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Anyone with a VOIP provider who is using SV1 (maybe SV1000) and been told its going EOL? Our provider (who I don't wholly trust has told us its going EOL on April 30th)

Evening everyone. I'm currently experimenting with setting up a Voip system to replace our old pbx. I've used freepbx at home for a few years so I'm pretty familiar with setting that up, provisioning the phones, dialplans ext... So far I've installed the Distro on an old server, and setup 10 Cisco 7941 phones (in SIP mode) and all is working well. My main concern now is the QoS across the LAN (we will still be using our ISDN lines for external via a Patton Gateway for now). At the moment the call quality between extensions is excellent, however what with the school being almost empty the voice traffic doesn't have much to battle against. Our network has a 1G backbone, and a mixture of 100mb/1G switches (Cisco & HP). We have a number of VLANs across the site. My understanding is that the phones & Asterisk by default (in sip.conf & on the cisco phone status page) add the CoS value (5) and the DSCP code (cs3) to the packets they transmit, so is it a case of making sure 802.1p is enabled on all managed switches? Or is it necessary to setup a new Vlan just for voice and create a trunk between each phone & the switch (tagged voice vlan & untagged data vlan for the pc passthrough)? Any experience of your own way of doing this would be interesting to hear, along with suggestions on the best method in this situation. Many many thanks in advanced!

We've rolled out Office 365 but can't finish the job because our Cisco VoIP system uses the Unified Messaging services of our on-premises Exchange servers, meaing that affected mailboxes (almost all staff) cannot be migrated. I've just been quoted a big heap of cash to integrate our phones with O365, but as the telephony servers and handsets are over six years old I'm reluctant to spend a lot on them. Does anyone have any recommendations for a phone system that will provide UM with O365? Preferably hosted - I see gaps in the server rack as A Good Thing these days. Currently our network is VOIP internally but using conventional ISDN to the outside. I'm assuming that any replacement would route calls through our newly-boosted 500Mb data connection to an external gateway.

Hey all, Is there any Cisco Call Manager experts? We currently use Callmanager and all is working fine and am happy doing the day to day config but over the Summer we have a local primary coming on board and we will need someone to help us set up the routing of there number on call manager. Also we've had real trouble getting voicemail to work so would be great if that could be looked at the same time. Anyone point me in the right direction? Thanks Simon

Hi all, Does anyone know if its possible to get a VOIP phone that would be able to comunicate with our internal phone system? We have a computer room at the other side of the school and have been told in no uncertin terms that it will be to expensive to install a traditional internal phone. Will be contacting our supplier soon but thought there might be some ideas here first . James

We have a dedicated call center in our hospital. We are using a IP PBX software from a outsourced vendor. The software is installed in our PCs to attend the enquiry calls.The client pcs using in call center is a data vlan in one of the access switch. The server IP PBX software is installed in data center server farm switch. For the past weeks, the software was having some issues at some time, when i checked with the vendor they said software side is ok upon checking on wireshark logs. They have made several observations & fixes - Key Observations: Wrong Sequence Number (Red Entries) - RTP packets are arriving out of order or being lost, likely due to network congestion or QoS issues. High Jitter Values (Max: 7.03 ms) - Increased jitter can degrade voice quality, causing choppy audio or lag. Packet Loss (0.00%) - No significant packet loss was observed, but sequence errors suggest potential packet reordering. Clock Drift (-79 ms) - A lack of synchronization between sender and receiver clocks can impact audio quality. Cause - Solution. Network Congestion - Prioritize RTP traffic using QoS settings on the firewall or router. Jitter & Delay - Implement jitter buffers or increase buffer size in VoIP settings. Packet Reordering - Check network routing and MTU settings to prevent out-of-order delivery. Clock Drift - Ensure both endpoints synchronize time using NTP. Pls help me what to do from the core switch side or on the access switch side. NOTE- When I checked the ping status from the client PC to Server IP there is no drop or that much latency. Core switches are CISCO 4506 model Access switch for call center department is Aruba Instant ON 1930 model.

Hi, I was wondering if anyone out there has been through a very tough process of moving away from local LEA and sourcing new broadband and IP telephony. We are looking at 100Mb broadband with VOiP provider. We have looked at LGfL which has very good price for the services provided. However, it comes without phones! We had a quote from atomwide telephony for £3000 set up (one form entry primary school) + quarterly charges + phone bills. Thanks for helping!

