The sipXecs IP PBX is adding native SIP trunking support. Release 4.0 of sipXecs, due out early 2009, will include a complete and fully featured SIP trunking solution. Testing is on-going and we are looking for ITSPs interested in participating. The list below identifies those ITSPs we have tested so far and intend to release as supported in that upcoming release 4.0 of sipXecs. The sipXecs project aims at becoming fully SIPconnect compliant, however, we also chose a pragmatic approach making SIP trunking work with what ITSPs can provide today.

How to tell the sipXecs team that you want to be part of it?

Subscribe to the sipx-dev mailing list and announce yourself. Someone will be happy to help. More information about how it works is available on the sipXecs Wiki. To try it out download sipXecs from the project server's main branch or 3.11 release (this is a development release that is not yet released even as Alpha).

Regression testing:  This link takes you to the sipXecs IP PBX regression test plan .

North American ITSPs				European ITSPs

 What is SIP Trunking?

Instead of using a traditional service provider to connect to the telephone network, you send all your calls over IP to an Internet Service Provider (ITSP). This saves you the cost of a PSTN gateway and instead you only need a data connection, which you most likely already have. Typically there is no difference in voice quality, however, you have to make sure that there is sufficient bandwidth available on the WAN side to accommodate all your voice calls and data traffic. It helps if your router or modem can assign quality of service priority to packets that carry voice traffic. sipXecs allows you to traverse local Network Address Translatoin (NAT) firewalls and makes sure all signalling and media is getting through. All you have to do is open certain ports on your firewall and direct traffic on these ports to sipXecs. sipXecs takes care of security and handles all that traffic approprietly.