Mediant 3000 Setup Guide
Mediant 3000 Setup Guide
Setup Guide
Release 1.0
March 2011
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Contents
Mediant 3000 has a compact footprint (2U) that meets needs of both service providers with
geographically dispersed networks, as well as those of large enterprises, where reliable and
dense VoIP gateways are necessary for business-critical communications.
Assigning an IP address
Mediant 3000 HA has two TP-6310 blades (Active and Redundant). Each blade has a private IP
address (different from one another) for maintenance and initial configuration. When you initially
configure the blades (using BootP/DHCP), the BootP server communicates with the two blades
using these two IP addresses. In addition to these private IP addresses, the device has a global
IP address used by the Active blade for communicating with the IP network. This global IP
address is configured using the.ini file, Web, or EMS management tools and is a valid IP
address that is different from the two private IP addresses of the blades, but in the same subnet.
From the perspective of the WAN interface (for example: proxy server), the product has a single
IP address (the global IP address); the redundant blade is a transparent entity.
Needed equipment
Mediant 3000 Media Gateway is shipped with the following equipment:
● Mediant 3000 Media Gateway, which at a minimum includes the following components:
- Mediant 3000 chassis.
● A Linux-based server for the Element Management System (EMS) server software.
- Linux CentOS 5.3 Rev 4.
- 1 console cable for network administration.
- 1 DVD - 5.8.90 for EMS Server.
- 1 DVD software Installation & Documentation 5.2.x (media gateway software, EMS
client and server software).
● A standard 19-in. (48-cm) four post equipment rack that is properly installed and solidly
secured to EIA-310D or equivalent standards. The rack cabinet must have adequate
ventilation.
● DC/AC power supply.
● Power cable from gateway to DC/AC power supply. Wiring needs to be stripped and
connectors attached on site.
● CAT5 Ethernet cable (gigabit).
● T3 cable to connect the Trunk Processing Modules (TP-6310s). This cable requires a
female mini-SMB connector.
● One incoming twin fiber optic cable with Dual-LC plug for each STM-1 optical interface.
● Avaya S8720 in XL configuration or S8730 Server with a DAL2 board installed, configured,
and operating. Only hardware duplication supports Mediant 3000 Media Gateway.
● Avaya G650 Media Gateway installed and operating. The media gateway must have an
adequate number of TN799DP C-LAN circuit packs and TN2302AP IP Media Processors
or TN2602AP IP Media Resource 320 circuit packs.
Mediant 3000 Media Gateway comes with
● 1 or 2 Trunk Processing Modules (TP-6310s).
- The second TP6310 functions as a standby module for redundancy.
● 2 SA/RTM for System Controllers, 2 RTM for TP-6310 (Active/Redundant type depending
on configuration). The active TP-6310 RTM has PSTN interfaces, and the redundant
TP-6310 does not.
● Cable to DS3 line (50-ft [15 m]).
● DSX3 broadband patch panel with associated monitoring patch panel jacks and cables
(recommended). The patch panel, collocated with Mediant 3000 Media Gateway, allows
the unobtrusive observation of incoming and outgoing traffic on the DS3 connection.
Contact your Avaya representative for details.
● Laptop computer with EMS client installed for staging
● Filled-out planning form with IP addresses. A blank planning form is available on http:/
support.avaya.com within the product documentation for Mediant 3000 Media Gateway. IP
addresses are required for several items, including certain TN circuit packs, such as
C-LANs, MedPros, and VAL; network servers, such as DNS, NTP, and NMS; and certain
components on Mediant 3000 Media Gateway. The following table provides guidance on
which Mediant 3000 item need IP addresses:
1 TP 6310
2 Alarm boards
3 TP 6310
4 Alarm boards
Pre-configuration tasks
Before beginning the configuration, make sure that you have all the equipment and it is set up to
simulate the customer’s environment as much as possible, including using the customer’s time
zone.
In addition, ensure the following:
● Avaya Mediant 3000 Media Gateway has power. Refer to Installing and Operating the
Avaya Mediant 3000 Media Gateway (03-601918).
● AudioCodes EMS server software is installed on the Linux server. Refer to EMS Server
Installation, Operation, and Maintenance Manual (LTRT-94108) and EMS User’s Manual
(LTRT-91006)
● AudioCodes EMS client software is installed on the laptop. Refer to EMS Server
Installation, Operation, and Maintenance Manual (LTRT-94108) and EMS User’s Manual
(LTRT-91006)
● EMS server has the current software, available from PLDS.