Hello all, As our move to BSF managed service grinds inexorably on we have a large amount of perfectly sericeable kit that is surplus to requirements We are looking for the best offer for all of these items, and for the buyer to collect them from us (just off junction 19 of the M62) The following items are available Tadiran Colal IP office phone system - 70 extension VOIP phone system with 4x analog and 4x digital ports, plus ADSL card and voicemail. plus roughly 70 Handsets - new cost approx £15000 5 x Netgear GS724T - 24 Port Gigabit smart managed switches - new cost approx £180 3x Netgear FS726TP - 24 port (12 are PoE) 10/100 Smart managed switches with +2 Gigabit ports - new cost approx £220 2x Netgear FS726T - 24 Port 10/100 Smart managed switches with +2 ports 3xNetgear WFS70TP - 8 Port PoE Managed wireless controllers, each can control up to 16 light access points and can funtion individually or all 3 can be stacked to give up to 48 APs - new cost approx £1250 24x Netgear WAGL102 - 802.11abg Light Access points - new cost approx £95 7x Netgear WGL102 - - 802.11bg Light Access points - new cost approx £85 As I said we are looking for best offers for this kit so if you are interested in some (or all) of the above drop me a PM Cheers

Hi all, I am looking at migrating our small MAT (2 schools) to a VOIP phone system. (I know I should have already haha!) I will be using a 3rd party MSP to host everything in 3cx, and each schools SBC will be built into the handsets. So this irons out any need for on-prem hosting etc. My main question is regarding SIP trunks and channels. I believe I know how this works, but please correct me if I am wrong.... 4sc license (between both schools) 1 SIP trunk with 4 channels for the larger school, and 2 for the smaller schools. If my knowledge is correct, the school will have the capability to have 6 concurrent calls between them? however, because they are on a 4sc license, they will be limited to 4 calls between them. This is what the MSP said anyway. He did mention, that if 4 calls is to little, you could upgrade to a 8sc license which will allow them to utilise the 6 channels, (6 concurrent calls) and then there is room to add two more channels to the sip trunk for them to be able to have 8 calls between them. Is this correct? Sorry if this post makes no sense! just wanted to confirm that this information is accurate. TIA.

Hello! Our reception team wanted to change up our greeting on the phone system. We use Avaya IP 500 v2 and for the most part have set up the new greeting correctly using Voicemail Pro and the manager with what was already there. However, they get put through a menu (press 1, press 2...) and then they're now meant to hold to be put through to the reception phone rather than pressing a button. I can't find the setting though for this and it just never connects. All the other bits work. Is anyone able to share their VOIP wisdom with me? Thanks!

Does anyone here use Spitfire with 3cx, I've got a trial with them and followed the trunk setup in their guide but I'm still having difficulty dialing out correctly, I can get outbound to mobiles if I dial the full international number eg 0044etc. Inbound is fine, all calls are actually getting out over the trunk as I get the recorded message from Spitfire saying this number is not registered on the spitfire network. As it stands there's no number transforms or anything, it's had me stumped all day Friday. I do have a support call open with them so I'm sure they'll fix but just wondered if any edugeekers new what I could be missing! Thanks Simon

OK, CISCO is causing me a headache, any experts out there willing to contribute would help tremendously! I have a Direct Dial Number that when I phone it, it automatically hangs up. Now obviously this is because incoming calls to that DDI are being refused/denied/blocked. Problem is, I cannot find where to re-enable for this extension. Any pointers or general shoves in the right direction greatfully received! Thanks

Most Birmingham schools have probably seen this already. For those that haven't........ "Virgin Media have advised us that with effect from 20 June, they will be increasing the price of their Centrex telephony lines from £5 per month to £9.50 per month. There is no justification for this price increase and despite many years of customer loyalty, Virgin Media are unwilling to negotiate to improve this offer. To counter this unwelcome news, Link2ICT are now accelerating deployment of a fully cloud based VoIP telephony solution which will provide schools with a far cheaper alternative that includes all call charges. This will be available during the autumn term." We don't subscribe to Link2ICT's services but do have their Virgin phone lines (non VOIP). So for us, line rental has doubled. Anyone mind posting what you're paying for your VOIP service, and how much bandwidth its using up. I'm reading its around 1mbps per 10 concurrent calls. Doubtful that its worth it for us as we've got less than 20 phone lines. We do however have a 50mbps leased line which is plenty.

Hi - Here's a good one - we have an external building that I want to put two Yealink VOIP phones into - The phones are all on their own, completely separate network LAN. I have run a cable from a network point in the main building to the cabinet in the external building - Patched that back to the dedicated VOIP/POE switch. Plugged the other end of that cable into a dedicated switch in the external building and patched that through to the two phone points.. One works, one doesn't - But works back in the main building. Its as if the line from the main building to external building will only allow one phone. Have I missed something?

