Aastra OpenCom 510 User Manual page 123

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Voice over IP (VoIP)
characters after "@" is the "Domain Name" or the "SIP Domain". The IP address
of the OpenCom 510 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 510. Switch these functions off if possible.
The "G.711 µ-law" codec (North America, Japan) is not supported. If the SIP tel-
ephone offers such a setting, you should switch on the "G.711 A-law" (Europe)
instead.
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 510 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 510 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 dis-
play (CLIP)
Parallel connection of
multiple SIP subscribers
Enquiry, toggling, call
waiting, three-way con-
ference, reject
Notes
A SIP telephone requires a call number display for
CLIP.
SIP subscribers must be logged on under the same
user identification.
Operation or feature must be available on the SIP tel-
ephone or in the SIP software.
SIP Telephony
121

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