z
Outbound
Caller
ID
:
The
number
you
want
to
display
to
the
called
party.
z
Without
Authentication
:
If
the
service
provider
doesn’t
require
a
username
and
password
for
this
account
to
register
to
their
server
then
you
can
enable
this
option.
z
Username
:
Username
provided
by
VoIP
Provider.
z
Authuser
:
The
optional
authorization
user
for
the
SIP
server
z
Password
:
Password
provided
by
VoIP
Provider.
Advanced
Options
z
From
Domain
:
Your
service
provider’s
domain
name.
z
Insecure
:
Default
value
is
“port,
invite”
;
“port”
‐‐
Allow
matching
of
peer
by
IP
address
without
matching
port
number;
“invite”
‐‐
Do
not
require
authentication
of
incoming
INVITEs.
z
From
User
:
fromuser=yourusername;
Many
SIP
providers
require
this.
z
Qualify(sec)
:
Asterisk
sends
a
SIP
OPTIONS
command
regularly
to
check
that
the
device
is
still
online.
Default
value
is
2(sec).
z
DID
number
:
Self
defined,
and
can
be
used
to
setup
number
DID.
z
Transport
:
Default
transport
type
for
SIP
messages.
z
DTMF
Mode
:
Used
to
inform
the
system
how
to
detect
the
DTMF(Dual
Tone
Multi
Frequency)
key
press.
Choices
are
inband,
rfc2833,
or
info.
By
default
we
use
RFC2833.
z
NAT
:
With
this
option
enabled,
Asterisk
may
override
the
address/port
information
specified
in
the
SIP/SDP
messages,
and
use
the
information
(sender
address)
supplied
by
the
network
stack
instead.
This
feature
is
often
required
when
there
is
a
firewall
located
between
the
PBX
and
the
service
provider.
z
Context
:
Custom
dial
plan
for
this
trunk,
by
default
it
uses
the
“default”
dial
plan.
Configure
only
if
this
trunk
is
for
branch
office
integration,
so
calls
coming
from
the
other
side
can
dial
out
from
this
IPPBX
trunk
directly.
DO
NOT
change
unless
you
fully
understand
how
this
feature
works.
z
Language
:
You
can
choose
adesired
language
of
the
system
voice
prompts
to
play
to
the
incoming
calls
from
this
trunk.
For
example,
if
the
call
is
not
answered
or
the
user
is
busy
the
IPPBX
system
will
notify
the
caller
to
leave
a
voice
message
in
the
language
you
set.
z
Audio
Codecs
:
Select
the
audio
codec/codecs
the
provider
can
support.
z
Video
Codecs
:
If
the
ITSP
supports
video
calls
then
you
can
enable
compatible
video
codecs
here
for
video
phone
calls.
With
the
exception
of
configuration
options
related
to
your
service
provider
and
your
account
details,
please
do
not
change
the
trunk
advanced
parameters
if
you
are
not
familiar
with
them.
After
the
SIP
trunk
is
successfully
added
you
can
see
it
listed
here
on
this
page.
Содержание CooVox-U100
Страница 1: ......
Страница 46: ...For this example if the caller 02885337096 calls the office number the call will go directly to extension 405 ...
Страница 47: ......
Страница 82: ...Please see chapter 4 8 2 ...
