General
The Voice SIP Trunk from 1&1 Versatel was first tested and released with a SwyxWare 13.27.
The Voice SIP Trunk from 1&1 Versatel is a SIP Trunk with User Authentication, i.e. you receive information about UserID, UserName and UserPassword from 1&1 Versatel. The names can be different, or you receive only the user name and password. In this case the UserID is identical with the UserName.
Platforms
SwyxWare
The Voice SIP Trunk from 1&1 Versatel can be used according to the settings listed below from SwyxWare 13.27 on. Due to incompatibilities it is not possible to connect an earlier version of the SwyxWare to the The Voice SIP Trunk from 1&1 Versatel.
SwyxON
The Voice SIP Trunk from 1&1 Versatel is not released for central connection in the data center.
Configuration
During creation of the SIP trunk group, profile '1&1 Voice SIP (DE)' must be selected.
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IMPORTANT! Due to incompatibilities between the Voice SIP Trunk and SwyxWare, a connection to the Voice SIP Trunk variant with encryption is currently not possible. |
Codec Configuration
During creation of the actual trunk, in the codec configuration dialog, T.38 has to be deselected as supported codec. This step could also be performed afterwards in the properties of the SIP trunk, underneath the 'Codecs' tab.
SIP URIs
When creating the actual SIP trunk, a SIP URI must be entered after configuration of the phone number range so that incoming calls can be assigned. The assigned number block must be taken into account in the correct format.
Example:
Number block: +49 231 9988776600 ... +49 231 9988776699
In this case, the SIP URI must be entered in the format +4923199887766*@*.
CLIP No Screening
To activate the feature CLIP No screening in the SwyxWare, it is necessary to set the option "Always use originator's number" in the tab "Number signalling" of the SIP trunk settings.
Tested Features
- Incoming/outgoing national calls
- Incoming/outgoing international calls
- Correct Calling Line Identification (CLI) on national and international calls
- CLIP No Screening
- Holding and retrieving calls
- Toggeling between two calls
- Call transfer
- Conferencing
- Mutually supported codecs:
- G.722
- G.711A
- G.711u
- G.729
- Incoming/outgoing DTMF signaling in accordance with via RFC2833
- Fax transmission via G.711
Restrictions
None
Tests
13/06/2023 with SwyxWare 13.27
Disclaimer
Verifying interoperability between different SIP providers and the Swyx solution is part of the Enreach Technology Alliance Programme (TAP) tasks. Due to different implementations of the SIP RFCs, it is necessary to determine and provide an individual configuration for each SIP provider.
Within the TAP, Enreach tests a large number of national and international SIP providers in order to make statements about the interoperability with the Swyx solution and the functional scope of the respective SIP trunk. These results and, if needed, configuration advice are published in Help Center articles.
Enreach is in close contact with selected SIP providers to ensure a quick troubleshooting in case of incompatibilities. In addition, tests are repeated at regular intervals to ensure consistent quality.
Despite regular retests it cannot be excluded that incompatibilities with the Swyx solution may occur due to changes in the SIP connection on the provider side or that the provided profile is no longer valid. After creating the trunk group or SIP trunk, the profile settings should be compared with the provider's information and adjusted accordingly in case of deviations.
However, should incompatibilities or problems occur with a tested SIP trunk, no claims for recourse against Enreach GmbH can be derived from this. In such cases, you can contact Enreach Support via your specialist dealer.
The third-party contact information included in this article is provided to help you find the technical support you need. This contact information is subject to change without notice. Enreachin no way guarantees the accuracy of this third-party contact information nor is responsible for it's content.
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