similar to: Yealink SIP-T22P Auto Provisioning via HTTP ?

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, Sorry for an OT post but striking out a bit at the moment trying to get a response from Yealink R&D. Has anybody successfully managed to get a Yealink phone to work across Open-VPN when using tlsauth ? We really do hope that it is possible due to the benefits tlsauth offers against DoS. -- Thanks, Phil -------------- next part -------------- An HTML attachment was scrubbed... URL:

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 { #

Hi everybody, I'm seeking information on how to report an IP phone on a call that is occurring on another IP phone. Example: While the A phone is ringing, Asterisk sends a notification to a phone B on the call that is going to A, but this notification is displayed on the B phone display and the user does not need to hit anything to view the information. I'm

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:

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'

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

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

Hello. Anybody in the list is using this IP phone? Regards, Marcelo H. Terres <mhterres at gmail.com> IM: mhterres at jabber.mundoopensource.com.br https://www.mundoopensource.com.br https://twitter.com/mhterres https://linkedin.com/in/marceloterres

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi List, What I hope is a simple question... As the subject states, I would like to know if anyone has setup a Multi Tenant Asterisk Server ? If so, what would I need to do to get to a Multi Tenant setup (preferably an Open Source solution) ? Any suggestions/comments/pointers/URLs ? - -- Gavin Spurgeon. AKA Da Geek -

Hi List, (Long time lecher, first time poster) Not sure if this is the best place to ask, but here goes.. I have a few test Guests on a Xen host, and I would like to start monitoring (and maybe restricting) the network traffic of the guests but not the Host. No I know this can be done with iptables, but I just wondered if anyone knew of a package or how-to to set this up with Xen on CentOS 5 ?

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

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

Hi, We have a user who is putting large call volumes through Asterisk, and wants to BLF monitor up to 90 extensions. We are struggling to find a handset that can keep up with Asterisk :) 1) Is there a handset that will do this? 2) Is there a different (standard) way to send BLF and allow directed pickups? 2a) Or even a handset specific way? Asterisk handles the BLF volume fine, even on quite

Hello! Oh, wise ones, ponder with me over two of the surprises that populate the universe! I have a phone, that I sometimes cannot reach, connected via pjsip. It can call other extensions just fine, it can call out over a trunk to my cell, all is well, but getting a call? Forget it most of the time. Here is all the config relevant to that phone: [murftest12] type=aor qualify_frequency=1992

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

Hello, using Asterisk 1.8.12.2 case : I call with my cellphone to our public telephone number Our receptionist answers the incoming call and does an attended transfer to my colleague ( A ) Colleague answers and the receptionist tells him that I am on the other side. Receptionist transfers the call and I am connected to my colleague ( B ) My question is about the CallerID that the

----- 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