Table of Contents
Advertisement
Voice over IP (VoIP)
characters after "@" is the "Domain Name" or the "SIP Domain". The IP address
of the OpenCom 100 must always be used here.
■
A STUN server (Simple Traversal of UDP over NAT) or a SIP proxy is not required
because internal SIP subscribers on the LAN are directly connected to the
OpenCom 100. Switch these functions off if possible.
■
With a SIP terminal, you can enter an international phone number with a
leading plus. When you enter a call number in the E.123 format, the plus char is
substituted by the "00" number sequence and the immediate line seizure via
the standard route is activated for the call. If you prefer to dial in this number
format, you should activate the international call number conversion (see E.164
conversion starting on page 163).
Features
SIP subscribers can establish connections to all other terminals and trunks. The SIP
protocol generally works with block dialling. This is why the selected call number
is only activated after an internal clock has expired or activated immediately via
the hash key ("#") when dialling. This is why code number procedures without the
hash key and code number procedures with a concluding hash key can be used.
An overview of code number procedures that can be used is located in the Config-
urator on the System Info: Codes page. Activate "SIP phones". Please also note
the corresponding information in the "OpenCom 100, Operation on Standard Ter-
minals" user guide.
Alongside code number procedures, SIP subscribers can also use a series of func-
tional features realized via the SIP protocol. The OpenCom 100 is the ending for all
SIP connections as opposed to what is usually the case on the Internet. This
enables SIP subscribers to use OpenCom 100 features. Direct data exchange is
thus not possible between two SIP subscribers. The following table shows the pos-
sible features.
Features
Incoming and outgoing calls with call
number display (CLIP)
Parallel connection of multiple
SIP subscribers
Enquiry, toggling, call waiting, three-
way conference, reject
Notes
A SIP telephone requires a call number
display for CLIP.
SIP subscribers must be logged on un-
der the same user identification.
Operation or feature must be available
on the SIP telephone or in the SIP soft-
ware.
SIP Telephony
119
Advertisement
Table of Contents
