VoIP standards such as SIP or IAX2 directly in the Intercom Server. Via the integrated web interface, easy remote configuration
is possible.
• Asterisk® support onboard for direct connection of SIP phones to the Intercom System
• Up to 10 SIP trunk lines with a total of 8 simultaneous speech channels
• Up to 2 IAX trunk lines with a total of 8 simultaneous speech channels
• Up to 50 SIP users (i.e. SIP phones to be registered on the ”SIP Registration Server” of the card)
• Voicemail is available for each SIP phone
• Web based Asterisk®GUI for maintenance and remote configuration
• SSH (secure shell) for maintenance and remote configuration
• Supported audio codecs additionally to G.711 a-law:
– G.711 u-law, gsm, G.722, adpcm, slin, lpc10, G.726, G.726aal2, iLBC
– “pass-through only” (no recoding): speex, G.723.1, G.729A
• Simultaneous speech channels:
– G.711 a-law: 8 simultaneous conversations
– G.711 u-law: 8 simultaneous conversations
• Adaptive jitter buffer
• Supported protocols: SIP, IAX2, RTP, UDP, TCP, NTP, SMTP, HTTP, HTTPS, DNS, DHCP,
TLS/SSL, SSH, SFTP, ICMP and more ...
• One-click backup over web interface
• Automatic transmission of the caller-ID as text indicator at SIP phones
• Direct dialling to Intercom stations possible
• Pre-recorded audio e.g. for waiting information