How we tested open source IP PBX

The open source-based IP PBX products were connected into our standard IP PBX test bed, which was configured with a subnet using an Extreme Networks' Summit 24 L2/L3 switch simulating a typical enterprise location.

The Extreme switch was connected to the Internet via a Cisco 1601 router. Multiple hard phone and softphone endpoints were registered to each system.

The endpoints used were Polycom 650, 501, 430 and 301; Snom 360, 320 and 300; Thomson ST2030; Cisco 7960; CounterPath EyeBeam 1.5 retail Session Initiation Protocol (SIP) softphone; X-lite freeware SIP software; PortSIP VOIP SDK freeware SIP softphone; Phoner and PhonerLite freeware SIP softphone; and the Asteriskguru IDEFISK freeware IAX softphone.

Each IP PBX was installed, all supported endpoints were provisioned accordingly, and accounts were created within the respective IP PBX system. Each account was set up as a standard SIP or Inter-Asterisk eXchange (IAX) (if supported) device. If the system supported plug-and-play capability, then no account was created and the endpoints were directly added to exercise this feature. Each account was created with voice mail, conferencing, call forwarding, user name, extension and other standard features.

The softphones were installed on a pair of Dell Latitude D610 laptops with 1.7GHz processors. Each softphone was configured with a display name, user name, password, authorization user name, and an authenticating server IP address. The softphones were used to make multiple SIP or IAX calls between all supported endpoints. The endpoints that supported the use of presence also were configured and tested.

All the endpoints were successfully provisioned and connected to all IP PBXs except for in two cases. Fonality requires phones be provisioned by them, so only the Polycom and PBXtra Eyebeam Softphone were reviewed. The other IP PBX system in question here, Pingtel, does not support IAX in a back-to-back user mode.