Support FAQ

SIP trunks will dramatically reduce your calling costs by routing your calls over an IP network as opposed to the traditional public telephone network which uses analog or ISDN lines. There are also several other advantages. For example, your telephone numbers move when you do, and you can answer a London call in Belfast.

With the correct configuration you can expect quality equivalent to that of an ISDN line.

Voiceflex itself is highly reliable. At the time of publication, we have had no interruptions to service since our first server went live more than three years ago. The only threats to availability are in the network connection between your premises and our data center. Either a line fault on your ADSL or a core network fault can cause an interruption of service. These risks also apply to traditional telephone trunks, so the SIP service has reliability comparable with traditional phone lines. It is also necessary that your PBX and ADSL router have power, so without backup power, an electricity outage will imply a phone outage at the same time.

Yes, calls to 999 and 112 will connect you to the emergency services. However this is subject to your phone and internet equipment having power. Your interconnection connection must be switched on and working. If this is not the case, you will need to use a mobile or analog phone to call.

While there are many providers in the UK, Voiceflex leads the way for business customers with PBXs using multiple trunks. We have tested extensively with PBX manufacturers and can provide a smooth transition from ISDN to SIP.

You need a SIP enabled device (PBX or handset) and a broadband internet connection, such as ADSL. We recommend the use of our Business Connect broadband product, which includes both the connection and the router.

Yes, old number can be ported so that you can receive incoming calls via SIP.

New numbers can be ordered on our web-portal. You can assign as many numbers to each SIP account as you want. Select the code required and you'll be assigned a number local to that area.

It is important to configure your device (PBX or handset) correctly. You can obtain the configuration details for your SIP trunk from the portal. You may also be having problems with NAT as implemented by your ADSL router. If you feel you have configured everything correctly but it still doesn't work, please feel free to contact our support team who will assist you in diagnosing and repairing the problem. You can email support@voiceflex.com or call 0203 301 6000.

Network address translation (NAT) is a technology which allows several computers and devices inside your network to connect to the internet, despite having private IP addresses and only a single shared public address. This public address is the external address of you NAT device, and the NAT device is most often your ADSL router. The NAT device also serves as a network firewall. SIP and VoIP are not always allowed through this firewall, and it is often necessary to adjust the configuration of the SIP and/or NAT device in order to obtain correct operation of the VoIP service. More information is available in this white paper, this IETF draft, or by contacting our support team.

No, not currently. Fax over SIP is a very new technology. We are monitoring developments and plan to add this feature to our service in the future. We provide a fax-to-email service for customers where the fax number is part of the DDI range.

One way audio is most likely due to problems with NAT traversal. More information is available in this white paper, this IETF draft, or by contacting our support team.

DTMF can be sent in one of many ways. By default, we support and expect DTMF using RFC-2833. It is possible that your system uses a different convention. To resolve this, review the documentation for your PBX, where you should find details of the method used to send DTMF. If it is configurable on your PBX, please set it to use RFC-2833. If not, please contact our support team and let us know which method your PBX supports and we will configure our system to recognise that method for your SIP trunk.

Voice quality is affected by several factors. The most important of these are packet loss, latency, and jitter. These are properties of the network between your device and us. If you are using our Business Connect service to connect, we are able to monitor and control these parameters, and you should contact us immediately if you experience degradation in voice quality. If you are not using Business Connect, we are unable to do anything about your connection and you should contact your ISP for assistance. Please note however that most ISPs will not see high latency or jitter as faults since these factors do not affect internet usage, and they do not treat voice differently to internet usage.

Simple Traversal of UDP through NAT (STUN) is a network protocol used by devices behind NAT in order to discover their public IP address and port, as well as the type of NAT which is being used. If your device requires you to enter STUN server settings, please refer to the configuration information provided in the portal, or contact our support team.

By default we support the high quality G.711 codec using μ-law or A-law. We can also enable G.729 on request. Please contact our support team if would like G.729 enabled.

If you are having trouble connecting, or experiencing bad call quality, you can accelerate your support request by supplying our support team with a network trace. This should only be attempted by users with technical knowledge. It can be done with the free Wireshark program. Install the software on the computer on which your PBX server or soft-phone is installed, or alternatively on a computer which shares a hub with your PBX. Read the user-manual and familiarize yourself with the privacy implications before using this software. Run a trace on port 5060 (connections problems) or all ports (quality problems) while attempting a call. Save the trace in PCAP format and attach this to an email to the support team.