Recently I acquired an iPod Touch, and one of our other technicians (whom I purchased the Touch from) acquired an iPhone. We've since proved our case to the network manager that they're invaluable network tools, and he's gone and got himself one as well. What we WANTED to know however, was how to use them as VoIP Phones tied to our internal phone system as SLT had made the decision to get rid of the work-mobiles that we all had. Information online for this was sketchy with very little info available. Folks said they'd done it, but no-one really said how... Well, we've sussed it out... so I'm posting it here in case anyone else out there has been looking. We're using the following: - iPhone / iPod Touch on OS3.1.2 connecting to WiFi via internal Trapeze system - Mitel ICP3300 1 - Ensure you have SIP User Licenses available for number of extensions you want to add... see "License and Option Selection" in the Mitel System Administration Tool. 2 - Create a User extension via "User Configuration" in the Mitel System Administration Tool. Fill all fields in as normal. Set Device Type to "Generic SIP Phone", and set and confirm the "Login Pin" 3 - Download "iSipSimple" from the iTunes AppStore, and transfer to your iPhone or iPod Touch (NB: iPod Touch requires headphones & mic, wired or bluetooth) 4 - Connect to your WiFi connection. Note, Mitel ICP3300 needs to be accessible from this WiFi connection. Use PingLite from Appstore to see if you can ping the ICP3300's IP address from your device. If not, start looking at your infrastructure and routing tables etc - sorting that out is beyond the scope of this post. 5 - Open "iSipSimple", head to "Settings" page, and make sure "RFC3605 Support" is set to ON (otherwise you get patchy audio with frequent dropouts) 6 - Go to "Accounts" page, click "Edit", then "Add new SIP account". 7 - Select "Other SIP Provider" 8 - Set account details as follows... Display Name: telephone-extension-number Domain: IP Address (or resolvable DNS) of ICP3300 Username: telephone-extension-number Password: login-pin Click "Advanced" and set as follows Auth Username: telephone-extension-number Voicemail Number: whatever number you dial on your system to reach voicemail Phone Number: telephone-extension-number [the above settings are the same if you're using X-Lite PC Softphone client with Mitel also] Leave all other settings as standard. Click "Sip Accounts" in top left corner. Click "Save" in top right corner. Click "Account Manager" in top left corner. Your account should be listed alongside an ON/OFF switch. Turn it ON. Once it's connected, the text should become green. Done! Head to the "Dialer" page. You can now be reached on the extension number you programmed, and can make calls to any other internal extension, or to standard numbers, all via your Mitel ICP3300. NOTE: This only functions when iSipSimple is open on your iPhone / iPod. As soon as you close iSip and head to your homepage, you will no longer be reachable. If you want the client to run in the background, you will need a jailbroken iPhone or iPod running Cydia's Backgrounder app. If you have this, then open iSip, turn your account on, and then hold the HOME key until "Backgrounding Enabled" appears on screen. iSip now remains running in background while you work in other apps on your device. This tutorial only covers connectivity using WiFi LAN, where the ICP3300 is connected to the same LAN. For connection from anywhere via internet, again, you'd need a jailbroken device as Apple have disabled using VoIP via the 3G-Broadband connection. Your Mitel would also need to be accessible via the internet so relevant ports would need to be opened and forwarded to the ICP3300 on your company's router. To avoid having to use a JailBroken device, you can pay for "iSip with Push" from the AppStore, but your Mitel ICP3300 MUST be accessible via the internet for the push notifications to work... even if you only intend to use your device on the local network. Hope this helps some of you...!

I've been asked to look at phone systems so I thought I'd ask you guys how you do it. Do you have a POTS, Hybrid or a VOIP system? Do you manage it youself or is it looked after by an external company? Did you use off the shelve hardware and software or is it a dedicated box? How much does it cost? How many internal/external lines do you have? Currently we have an LG 'GDK-186' system. We have an ISDN-30 line coming into school although we only use 8 of these lines. We have 48 extensions in school although a new building is going up, so there will be some more anolog lines going in for that. No idea about the current costs though.

Hi Everyone, Has anyone ever used Nimvelo (https://www.nimvelo.com) as a VOIP provider, they are offering us a very good price for a new cloud based VOIP phone system for our MAT and we would like to know if anyone else in the education sector is using them at the moment. Matt

We currently run cisco CUCM business edition and Unity, I'm looking to change a configuration for the voicemail addressing but uncertain where this settings is or how to do it if its not just a GUI option. When a voicemail is listened to Unity will read the senders directory name or play their recorded personal name but if its a number not in our directory such as external calls we just get the automated message Message from *queue robot voice* 'unity connection messaging system' Ideally we would like this to read out the senders number and Google attempts have led me to other things normally related to recording personal greetings or other settings for VOIP phones which I'm not looking to configure.