H.323 lines – 2N OfficeRoute - User manual, 1493 v1.9.0 User Manual
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deleted. For a new line enter the user name according to your VoIP provider.
Password – Password is unnecessary for the internal SIP line and may be
deleted. For a new line enter the password according to your VoIP provider.
Codecs – This parameter is necessary to highlight the codecs to be used (press
and hold CTRL to select multiple codecs). You can shift the codecs to define their
priority.
Add Phone context to REGISTER request – This can be useful in some special
cases. Mostly it is unnecessary to fill in this item.
Register expires – Enter the value according to your VoIP provider.
Unnecessary for the internal SIP line.
Register with proxy – Unselect this checkbox for the internal SIP line. For a
new line check it according to your VoIP provider.
Enable CLIP – Select this checkbox for the internal SIP line.
Maximum concurrent calls – Limit the count of concurrent calls via the
selected SIP line..
Get callee from – This can be useful in some special cases. Mostly it is
unnecessary to change these settings.
Use Diversion header – Tick off to include the Diversion parameter in the SIP
INVITE message, carrying information on the originally called line of call
forwarding occurred during the call.
Choose method for transmiting DTMF characters for the SIP
DTMF method –
line.
–
Tick
off
to
deactivate
the
Don't
send
P-Asserted-Identity
P-Asserted-Identity header in the INVITE message. This header is used for CLI
information transmission even if CLIR (Calling Line Identification Restriction) is
enabled.
– Tick off to deactivate the
Don't send P-Preferred-Identity
P-Preferred-Identity header in the INVITE message. This header is used for
transmission identification of user which has call forwarding enabled.
Check off this item in case the VoIP module is behind the NAT or
Enable NAT –
firewall.
NAT port begin – Set which port will be used first for the RTP protocol.
NAT port range – Set how many ports will be used for RTP packets.
NAT IP address – The public IP behind the PBX VoIP module.
No route code – Enter the SIP cause code for terminating call if the route is not
defined.
SIP TOS/DiffServ Value – Set one of the SIP packet parameters, which sets
priority in packet processing by network elements.
RTP TOS/DiffServ Value – Set one of the RTP packet parameters, which sets
priority in packet processing by network elements.
H.323 Lines
Another signalling standard supported by the IP telephony is a group of protocols called
H.323. It could be set only one H.323 line, but it can transfer more calls at the same
time. This line can log in to the Gatekeeper or communicate directly with VoIP
telephones or other gateways.