Sip Telephony - Aastra OpenCom 510 User Manual

Voice over IP (VoIP)
3. Enter the desired IP address in the IP address configured field. Enter "0.0.0.0"
to obtain an IP address via DHCP.
4. Confirm with Apply.
The configuration page displays the MG+ETH1-1 interface card's MAC address. You
will need this for the static assignment of an IP address via a DHCP server.
10.3

SIP Telephony

The SIP Internet (Session Initiation Protocol) protocol provides you with a low-cost,
standardised option for telephoning via the Internet. OpenCom 510 enables you
to use external SIP telephone connections (SIP trunk lines). The PBX Configu-
ration: Trunks: Route menu gives you the option of making an SIP connection on
a normal ISDN line in case of bundle overflow, breakdown or over-occupancy. You
can also set up your system to route certain types of calls, such as international
calls, to an SIP connection.
You will also need a fast Internet connection such as DSL for SIP telephony.
You will also usually need the services of a SIP provider. A SIP provider operates a
special server (the SIP Registrar) to handle connections. The SIP provider also ope-
rates a gateway to the ordinary telephone network which users pay to use and
which enables the SIP provider to provide calls to the telephone network.A SIP
connection can also accept incoming calls from the telephone network.
The same voice transmission techniques as those explained in Fundamentals
starting on page 112 are used for SIP telephony. SIP telephony has the following
distinctive features:
■
Subscribers are identified through an e-mail-like "SIP ID" such as
12345@domain.net or name@sip-provider.com.
■
SIP transmits dialling numbers always in a single data package (block dialling).
Dialling can therefore be concluded with the hash key
minal, or the end of the number will be indicated by a time-out. The value for
this time-out can be defined for each SIP provider separately.
120
Note: You will need a Media Gateway card for SIP telephony.
SIP Telephony
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on the system ter-