alice can check she dialed the correct number. Before diving into Asterisk documentation, I would be happy to be

Hi Krzysztof, Sure, please see below. DAG.dump.() before and after, annotated with what I believe the DAG means. I've spent some time debugging the method but it's proving difficult to determine where the logic is misfiring. Disabling the entire combine causes a lot of failing x86-64 tests - I may have to learn an upstream vector ISA to make progress on this. Thank you >From your

Hello list, I need to set Voice QoS and SIP QoS for YeaLink. The possible values are 0 ~ 63. With Grandstream I can fill in DiffServ 46, which is EF. That's what I want. With Snom I fill in 184, which corresponds to EF or DSCP 46 (according to their wiki) But what value do I want to fill in with this YeaLink ??? This is a conversion table :

Hello, We are toying with setting up a redundant data center for our hosted PBX product, and plan to use the OpenVPN feature of our Yealink phones for connectivity to each data center. The feature has been fantastic with the first data center, allowing us to bypass all SIP NAT issues entirely and allowing remote access to the phones' web interface without having to touch the customers'

I have two problems. The first is that when I dial my number from a phone on the Internet or any phone outside my LAN, Asterisk does not respond in any way, which means somehow my system is not picking up the fact that there's an incoming call to it. The second problem is that I thought I'd try an internal phone to see if I could get the hello-world stuff working at the least. I

I have a setup with Yealink phones & Asterisk Server (all latest patches). I am using BLF to display the states of other phones. While this works MOST of the time (busy, being called) it does NOT work when a phone is NOT regisstered at all, the yealink phones display a green dot EVEN if a phone is turned off (try explain this to users, they are shaking their heads!!!) I can see on the

I'm having a strange problem recently with a Yealink SIP-T28P phone connected to Asterisk 11.4.0 via openvpn. It was working fine for months, and now when I dial anything from the phone, it shows "Forbidden", and the Asterisk console shows: [May 21 10:47:49] NOTICE[28518][C-00000004]: chan_sip.c:25189 handle_request_invite: Failed to authenticate device "Ext 110" <

So I've got a bit of a head scratcher. Wanted to get some insight. I've got a PBX running 12.3.0 We're a ULAW shop from end to end. But I've been playing with G722 just for fun. I've got a Yealink T46G on my desk, And my colleague, A Polycom IP650 (Same office). Basically, Whenever I make an outbound call to a destination to something not G722 ready, I get no

I see we're missing an isel pattern for add producing carry and doing a memory RMW. I'm going to see if adding that helps anything. ~Craig On Mon, Feb 27, 2017 at 8:47 PM, Nirav Davé via llvm-dev < llvm-dev at lists.llvm.org> wrote: > Yes. I'm seeing that as well. Not clear what's going on. > > In any case it looks to be unrelated to the alias analysis so barring

On 10/20/2016 9:28 AM, Cameron McInally via llvm-dev wrote: > I should have attached the generated asm to save some trouble. > Apologies for that and attaching now... > > > > On Thu, Oct 20, 2016 at 12:26 PM, Cameron McInally > <cameron.mcinally at nyu.edu> wrote: >> On Thu, Oct 20, 2016 at 12:05 PM, Mehdi Amini <mehdi.amini at apple.com> wrote:

Hello. I'm having problems at instruction selection with my back end with the following basic-block due to a vector add with immediate constant vector (obtained by vectorizing a simple C program doing vector sum map): vector.ph: ; preds = %vector.memcheck50 %.splatinsert = insertelement <8 x i64> undef, i64 %i.07.unr, i32 0

Hello, I am running Asterisk 11.12.0 on CentOS 6.4. The asterisk server and internal phones are located on the 10.10.32.0/21 LAN subnet. I have many internal SIP phones, which appear to be working correctly. I have a few external phones (Yealink SIP-T32G or other Yealink model) on 192.168.32.0/24 which have an OpenVPN client configured on them that connects back to the LAN network through a

----- Original Message ----- > From: "Administrator TOOTAI" <admin at tootai.net> > To: asterisk-users at lists.digium.com > Sent: Thursday, April 30, 2015 4:43:33 PM > Subject: Re: [asterisk-users] OpenVPN Clients Intermittently Cannot Call In > > > I am running Asterisk 11.12.0 on CentOS 6.4. The asterisk server and > > internal phones are located on

Hi, Is there a kind of scope mechanism in the instruction lowering pattern language in order to control where instructions are inserted or how they are later reordered during the SelectionDiag linearization? I know the glue chain that stick instructions together. But such mechanism in not provided in instruction lowering pattern. I'm facing many situations where some patterns are lowered into

Hello, I have Snom, Cisco, Grandstream & YeaLink phones. Is there a way to push a centralized phone book to these phones ?? Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101202/ea25cd7e/attachment.htm