The following sections describe how to configure the SIP and PSTN trunks and call routing for
Mediant 3000 Media Gateway. You configure the media gateway using the GUI-based Element
Management System (EMS). If configuring in a staging area, the EMS client software is
installed on the staging laptop. If configuring at the customer’s site, the EMS client software is
installed on the SPS Server.
The first part covers configuring the network support parameters. The second part covers
configuring the Trunk Processing Module (TP-6310) board, which controls the call signaling for
and routing between the SIP and DS3 interfaces to which it is connected.
Note:
Note: Avaya suggests make changes to Mediant 3000 only through the EMS server.
To add the device to the EMS, log into EMS and adhere to the following main stages:
1. Add a Region
2. Add the software device version file (cmp) to the EMS Software Manager
3. Define the IP address and other initial setting of the device.
1. On the right pane under Regions List, double-click the first row entry.
2. In the right pane under the MGs List, select the entry corresponding to the Mediant 3000
Media Gateway device to be configured.
a. Yes: The device is locked only after the user-defined time in the Lock Timeout field
(refer to Step 3) expires or active traffic does not exist any more. In addition, no new
traffic is accepted.
b. No: The device is locked regardless of traffic. Any existing traffic is terminated
immediately.
3. In the Lock Timeout field, enter the time (in seconds) after which the device locks.
Note:
Note: If no traffic exists and the time has not yet expired, the device locks.
4. Click the Lock button.
The system displays a confirmation message box.
5. Click OK to lock the device.
Note:
Note: If Graceful Option is set to Yes, the lock is delayed and the system displays a
screen showing the number of remaining calls and time. Otherwise, the lock
process begins immediately.
2. Select General Config > Info and Security > General Setting.
The NFAS group can comprise up to 10 T1 trunks. Each T1 trunk is called an NFAS member.
The T1 trunk whose D-channel is used for signaling is called the Primary NFAS Trunk. The T1
trunk whose D-channel is used for backup signaling is called the Backup NFAS Trunk. The
primary and backup trunks each carry 23 B-channels while all other NFAS trunks each carry 24
B-channels.
You can configure ISDN-NFAS Trunks offline or in real time.
b. Select the EXPLICIT INTERFACE ID check box to configure the Interface ID (see Step
d) of a NFAS Trunk. If this field is not set, only the Trunk ID is recognized.
c. Select Primary NFAS Trunk, from the D-Channel Configuration drop-down list, for
the T1 trunk whose D-channel is used for signaling or Backup NFAS Trunk for the T1
trunk whose D-channel is used for backup signaling.
d. In the ISDN NFAS Interface ID field enter the Interface ID (0 - 255) of the trunk in the
NFAS group.
e. In the Group Number field, enter the device's NFAS Group Number. If this field is set
to 0, the trunk is not an NFAS trunk.
f. Click Apply.
g. Use Profiles to apply the configured fields to multiple trunks.
7. Select the General Settings tab to configure each trunk in the group with the same values
for the following parameters:
a. Protocol Type
b. Framing Method
c. Line Code
8. Write and reset the device after configuring all the trunks.
b. (Optional) Enter the Proxy's name. The Proxy name replaces the Proxy IP address in
all SIP messages. This means that messages are still sent to the physical Proxy IP
address, but the SIP URI contains the Proxy name instead. When no Proxy is used,
the internal routing table is used to route the calls.
c. Click the button, and then click Yes to confirm.
d. Enter the IP address of the Proxy Server.
e. Right-click the new entry, and then choose Unlock Rows.
7. Select the Registration tab, and make the following changes:
a. Configure Is Register Needed field:
● No: The device does not register to a Proxy/Registrar server (default).
● Yes: The device registers to a Proxy/Registrar server at the start of power and
every user-defined interval.
b. Click Apply and close the active window.
8. From setup bar, select SIP.
9. Click End Points.
On the SIP EndPoint screen, make the following changes:
a. Click the button to add a new entry, and then click Yes to confirm.
The system displays Phones screen.
b. Double-click each field to enter values.
c. Right-click the new entry, and then select Unlock Rows.
d. Click Apply and close the active window.
10. If a Proxy Server is not implemented, map outgoing telephone calls to IP addresses.
11. On the Tab bar, click on SIP and then Routing.
The system displays SIP Routing frame.
12. Select the Tel to IP tab.
a. Click the button to add a new entry, and then click Yes to confirm.
