An IP Centrex environment configured in PortaBilling® can include one
or more customer sites. A customer site defines a group of phone lines
that are managed as a single entity and usually placed in a separate office
building.
This allows you to apply certain configuration parameters or service
restrictions to the accounts in that group.
If a customer has more
than one location you can set the limitations for each location separately.
To do this, create an independent site entry with specific options for
each of the customer’s locations. Then assign these sites to the corresponding
accounts. Note that even if no customer sites are created, there is the
virtual default site in the system that includes all the accounts that
have not been explicitly assigned to a customer site.
Field |
Description |
Site Name |
Name for a group of accounts. |
Limit Simultaneous Calls |
Engage real-time checks of the number of concurrent calls
made by accounts that belong to this site. When the specified
number of concurrent calls has already been established (calls
are in a “connected” state) and the account tries to place another
call, that call will be rejected.
Choose Customer’s default option to use the
values defined at the customer level.
NOTE:
To enable the Limit Simultaneous Calls feature,
activate the send_start_acct option for the corresponding
PortaSIP instances on the Configuration Server. To increase the
features accuracy, activate the allow_reauth
option too. Note that these features may slightly increase the
load on the billing engine. |
Max Number of Simultaneous Calls |
Allow only a specific number of concurrent calls (regardless
of their type, such as incoming or outgoing) for accounts at this
site. |
Max Number of Incoming Calls |
Allow only a specific number of concurrent incoming calls
for accounts at this site. |
Max Number of Outgoing Calls |
Allow only a specific number of concurrent outgoing calls
for accounts at this site. |
Max Number of Forwarded Calls |
Allow accounts of this site to forward a specific number
of concurrent calls. This limit is only applied when calls are
forwarded to external numbers. |
Codec Connectivity Profile |
Select a suitable codec connectivity profile that will be
used for bandwidth allocation calculation. Every new call’s allocated
bandwidth is calculated by considering a negotiated codec and
its parameters to enable full use of the available bandwidth and
block new calls if no more bandwidth is available. |
Max Bandwidth |
This allows you to configure the bandwidth utilization limitation
to ensure that only an acceptable number of calls are allowed,
in order to avoid severe degradation of the sound quality on calls
in progress.
The system plays a 'limit reached' warning when allocated bandwidth
is used up.
NOTE: Playing warning prompts
requires an additional 8 Kbps of bandwidth. Therefore, set aside
a certain amount of bandwidth (about 8 Kbps) when you define the
bandwidth limit. |
Max Incoming Bandwidth |
This allows you to configure the bandwidth utilization limitation
for incoming calls.
|
Max Outgoing Bandwidth |
This allows you to configure the bandwidth utilization limitation
for outgoing calls. |
Location Information |
This allows you to define customer’s permanent location
for geo-IP fraud prevention. |
Allowed Mobility |
- Select Stationary user (constant location)
if this customer is not authorized to make calls from various
countries (e.g. as a residential customer would make calls
from his SIP phone). Calls made from any other country will
be screened.
- The Roaming user (frequent location) option
can be used for customers who travel frequently. In this case,
a change in location would be considered acceptable.
|
Current Location |
Select a user’s permanent location from the country list. |
Additional Info |
This option works if
the user’s Current Location is set to France
and contains the user’s location information that will be conveyed
by PortaSIP® within the “P-Access-Network-Info” SIP header to
comply with EU regulations.
The value for this option must be defined in the GSTNR1R2C1C1C3C4C5XX
format, where:
- GSTN is the network indication,
- R1R2 is the individual carrier code,
- C1C1C3C4C5 is the city code of the call’s origin, and
- XX are auxiliary digits (00 by default).
|
Dialing Rules |
The following options
are available:
Disabled – This deactivates the dialing
rules for accounts within this site.
Enabled – This activates the dialing
rules for accounts within this site.
Custom – This allows users to create
and use their own sets of dialing rules.
|
Dialing Format |
Select
existing dialing format defined earlier or create a new one by
clicking the Wizard
icon. You can define new dialing rule settings on the IP
Centrex page. |
Translate CLI on outgoing calls |
Allows outgoing calls
to be translated based on the selected dialing format. |
Translate CLI on incoming calls |
Allows incoming calls
to be translated based on the selected dialing format. |
Sip Contact |
Enable this feature to
define the way a customer’s IP PBX registers to PortaSwitch®. |
Use Registration of Account |
This option is usually
selected if a customer’s IP PBX uses a dynamic IP address for
registration.
When selected, it allows you to specify the account used for IP
PBX registration as a SIP contact on the PortaSIP® server. The
calls arriving to any DID numbers that belongs to the IP PBX and
being provisioned as accounts in PortaBilling® will be routed
to the IP address that is currently registered. |
Account |
Select the account used
for IP PBX registration on the SIP server. Its contact information
(i.e. IP:port) will be used by the PortaSIP® server to deliver
incoming calls to the IP PBX directly. |
Static Address |
Enable this feature if
the customer’s IP PBX can’t
perform SIP registration to PortaSwitch®. |
Use Original CLD |
Specify the destination
number that calls will be routed to. If left blank, the number
originally dialed will be used as the destination number. |
Host |
This contains the destination
host the calls will be routed to. A customer’s IP PBX can be identified
with one of the following options:
A valid IP address (four numbers separated by points,
e.g. 12.34.56.78).
A valid domain name (e.g. pbx.example.com).
A valid domain name with configured DNS SVR records. In this
case, PortaSIP® will round-robin through them. |
Transport |
Select the transport
protocol (either TCP or UDP) that is used to deliver incoming
calls. |