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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're running into some problems with our system and want to investigate different support companies (apart from B4B) to handle our Yeastar and firewall configuration. Likely this could be done remotely. I've drawn a blank on firms in Ireland who support Yeastar. Can anyone recommend a Yeastar-supporting phone company?

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

Hi All, I've seen that quite a few people are using Gamma for their SIP trunk. I've also seen that someone is using them with FreePBX. Just wondering if anyone out there is able to help with trunk configuration for Gamma in FreePBX. So far I've had no luck even with lots of holes punched in the Firewall. The provider that's sold the gamma trunk to us isn't being much help and I haven't been able to find contact details for Gammas technical support myself. Asterisk forums have had some posts but still had no luck. Any help would be hugely appreciated! Will

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...!

Hi All We have a Mitel ICP3300 and we are wondering if we can somehow connect our Windows Mobile 6 devices via our wifi network to the phone system so they act as extensions. If not WM6, we would look at getting Android devices to do the same. We only really need to be able to call each other on-site. The devices dont need to be able to call in/out from anywhere other than our site. We have tried something similar with iPhone and iSip and it worked well, but dont really want to buy half a dozen iPhones. Any suggestions would be greatly appreciated. I know Mitel do their own phones but I have a feeling they will be expensive. Looking for something very simple. Need to make and receive internal calls - thats it. Thanks

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

Hi, Has anyone had any luck getting Office 365 to work with on premise PSTN using a SIP trunk from Gamma? We are experiencing issues with 403 forbidden errors and Gamma are not being very helpful. CCE is configured as per Microsoft's guidance. Traffic is confirmed to be getting to Gamma through our firewall, but then rejected due to private IP being included in the SIP header as part of the VIA address. Any help would be appreciated. Thanks Trev

Hi All I have just upgraded my phone system to the Samsung OfficServ 7400. Am now looking to upgrade the phone lines I am using from Centrex to SIP trunking. Does anyone have a company they are using and would recommend? I am looking for something like 8x trunks 1 number ported across all trunks no extentions dedicated "broadband" connection the the service provider Any feedback is much appreciated Cheers Joe

I don't know if a product exists but what I'm after is a device that will allow me to use a SIP handset on an analogue phone line. Basically, I want to be able to be able to receive calls to my landline via my smartphone and (to a lesser extent) make calls via the landline. Ideally, the device will also act as a DECT base station. I don't want a lot do I? Of course, I suppose I could build an Asterisk box but it's going to be overkill!

I wanted to see if you guys can give me idea where this issue might be arising from. Scenario: We have Comcast as our internet service provider in the office. We use a company called GPhone to provide us with our SIP trunk. For our hardware we have a German phone system called Innovaphone. We use the same hardware with different providers in many countries without any issues. Problem: Every once a week or twice a week (actually just randomly) we have issues receiving calls. The system shows as if everything is running OK but we can not receive call from an external number. We are able to call within the office and we can even call external numbers but can not receive incoming calls when the error occurs. The problem is that our technician in Europe says that he sees nothing on the logs and Comcast says everything is perfect on their side. I am still waiting on a response from GPhone. The only thing we can do is to disable the SIP on our system and then enable it again for it to start working correctly. It seems as if there is an error and the connection to the SIP is not established automatically. The problem is that we do not receive any errors so we do not know the phones are down until someone is informed that the calls cant go in. I wanted yo know if anyone has any idea what could be the issue? I don't know if the issue can be through the provider of the SIP Trunk, the hardware, or ISP. Any suggestion will be really appreciated.

Hi, We are currently looking for a new phone system. We are looking at a voip system using SIP trunking. We have had various quotes but would like to know what everyone is using and also which company you all use. The best quote we've had so far is from Teleconnexions based in Carlisle, this would be ideal for us as they provide our phone lines also