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Asterisk Support Knowledge Base Products and services Technical

Gradwell IP Address ranges

At Gradwell, they send internet traffic from different addresses (known as IP addresses) to allow their telephony systems to work smoothly. Below is the list of IP addresses where their VoIP (Voice over IP) traffic will come from. It’s important that your firewall allows traffic from these addresses however they recommend you don’t set it to allow only from these, just that they are included.

The reason they say don’t allow only these addresses is that there network is dynamic and may shift or new items added and we don’t want this to affect your service.

There are a couple of things you should do to ensure you get the most from the Gradwell Voice services:

  • Check your firewall filtering – is there anything being excluded?
    • If yes – Allow the IP range traffic – this will most likely be in your firewall settings or tools (they all differ so they can’t exactly point you there)
    • If no – you’re good to go
  • If you use UDP traffic then you’ll need to allow Media ports (known as RTP) with the numbers 1024 to 65535

Current ranges as of summer 2021

109.224.232.0/22 109.224.232.0 to 109.224.235.255
109.224.240.0/22 109.224.240.0 to 109.224.243.255
109.239.96.132/31 109.239.96.132 to 109.239.96.133
141.170.24.21/31 141.170.24.21 to 141.170.24.22
141.170.24.5/31 141.170.24.5 to 141.170.24.6
141.170.50.16/28 141.170.50.16 to 141.170.50.31
185.47.148.0/24 185.47.148.0 to 185.47.148.255
194.145.188.224/27 194.145.188.224 to 194.145.188.255
194.145.189.52/31 194.145.189.52 to 194.145.189.53
194.145.190.128/26 194.145.190.128 to 194.145.190.191
194.145.191.128/27 194.145.191.128 to 194.145.191.159
195.74.60.0/23 195.74.60.0 to 195.74.61.255
213.166.3.128/26 213.166.3.129 - 213.166.3.190
213.166.4.128/26 213.166.4.129 - 213.166.4.190
213.166.5.0/24 213.166.5.0 to 213.166.5.255
78.40.243.192/27 78.40.243.192 to 78.40.243.223
87.238.72.128/26 87.238.72.128 to 87.238.72.191
87.238.73.128/26 87.238.73.128 to 87.238.73.191
87.238.74.128/26 87.238.74.128 to 87.238.74.191
87.238.77.128/26 87.238.77.128 to 87.238.77.191

To simplify things a bit listed below are the ranges in common formats.

Rules for Freepbx Custom file “firewall-4.rules”

-A fpbxreject -s 109.224.232.0/22 -p udp -m udp --dport 4569:5270 -j ACCEPT
-A fpbxreject -s 109.224.240.0/22 -p udp -m udp --dport 4569:5270 -j ACCEPT
-A fpbxreject -s 109.239.96.132/31 -p udp -m udp --dport 4569:5270 -j ACCEPT
-A fpbxreject -s 141.170.24.21/31 -p udp -m udp --dport 4569:5270 -j ACCEPT
-A fpbxreject -s 141.170.24.5/31 -p udp -m udp --dport 4569:5270 -j ACCEPT
-A fpbxreject -s 141.170.50.16/28 -p udp -m udp --dport 4569:5270 -j ACCEPT
-A fpbxreject -s 185.47.148.0/24 -p udp -m udp --dport 4569:5270 -j ACCEPT
-A fpbxreject -s 194.145.188.224/27 -p udp -m udp --dport 4569:5270 -j ACCEPT
-A fpbxreject -s 194.145.189.52/31 -p udp -m udp --dport 4569:5270 -j ACCEPT
-A fpbxreject -s 194.145.190.128/26 -p udp -m udp --dport 4569:5270 -j ACCEPT
-A fpbxreject -s 194.145.191.128/27 -p udp -m udp --dport 4569:5270 -j ACCEPT
-A fpbxreject -s 195.74.60.0/23 -p udp -m udp --dport 4569:5270 -j ACCEPT
-A fpbxreject -s 213.166.3.128/26 -p udp -m udp --dport 4569:5270 -j ACCEPT
-A fpbxreject -s 213.166.4.128/26 -p udp -m udp --dport 4569:5270 -j ACCEPT
-A fpbxreject -s 213.166.5.0/24 -p udp -m udp --dport 4569:5270 -j ACCEPT
-A fpbxreject -s 78.40.243.192/27 -p udp -m udp --dport 4569:5270 -j ACCEPT
-A fpbxreject -s 87.238.72.128/26 -p udp -m udp --dport 4569:5270 -j ACCEPT
-A fpbxreject -s 87.238.73.128/26 -p udp -m udp --dport 4569:5270 -j ACCEPT
-A fpbxreject -s 87.238.74.128/26 -p udp -m udp --dport 4569:5270 -j ACCEPT
-A fpbxreject -s 87.238.77.128/26 -p udp -m udp --dport 4569:5270 -j ACCEPT

