General
The SIP Trunk from Peoplefone (DE) was first tested and released with a SwyxWare 12.30.0.0.
The SIP Trunk from Peoplefone (DE) is a SIP Trunk with user authentication, i.e. you get information about user ID, username and password. In this case, the names may well differ, or you will only receive a name or ID with password. In this case the user ID is identical to the user name when configuring the SIP Trunk in SwyxWare.
Platforms
SwyxWare
The SIP Trunk of Peoplefone (DE) can be used with SwyxWare 12.30.0.0 or higher. If you use an earlier version of SwyxWare, incompatibilities cannot be excluded and you are responsible for your own usage.
SwyxON
Der SIP Trunk von Peoplefone (DE) ist nicht für die zentrale Anbindung im Rechenzentrum geeignet.
Configuration
When creating the SIP trunk group, select the profile 'Peoplefone (DE)'. If the profile is not included in the corresponding list, you can download the required CustomProviderProfiles.config for SwyxWare at the end of this article. For more information about using this file, see Configuration and Adaptation of SIP Providers via Provider Profiles.
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IMPORTANT: The profile 'PeoplePhone TCP (DE)' already existing in earlier versions of SwyxWare cannot be used due to incompatibilities. |
SIP URIs
In order to assign incoming calls to the correct trunk, it is necessary to configure a SIP URI for the trunk. The assigned number block and the numberformat must be considered.
Example:
Numberblock: +49 40 123456780 ... +49 40 123456789
In this case, the SIP URI must be entered in the format 494012345678*@*.
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.
Tested Features
- Incoming/outgoing national /international calls
- Correct Calling Line Identification (CLI) on national / intenational calls
- CLIP No Screening
- Holding and retrieving calls
- Toggeling between two calls
- Call transfer
- Call forwarding
- Conferencing
- Mutually supported codecs: G.711a, G.711u, G.729
- Incoming/outgoing DTMF signaling in accordance with via RFC2833
- FAX transmission via G.711
Restrictions
None
Test date
22/01/2021 mit SwyxWare 12.30.00.0
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. Swyx in no way guarantees the accuracy of this third-party contact information nor is responsible for it's content.
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