Field Description
particular provider (SIP account), you must configure a cor-
responding call routing entry. Internal calls (from internal ex-
tension to internal extension) that are only to be routed intern-
ally do not require an additional call routing entry.
• A* (37B: Select a SIP trunk account configured under
VoIP->Media Gateway->SIP Accounts. In this case, the call
routing for all extensions is handled by the session border
controller, all SIP messages are forwarded to the session bor-
der controller. Note that the call routing is handled by the me-
dia gateway if the provider is not available (backup).
Please note: Entries in Call Routing have priority ahead of the
session border controller configuration!
Media Stream Termina-
tion
Choose how RTP sessions are controlled by the system.
If the function is enabled, RTP sessions are terminated on the
media gateway, i.e. all RTP streams are controlled by the media
gateway and routed via the media gateway. The participating
terminal devices (e.g. SIP telephones) are not connected dir-
ectly with one another. Note that, for VoIP to VoIP connections,
there is no code translation for different VoIP terminal codecs.
The codecs of media gateway and VoIP terminals must there-
fore agree.
If the function is disabled, RTP sessions are not terminated on
the media gateway, i.e. all RTP streams are routed by the me-
dia gateway without termination. The RTP data packets can be
routed in complex networks and thus also via other gateways.
The function is enabled with ,.
The function is enabled by default.
Default Drop Extension You can specify an extension to which incoming calls are for-
warded if they cannot be assigned to an extension or connected
PABX.
Dial Latency Enter the maximum delay time before the system assumes the
call number entered is complete and starts the SIP dialling pro-
cess (sends the SIP INVITE message). This timeout is reset
each time that a button is pressed.
Possible values are  to .
20 VoIP Teldat GmbH
426 bintec Rxxx2/RTxxx2
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