The system displays Routing table.
b. Double-click each field to enter values.
c. Right-click the new entry and select Unlock Rows.
d. Click Apply and close the active window.
The routing list also has the property that if a call cannot be successfully routed to the
destination specified by a matched rule, then the next rule that matches is used to attempt to
route the call. This can be used to implement a failover strategy if one of the TN799DP C-LAN
circuit packs is not responding or not available. For this feature to work most effectively, during
maintenance operations, the Ethernet interface should be disabled on a TN799DP C-LAN
circuit pack as well as busied it out.
For more information about determining the best routing and alternate routing configurations,
contact an Avaya software specialist.
Note:
Note: The two rules set up in these steps implement a form of load sharing that is based
on the calling numbers of the inbound traffic.
4. Select the IP to Tel tab to display the SIP IP to Tel Routing List.
Board slots are numbered from 1 to 10 from bottom to top, as displayed on the left side of
the front panel.The TP6310 boards are in slots 9 and 10.
4. Right-click on the active TP6310 card shown in black and select Maintenance > Unlock.
5. Click Yes in the confirmation window.
The TP6310 board resets and returns to service after several minutes. The Alarm Browser pane
at the bottom of the window indicates the status of the board.
Note:
Note: Verify that all the board elements (DS3, DS1, routing rules, trunk groups, etc.) are
unlocked.
● How inbound calls from Mediant 3000 Media Gateway are directly mapped to 5-digit
extensions in Avaya Communication Manager
● How Automatic Route Selection (ARS) is configured for outbound calls through Mediant
3000 Media Gateway
Note:
Note: The following procedure is for Communication Manager that is using C-LAN
configuration.
Note:
Note: Ensure that the packetization time matches on the Communication Manager and
Mediant 3000 server.
! Important:
Important: The number of trunk members must be distributed across all the trunk groups
used for Mediant 3000 Media Gateway.
10. In the Trunk Features section, set the Numbering Format field to public.
11. Accept the default values on the remaining fields.
12. Press Enter to save your changes.
If you have more than one signaling group, repeat these steps for each trunk group.
! Important:
Important: Ensure that all trunk groups are added into any affected routes.
1. Type change dialplan analysis and press Enter to add the feature access code for
outside dialing.
2. Fill in all the fields. For example, type 9 if users must dial 9 to get an outside line, length 1,
and fac for Feature Access Code.
3. Type change feature-access-codes and press Enter.
4. In the Auto Route Selection (ARS) - Access Code 1 field specify the correct access
code for outside dialing. For example, type 9 if users must dial 9 to get an outside line.
5. Type change ars analysis n and press Enter to configure the route selection based
on the number dialed following the access code.
For example, type 720 if users must dial the area code 720 after dialing 9 for outbound
local calls.
6. Type change route-pattern n and press Enter to define the SIP trunk groups to be
selected for the corresponding route pattern.
7. Set Secure SIP field to N.
8. Type change locations and press Enter.
9. In the Proxy Sel. Rte. Pat. field, designate the SIP trunk routing patterns from step 6. This
provides support for features such as call transfer.
Refer to the Call Routing Guide available on the Avaya Support Web site (http://
support.avaya.com) for more information.
Note:
Note: These instructions are written assuming the Avaya Aura® Session Manager and
Avaya Aura® System Manager are installed on your machine and are fully
functional.
The following sections describes how to configure call routing in Session Manager for Mediant
3000 Media Gateway.
Adding locations
For bandwidth management, the Locations tab is used to specify the logical and physical
locations of SIP entities.
1. Make calls from a telephone on the PSTN to a SIP telephone. If inbound calls fail, verify
that the proper Tel to IP routing rules have been defined to support routing SIP calls to the
appropriate C-LANS.
2. Make calls between SIP telephones registered within each proxy to verify successful call
completion.
If outbound calls fail, verify that the proper IP to Tel routing rules have been defined to
support routing SIP calls to the appropriate DS3/DS1.
3. Log in to a SAT session on the media server.
4. Type list trace station to verify that the call is being properly routed through the SIP trunk
to Mediant 3000 device.
5. Change the transport protocol to TCP on the SIP trunk and use a SIP-capable protocol
analyzer to monitor the signaling messages.
6. Type change signaling group and press Enter.
7. Change the following fields:
● Transport Method: tcp
● Far-end Listen Port: 5060
Introduction
If proper and complete administration is not performed, the following issues can potentially
sabotage a Mediant 3000/CM integration:
● When routing inbound calls to CM (from the PSTN) is based on the ANI of the calling party
and NO ANI is received from the Service Provider. In this case, if Mediant 3000 is not
setup correctly, a one-way talk path may result.