Rules for IPtables file

-A INPUT -s 109.224.232.0/22 -p udp -m udp --dport 4569:5270 -j ACCEPT
-A INPUT -s 109.224.240.0/22 -p udp -m udp --dport 4569:5270 -j ACCEPT
-A INPUT -s 109.239.96.132/31 -p udp -m udp --dport 4569:5270 -j ACCEPT
-A INPUT -s 141.170.24.21/31 -p udp -m udp --dport 4569:5270 -j ACCEPT
-A INPUT -s 141.170.24.5/31 -p udp -m udp --dport 4569:5270 -j ACCEPT
-A INPUT -s 141.170.50.16/28 -p udp -m udp --dport 4569:5270 -j ACCEPT
-A INPUT -s 185.47.148.0/24 -p udp -m udp --dport 4569:5270 -j ACCEPT
-A INPUT -s 194.145.188.224/27 -p udp -m udp --dport 4569:5270 -j ACCEPT
-A INPUT -s 194.145.189.52/31 -p udp -m udp --dport 4569:5270 -j ACCEPT
-A INPUT -s 194.145.190.128/26 -p udp -m udp --dport 4569:5270 -j ACCEPT
-A INPUT -s 194.145.191.128/27 -p udp -m udp --dport 4569:5270 -j ACCEPT
-A INPUT -s 195.74.60.0/23 -p udp -m udp --dport 4569:5270 -j ACCEPT
-A INPUT -s 213.166.3.128/26 -p udp -m udp --dport 4569:5270 -j ACCEPT
-A INPUT -s 213.166.4.128/26 -p udp -m udp --dport 4569:5270 -j ACCEPT
-A INPUT -s 213.166.5.0/24 -p udp -m udp --dport 4569:5270 -j ACCEPT
-A INPUT -s 78.40.243.192/27 -p udp -m udp --dport 4569:5270 -j ACCEPT
-A INPUT -s 87.238.72.128/26 -p udp -m udp --dport 4569:5270 -j ACCEPT
-A INPUT -s 87.238.73.128/26 -p udp -m udp --dport 4569:5270 -j ACCEPT
-A INPUT -s 87.238.74.128/26 -p udp -m udp --dport 4569:5270 -j ACCEPT
-A INPUT -s 87.238.77.128/26 -p udp -m udp --dport 4569:5270 -j ACCEPT
Categories
Handsets Products and services

Sangoma P310 and P315

These phones are very competitively priced; and are perfect for anyone that needs reliable, basic calling functionality or needs a lot of phones either in one place or over a distributed set of large-facility locations. 

However, you can be rest assured that they are manufactured to very high standards and have all the qualities you would expect in more expensive models.

SPECIAL OFFER: When you buy 9 Phones you’ll get another one free

With these two new phones you can utilise essential Unified Communications features without sacrificing the function and performance that is usually only available with more expensive IP phones!

Want a closer look? Watch the first P315 being unboxed in Europe! Contact us for how you too can get your hands on the P310 or P315 and for bulk pricing.

Unboxing the P3xx handsets

Categories
Knowledge Base Products and services

Aastra 6753i Transfer

Step By step instructions for call transfer when using the Aastra 6753i with firmware 3.x.x and above.

Phone Idle screen.

Once a call is answered their number will show and an icon of a ‘off hook phone’ will also show

To transfer the call press your ‘Transfer key’. Another ‘line’ will show numbered 2 with a ‘ > ’ next to it.

Enter the number you want to dial and press ‘>‘ dial if the call isn’t immediately dialed.

