Introduction to PSTN Gateways
Introduction
sipXecs, the Open Source SIP PBX for Linux is based on a fully distributed architecture that allows you to leverage any SIP compliant component anywhere in the network. In particular, PSTN / SIP gateways can be added as autonomous devices (with their own IP address) where sipx provides centralized configuration and management (certain vendor's models), and flexible XML-based call routing including least cost routing of calls and load balancing. Redundancy and HA is therefore achieved easily as an inherit feature of the chosen architecture. The sipXecs system also scales using load balancing mechanisms that rely on DNS SRV records and providfe alternative routes on demand to distributed gateways. In particular, this means that for large PBX systems a large number of PSTN gateways and phones register with a few strategically located sipXecs servers. In contrast to other systems, SIP strictly separates between media (RTP) traffic and signalling (SIP) traffic optimizing performance for both.
For more information on how to add additional gateways and phones refer to:
Supported Enterprise SIP VoIP Gateways
Any SIP compliant gateway is compatible with sipX and should work just fine.
SIP Media Gateways
- AudioCodes TP-260 PCI card, Mediant 2000 media gateway (and analog gateways)
- Epygi Quadro E1/T1
- Cisco 2600, 3600, 5400
- Mediatrix 1204, 1400, 1500, 1600, 2400, 2500, 2600, 1102, 1104, 1124
- Quintum Tenor Series (it is not known if these work properly, pease post a config if you have one)
- Vegastream Vega10, Vega20, Vega25, Vega50, Vega100, Vega400
Residential and SOHO SIP VoIP Gateways
All the gateways and terminal adaptors below are claimed to be SIP compliant by their respective manufacturers and therefore should work with sipX.
- Cisco ATA-186 / 188
- Citel gateway allows reuse of proprietary PBX phones in a SIP environment
- D-Link 1402S/L VoIP Router with 2 phone ports
- D-Link DVG-1120 VoIP Gateway with Two Voice Ports
- Fritz!Box Fon from AVM
- Grandstream HandyTone 286/486 Voip TA and IAD
- Linksys PAP2 Phone adaptor with 2 ports for VoIP
- Linksys RT31P2 Broadband router with 2 phone ports
- Linksys WRT54GP2 Wireless G broadband router with 2 phone ports
- Sipura SPA-2000/3000 phone adaptor
- Uniden DTA-200 VoIP adaptor
- DrayTek Vigor 2600V(G) ADSL Router with VoIP
- ZyXEL P-2602HW POTS, ISDN VoIP IAD
- ZyXEL P-2602HWL POTS, ISDN VoIP IAD with PSTN lifeline
- ZyXEL P-2002 VoIP ATA
- ZyXEL P-2002L VoIP ATA with PSTN lifeline
- ZyXEL P-2302R VoIP Station Gateway
Digium and Sangoma gateway cards
Digium and Sangoma cards at present cannot be used with sipX as there is no SIP driver software available. However, a PC running the Asterisk PBX or the Yate Telephony Software in conjunction with Digium or Sangoma cards can serve as a media gateway for sipXecs.
VoIP SIP Firewalls
- Nayika Enterprise SIP Firewalls
- Intertex IX66 SIP SOHO firewall
- Intertex IX67 SIP firewall
- Jasomi PeerPoint Enterprise SIP firewall
- BorderWare SIPassuree SIP firewall
- Ingate 1200, 1400, 1600, 1800 SIP Firewalls
- Sonicwall VoIP Firewall
- Checkpoint VoIP whitepaper