- Telecom Services
- User Guides
We provide a high quality SIP Trunking service connecting your PBX directly into our network, via an DSL or Ethernet IP connection to receive and terminate your voice calls over the public telephone network.
This service provides a highly flexible alternative to ISDN and has been fully tested with many leading IP PBX applications such as Asterisk, Avaya, 3CX, Freeswitch and SNOM.
Our free SIP trunking service offers access to our industry beating call rates with an unlimited number of concurrent channels and built-in PSTN failover.
The following voice and video codecs are supported: PCMA/U, G722, G722.1, CELT, GSM, Speex, iLBC and T.38 fax. Our network supports both PBX equipment and individual SIP Phones as well as VoIP Software packages on PCs. We also support NAT Traversal, Voice encryption and access from mobile phones.
We provide high strength encryption of both signalling and media to suit the needs of government and public sector operations, compatible with most commercial IP PBX and SIP phones.
If your equipment needs to know where calls to your trunks will originate from on our network, all calls are sent from our Session Border Controller on the following IP address: 18.104.22.168.