User guide

IP Network Connections160
PBX Networking
via the SIP tie line protocol. This makes it possible to automati-
cally 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 Forum systems.
Two licences are required when networking two Forum 523/524
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 Config-
urator 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 sys-
tem, 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 Forum 523/524.
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 another VPN connection is necessary for a
connection via SIP tie line.