To ‘Blind’ transfer the call press the Transfer Button again or put the Handset down. NOTE if you do this you will not be able to get the call back.

After pressing dial the Phone Icon will show ‘ringing’

To get the call back while it is ringing press the ‘ < ‘ button shown on the display next to ‘Cancel’. Then L1 in this example will flash and ‘call held’ will show on the display as below, you need to get the call back by pressing the Flashing Line Key.

If the call goes to Voicemail or the caller answers the display will show the ‘off hook’ icon against 2

If the Caller wants the call then Press the ‘Transfer key’ the Red ‘Hangup key’ or put the handset down and the call will be transferred to them. Do not press the ‘>‘ Drop button.

If they don’t want the call or it goes to voicemail and you want to get the caller back, Press the ‘ > ‘ Drop Button and that call will be dropped and as before ‘call held’ will show on the screen and you press the L1 button to get the caller Back

Categories
Handsets Software

Sangoma Connect Mobile

We know that right now, communication is key. We also know that you need to be able to work where it makes the most sense for you, so mobility is a must. The latest milestone in our mission to help you communicate more effectively is the next-generation mobile softphone app for FreePBX and PBXact. It’s called Sangoma Connect Mobile, and it is now available to download!

Sangoma Connect Mobile combines the best of our existing solutions with new core mobile technologies to provide a greater communication experience for calling and collaborating with coworkers.

Sangoma Connect Mobile provides rich voice and video calling for PBXact and FreePBX, on iOS and Android devices. We’ve made the onboarding experience very simple for system administrators to provision, and for users to install:

  • Administrators: enable Connect users via the User Manager within the PBXact or FreePBX admin interface. That automatically sends an invitation email message to each user.
  • Users: open their invitation email and follow simple instructions to download the app from the app store, then click a ‘magic log-in link’ in the email to immediately configure – no need for details like hostnames, extension numbers, or passwords.

And that’s it! It just works. Issues like certificate management, which can be a manual process with other apps, are handled automatically.

Features Included:

  • Blind and assisted transfer
  • 3-way calling
  • Contacts and favorites – including BLF
  • BLF Lamp status of features that support it such as Day/Night and CallFlow
  • DND with scheduling for out-of-office convenience

Another great feature is that we’ve integrated Sangoma Meet, our award-winning video conferencing platform within Sangoma Connect Mobile, making it super simple to create or join a multi-party video conference.

How to Get It

Sangoma Connect Mobile is included, free-of-charge, for all new and existing PBXact systems.  It is also free-of-charge to FreePBX customers with an active Zulu license, for the same amount of users already purchased. Simply perform an update on your system, and Sangoma Connect will be available within the User Management Module.

Purchasing Options

FreePBX customers, without Zulu, wishing to purchase a Sangoma Connect Mobile license can do so from the Sangoma Portal Store.  We have integrated Sangoma Connect Mobile with Zulu Desktop so that customers  can take advantage of even further collaboration tools for your MacOS or Windows computer. Simply purchase the Zulu license from the portal store and choose to use Sangoma Connect Mobile, or both! If needed, here are some helpful instructions on how to purchase a license and how to install on your FreePBX system.

Download the Mobile Client

End users can download the iOS or Android app, for free, by searching for ‘Sangoma Connect’ in the respective app stores. A direct download link is also available within the invitation email end-users receive. Here is information on how to use the app.

Categories
Knowledge Base

Disabling SIP ALG on a Thomson Router

Introduction

SIP ALG is used to try and avoid configuring Static NAT on a router. Its implementation, however, varies from one router to another, often making it difficult to inter-operate a router with SIP ALG enabled with a PBX. In general, you would want to disable SIP ALG and configure one to one port mapping on the router.

In this article, we will show you how to disable SIP ALG on a Thomson router. SIP ALG on this router is known to cause problems with VoIP calls. Proceed as follows:

  1. Open Command Prompt – “Start” → “Run” → type “cmd” and press “Enter”.
  2. In Command Prompt, type “telnet 192.168.1.254” and press enter. 192.168.1.254 is the default IP address of the router. If you are running on Windows 7/8/8.1/10, you might need to install the telnet client from “Control Panel” → “Programs and Features” → “Turn Windows features on and off”.
  3. The default username is “Administrator”, and there is no default password, leave blank.
  4. Type “connection unbind application=SIP port=5060” and press “Enter”.
  5. Type “saveall” and press “Enter”.
  6. Type “exit” and press “Enter” to exit the telnet session.

