Teams Connector

FAQ About Our Phone Lines

You have set up the parallel call function and lose calls if they are signalled in parallel on a mobile phone number? In the following, we will explain what the problem is.

Many mobile providers do not use “Early Media”. This means that if the mobile connection is not available, the respective provider accepts the incoming call and switches the band announcement via RTP. This means that the parallel call is also accepted and no longer displayed on other devices, even though these may be “free”.

If, however, Early Media were used, we would be able to see from the SIP feedback that the call was not accepted. This would ensure that the incoming call would continue to be signalled on the registered device.

Another problem caused by the lack of Early Media: The call accepted by the recorded message is charged to you.

The following mobile operators do not use Early Media: All mobile operators using the Vodafone network.

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Information about which telephone systems are suitable for your SIP Trunk.

All IP-PBXes which can cope with SIP, the technologies of SIP-DDI and CLIP no screening and Codec g.711 are suitable for easybell Business VOICE.  For HD calling, g.722 is necessary.

There are lots of different IP-PBXes. Please make sure, you take one for your needs. Therefore, you have to think about the needed features. A FRITZ!Box is often fine for SIP trunking, because it is able to deal with up to 20 extensions. To make sure you'll find the best suiting solution, we offer an interactive comparisson of PBXes on our website (in German).

To get the full functionality, we recommend you to use one of our certified IP-PBXes. We made extensive tests in cooperation with the PBXes’ producers to guarantee a full compatibility with our SIP trunks.

You can find different guidances to set up our certified PBXes in our help center.

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The right upload speed is often crucial for trouble-free calls via Voice-over-IP. In this article, you will learn how much bandwidth is required per voice channel, why a security buffer is important and how to calculate the requirements for multiple parallel calls.

For calls to the public telephone network, you should plan for at least 80 kbit/s per voice channel - regardless of whether you are using an IP phone, a local telephone system or a cloud PBX.

The actual bandwidth may vary depending on the codec, encryption and network quality. You should therefore plan for 20-30% additional bandwidth to compensate for fluctuations and parallel data transmissions.

Practical example: For 10 simultaneous VoIP calls, you need at least 1 Mbit/s upload speed.

Our VoIP offer can be used worldwide - from any Internet connection that meets these requirements.

Extra tip: Your broadband connection is not charged for internal calls via a local telephone system.

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You can register five devices at one SIP account. 

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You can see if you still need a Telekom telephone connection here:

No. For telephony and Internet, you only need the complete package from Easybell.

Easybell also offers IP telephone connections without Internet. In this case, you must book an Internet connection. Whether DSL, cable, fibre optic (FTTH) or even 5G does not matter as long as sufficient bandwidth is available.

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To use your internet or IP phone service, you'll need your login credentials. In this article, we'll explain when and how you'll receive them.

If you have ordered an Easybell complete connection, you will always receive your DSL login credentials by email no later than the activation date. For security reasons, we do not send out the login credentials for VoIP telephony. You can view these in the customer portal.

If you have ordered a router from us, it is already preconfigured. All you need to do is connect it.

You can also find all access data at any time in my.easybell after activation.

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easybell routes incoming telephone calls via the routing partner QSC/Plusnet, which as a competent carrier can not only ensure very good accessibility from all networks, but also high voice quality. This makes it possible for easybell to meet the demand for high-quality coverage.

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Find out here which audio codecs are supported for internet telephony.

Internet telephony supports all important audio codecs so that the best voice quality can be guaranteed at all times.

  •     G.711 A Law and G.711 U Law - packet sizes of 5, 10, 15, 20, 25, 30, 35 and 40 ms
  •     G.722 (HD) for pure SIP calls
  •     T.38 (Fax)
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The following article shows to what extent you can use SRTP with your easybell connection.

RTP stands for Real-Time Transport Protocol. In VoIP telephony it is used, for transmiting audio signals e.g. the spoken word. Based on this, the Secure Real-Time Transport Protocol (SRTP) additionally encrypts these data streams.

easybell supports SRTP according to RFC 3711.

You can transmit the RTP communication on the route between your terminal device and our servers using SRTP. This means for you that your calls with other easybell customers are encrypted - provided that all participating terminals are correctly configured and support encryption.

If you use our cloud telephone system, you can even have terminals from Snom, Htek and Yealink automatically configured for encrypted telephony via ACS. The necessary settings for manual setup can also be found in our VoIP expert settings.

However, calls that are routed beyond our network boundaries are only accepted unencrypted by the other network operator.

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