IP Network Connections
234
PBX Networking
Connection via SIP Tie Line
The Forum 5004/5008/5012 communications system supports
connections using a SIP tie line for TC system networking. A SIP tie line is a SIP
line which requires no login which can establish multiple call connections
simultaneously. No SIP provider is required for establishing the connection via
SIP tie line.
Note:
A network connection between two Forum communications
systems via SIP-Tie-Line requires at least one license
(Forum SIP Tie line). The number of purchased licenses defines
the number of simultaneous calls.
The number of simultaneous calls depens – apart from the purchased
number of licenses – also depends on the network or internet connection
capacity and the compression procedure. The data of a SIP tie line connection
is subjected to codec compression (see under
page 155 in the chapter
). Call data is transmitted directly
from terminal to terminal via the RTP protocol with SIP tie line as well. In
certain cases, for example when an incoming external call is switched via
multiple TC systems, there may be one or multiple RTP proxies involved.
One of the special features of a SIP tie line connection is using transparent
codec interconnecting, for example to make use of HQ audio or video
telephony with appropriate terminals (see
Transparent codec interconnection
starting on page 160). 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.
You can use SIP tie line connections between Forum communications
systems. With SIP tie line connections to other manufacturers systems only
basic call features may be used
ForumOpen 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 Forum 5004/5008/5012.
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