SIP ALG in now disabled on your Thomson router.

Notes

  • The SIP phones behind this router should be configured not to use STUN
  • The SIP phones must NOT be configured with a local port of 5060 or 5061. The local port of the phone must be changed to something else.
  • Configuring a SIP Phone behind a Thomson router might require port forwarding to be implemented on the router. So you will need to port forward the SIP and AUDIO (RTP) ports on the thomson router and point them to the SIP Phone’s IP Address.
Categories
Asterisk Support Covid-19 FreePBX Knowledge Base Remote Working

Disabling Router SIP ALG

With many companies asking their employees to work from home, a common problem when trying to use a sip phone on a home network is the SIP ‘helper’ or ALG, Here is some advice on how to disable it on the more common routers that you may encounter.

SIP ALG (Application Layer Gateway) modifies VoIP traffic with the aim of solving NAT and firewall related problems. SIP ALG does this by inspecting SIP packets and modifying SIP Header and SDP data.

Unfortunately, SIP ALG was poorly implemented in a lot of cases, which has lead to it causing more issues than it corrects and due to this, we believe that, in general, it is best disabled.

Note – Many routers will re-enable SIP ALG after being powered off and on, or sometimes after a firmware update, therefore if it has been disabled in the past, and you know that the router was recently updated and powered off and on again, then it is always a good idea to log in to the router and double check the setting.

Virgin SuperHub: SIP ALG cannot be disabled in the settings of SuperHubs. Please click here for advice troubleshooting issues with SuperHubs. 

BT: SIP ALG cannot be disabled in the settings of BT HomeHubs, but can be disable with BT Business Hub versions 3 and higher:

Disabling a BT Business Hub 5’s SIP ALG

Fritz!Box: SIP ALG can’t be disabled.

DrayTek routers: Log in to your DrayTek via Telnet using an SSH client such as Putty: http://www.putty.org/

Check if SIP ALG is Enabled or Disabled:

To check if SIP ALG is Enabled or Disabled enter this command: sys sip_alg ?

If SIP ALG is disabled a ” 0 ” result will be returned.  If SIP ALG is enabled the result will be ” 1 “.

Disabling SIP ALG:

To Disable SIP ALG enter the following:

sys sip_alg 0
sys commit
sys reboot

The router will restart and save your changes.

Click here for additional general information about DrayTek Firewall setup. 

TP-Link routers: How to Disable SIP ALG on TP-Link ADSL modem router

Linksys: Check for a ‘SIP ALG’ option, in the ‘Administration’ tab under ‘Advanced’. 

May also need to disable SPI Firewall. 

Microtik: Disable ‘SIP Helper‘. 

Netgear: Look for a ‘SIP ALG’ checkbox in the ‘WAN’ settings.

Port Scan and DoS Protection should also be disabled.

Disable STUN in VoIP phone’s settings. 

D-Link: In your router’s ‘Advanced’ settings –> ‘Application Level Gateway (ALG) Configuration’ uncheck the ‘SIP’ option. 

Huawei: Many routers support SIP ALG (usually found in the ‘Security’ menu). 

SonicWALL Firewall: Under the VoIP tab, the option ‘Enable Consistent NAT’ should be enabled and ‘Enable SIP Transformations’ unchecked.  

Thomson: How to Disable SIP ALG on a Thomson Router HERE

Test with STUN disabled in your VoIP phone’s settings.

Adtran Netvanta: Disable SIP ALG under ‘Firewall/ACLs’ –> ‘ALG Settings’.

For Technicolor TG588V routers see this document for step by step details

Even if there isn’t a SIP ALG option in your router’s settings, it may still be implemented. TelNet commands must be used to disable SIP ALG with TechnicolorThomsonSpeedTouch, some Draytek and some ZyXEL routers. 

