SIP Provider A1 Telekom (AT) (kb5160)

The information in this article applies to:

  • SwyxWare 11.00

[ Summary | Information | Links ]


Information und configuration hints for SIP providerA1 Telekom (Austria)


General information

The SIP trunk of A1 Telekom was first tested and released with SwyxWare

The SIP trunk of A1 Telekom is a SIP trunk with user authentication, i.e. you get information about user ID, user name and password from A1 Telekom. The names may 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.


SwyxWare: The SIP Trunk of A1 Telekom can be used without restrictions from SwyxWare on. When using an earlier version of SwyxWare incompatibilities cannot be excluded and use is at your own risk.

SwyxON: The SIP Trunk of A1 Telekom is not suitable for central connection in the data center due to the dependency from the internet access.


When creating the SIP trunk group, select the profile 'A1 Telekom (AT)'. 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 (kb3436).

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.

Number block: +43 1239988776600 ... +43 1239988776699
In this case, the SIP URI must be entered in the format +4312399887766*@*.

When creating the actual SIP trunk, all audio codecs except G.711 can be deselected  in the codec dialog. T.38 has to be deselected as supported codec.

Tested Features
  • Incoming/outgoing national calls
  • Correct Calling Line Identification (CLI) on national and calls
  • CLIP No Screening in accordance with RFC3325
  • Holding and retrieving calls
  • Toggeling between two calls
  • Call transfer
  • Conferencing
  • Mutually supported codecs:
    • G.711a
  • Incoming/outgoing DTMF signaling in accordance with via RFC2833
  • FAX transmission via G.711a

Restrictions: -

Test date
07/09/2018 with SwyxWare

Overview of tested SIP provider


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