External Sip Connections - Aastra OpenCom 100 Series Mounting And Commissioning User Manual

External SIP Connections

The Telephony: Trunks: Route menu gives you the option to configure a bundle
overflow, which automatically occupies a second line in case of a breakdown or over-
occupancy of the SIP connection. You can also set up your system to route certain
types of calls, such as international calls, to an SIP connection.
Note
You will need a Media Gateway card for SIP telephony.
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 operates
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 110 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
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.
Tip:
• You must log on ("Login") to the SIP registrar before you can use SIP telephony. Use
the OpenCom 100 to manage important information for the registration (user
name and password) of one or more SIP accounts. It is possible to make several
calls simultaneously using a single SIP account.
• A SIP connection causes constant Internet data traffic, so do not use SIP with
Internet access which is paid for according to the time used.
The OpenCom 100 communications system can have a clipboard
for keeping track of the most recently dialled call numbers, opti-
mising en-bloc dialling. To do so, activate the Dial out cache op-
tion for a SIP line or a SIP tie-line bundle.
Voice over IP (VoIP)
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on the system terminal,
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