Categories
Blog Gateways Services

ISDN Switchoff…

With the end of ISDN looming, the switch off could affect over two million businesses in the UK which will come at a cost for those affected. For many, the cost implications to replace their entire Telphone system would simply be too much, but with Sangoma’s offering, the transition from PSTN and ISDN to VoIP can provide a long-term, cost-effective solution.

Why remove a PBX that gives you everything you need? Sangoma’s Vega Gateways provide seamless connectivity to SIP providers, with no need to change any existing hardware or to re-cable your system, these gateways are designed to migrate from PSTN to SIP with minimal downtime.

Available in a range of varieties, Sangoma Vega Gateways suit businesses of all sizes from the SME to enterprise corporations. What’s more, users can benefit from advanced functionalities such as least cost routing (selecting the cheapest route for a call) and enhanced network proxy features which can help with the failover of calls from VoIP to PSTN in the event of a loss of internet connection.

If you would like more information or advice on what is the best option for your business please call or email us. Unlike many we have ISDN simulators so that we can configure your new gateway and install it with limited disruption to your business

Categories
Blog Knowledge Base

Recording Announcements in FreePBX 13 and later

This was recorded a while ago as an aid to a customer, Its a short video on recording prompts and then adding them to an announcment so they can be used in call flow.

Categories
Conference Phone Products Special Offers

Snom C520 Conference Set

Thanks to advanced DECT and Bluetooth technology, the C520 is the perfect device for team conferencing and offers three cutting-edge microphones. The first of these is built into the high-performance full-duplex speaker in the main base station with two further wireless DECT microphones that can be freely placed or carried in the room as required.

Snom C520

The C520 microphones use dynamic noise reduction and adaptive feedback control to provide crystal clear HD audio transmission, even in crowded or spacious rooms. The microphones synchronize with the base station in real-time according to their location. This technology allows the user freedom of movement without even having to raise their voice – regardless of meeting room size.

Connectivity is now one of the most important factors for effective collaboration. This is why the C520 features a versatile Bluetooth interface allowing smartphones or DECT phones to be easily paired and used.

To extend the C520 conference system across even larger areas, simply couple wireless C52 loudspeaker units to the base unit and the range of the loudspeakers and microphones will be increased greatly.

A compact device at an Special introcutory price of £225 +Vat and Delivery

Categories
Handsets Products

Gigaset N870IP DECT

The Gigaset N870 Multicell System is an innovative breakthrough solution that combines DECT base stations and a DECT manager to create a highly scalable yet reliable solution for organisations that need guaranteed mobile communication across sites of any shape or size. Each DECT multicell system supports up to 250 handsets and 60 simultaneous calls making it ideal for small and medium organisations.

For those companies which go beyond the SMB level, widen the network with multiple DECT multicell systems up to the level of an Enterprise, where seamless handover and roaming is applicable in and between all zones. With seamless connectivity with leading IP based on-premise and cloud telephony services, HD audio quality, seamless handover and roaming, the Gigaset N870 Multicell System provides complete coverage across even the most challenging sites. Simple deployment with absolute reliability is key.

The Gigaset N870 IP PRO DECT system offers the latest innovative breakthrough in DECT technology. Combining DECT base stations and a DECT manager in one, the N870 removes the need to purchase a seperate DECT manager, making it a more economical solution.

Offering a highly scalable solution each N870 multicell system supports up to 250 handsets and 10 simultaneous calls per base station, ideal for growing businesses. What’s more, users can rest assured that with continuous connectivity, HD audio quality, seamless call handover and roaming, the N870 DECT solution provides complete coverage across even the most challenging sites.

Compatible with all recent Gigaset pro DECT handsets, the N870 solution can be designed to suit each individual application and its users.

Features

  • Seamless in call handover
  • HD audio quality
  • Each DECT multicell system supports:
    • Up to 250 users/SIP accounts/handsets
    • Up to 60 base stations
    • Up to 10 simultaneous calls per base
  • With the Gigaset Integrator (virtual machine) you can combine up to 100 DECT multicell systems to one large handover and roaming domain
  • LEDs on front for status and visual power check
  • Professional zero-touch configuration via auto-provisioning
  • Compatible handsets include:
  • PoE (PSU available to purchase separately)
  • Wall or ceiling mountable

RRP for each Base station is £395.00 please contact us direct for solution pricing.

Full Fact Sheet.