Aastra OpenCom 100 Series Mounting And Commissioning User Manual page 167

PBX Networking
priate terminals (see Transparent codec interconnection starting on page 116). In
addition, you can find out what features are supported by the other station via the SIP
tie line protocol. This makes it possible to automatically adapt to the respective other
station.
Currently, there is no generally accepted standard for the protocol used with a SIP
tie line. This means via SIP tie line you can only use connections between
OpenCom 100, OpenCom 1000, Aastra 800 and Aastra 5000 systems.
Two licences are required when networking two OpenCom 100 with a SIP tie line – a
licence for each end point. The number of possible call connections is not restricted
by the licence.
Open the Telephony: Trunks: Trunk group page in the Configurator to configure a
connection via SIP tie line. Create a new bundle and select Access Type "System
access". Select "SIP Tie-Line" under Protocol. Configure the IP address of the other
system, the port number to be used (the same port number at both end points), the
number of possible call connections. Select a VoIP profile for codec selection. Please
note the corresponding help topics in the online help OpenCom 100.
En-bloc dialling only is supported with a SIP tie line as with other SIP connections. To
establish a call connection you have to first end call number entry with the hash key
or wait a certain length of time. This length of time can be defined in the Time to
ready dial out input field when configuring the SIP tie line (3-5 seconds are usual). In
addition, you can activate a cache for accelerated dial out conclusion for the call
numbers most recently dialled with the Dial out cache option.
Note
A SIP tie line cannot be conducted via a NAT connection. A branch connection or an-
other VPN connection is necessary for a connection via SIP tie line.
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