● Failure to administer Mediant 3000 and CM so there are enough trunk members in CM
trunk groups as well as the ability to account for 'alternate routes' to CM in the event of
network issues or resource limitations.
This section takes you step-by-step through the administration process to avoid potential pitfalls
in these situations as well discuss the reasons for each step and the concepts involved.
Note:
Note: In CM, it's possible to create two (or more) SIP signaling groups where the
'near-end' and 'far-end' nodes have the same exact values. Doing so does NOT
add to the count of utilized TLS links. Two identical SIP signaling groups will still
only count as one TLS link. This sort of programming can also be used to get
around the SIP trunk group limit of 255 members.
2. SIP trunks: For each set of C-LANs supporting a TP6310-DS3 connection, assign at least
as many CM SIP trunk members as DS0 channels they are supporting. Note that there is a
maximum of 255 members per SIP trunk group. Therefore the C-LANs supporting the max
~640 channels on a DS3 (total number of DS-0s vary by DS-1 signaling type) should have
combined at least that many CM SIP trunk members. This can be achieved by creating
three SIP signaling groups, each with 255 trunk members, using three C-LANs, etc.
3. C-LANs: There is no real upper limit to the number of signaling groups or trunk members
per C-LAN. The number of C-LANs to allocate for each DS3 board should be determined
by the call traffic volume (BHCC) and the desired redundancy.
4. Decide, or estimate, how many SIP calls per hour a C-LAN can handle. This number is not
fixed, but depends on the complexity of the call. A C-LAN may be able to handle up to 9k
calls/hr for a standard, no-frills, inbound-to-CM-endpoint call, but only 50% or less of the
more complex call center calls with multiple connections to announcements, IVR,
transfers, each of which is a separate SIP re-INVITE transaction.
- For a call center, consider starting with no more than 5k calls/hr for each C-LAN.
5. Dividing this calls/hr load for a C-LAN into the total call load for a DS3 connection gives us
the number of C-LANs required to support that connection.
- Mediant 3000 can route calls based on dialed digits, which means separate C-LANs
supporting the same DS3 board may see different (uneven) traffic levels based on the
application or service provided on the extensions handled. In this case, C-LAN
allocation may be done according to each type of application traffic on each board.
6. Combining the previous two rules gives the number of CM SIP trunk members on each
C-LAN.
7. If the total SIP trunk use is not close to the CM limit of 12,000 total SIP trunks, and there's
more than one C-LAN supporting a DS3 board, then allocate a number of trunks about
10% greater than the actual number of DS0s. The reason for this is that when the
channels on the DS3 are almost all occupied during very busy periods of time, there is a
greater chance that one or more of the C-LANs supporting that connection will run out of
SIP trunk members.
8. Using the previous example, if you are using three C-LANs to support the full ~640 DS-0
channels on a DS3, you can allocate 255 SIP trunk members on each C-LAN, which is the
maximum amount for one signaling group. Doing this across three trunk-groups gets
765 members (255+255+255=765).
9. Lastly, you want to be sure you have properly set up Alternate Routing in Mediant 3000
(which is discussed below) for the case where you do use up all trunk members in a CM
trunk group. While Mediant 3000 does not have the ability to know how many trunk
members are available in a CM trunk group, it can re-route a call to another trunk group
when it receives any one of four (programmable) SIP reason codes. In this case, CM
returns a 503 reason code to Mediant 3000 when all the trunk members are active or
otherwise unavailable.
Note:
Note: You can add any other SIP reason codes that are applicable to your environment.
None of the scenarios are enabled by default, though, for a thorough Mediant 3000
implementation, all three should be enabled.
Example scenario
Below is the theoretical configuration that will be used in the example:
1 1 10.10.10.1
2 2 10.10.10.2
3 3 10.10.10.3
2. Based on the information in the above figure, add the Tel to IP entries as follows:
Note:
Note: Double-clicking a row enables you to edit an existing entry.
Selecting + from the upper right corner of the EMS client enables you
to add new entries.
Note:
Note: For each primary rule, there is a nearly identical rule, the difference being the
destination IP address. Since each IP address represents a different CM
Signaling Group, ultimately the call is being sent to a different CM trunk group.
When Mediant 3000 sees two identical rules it knows the first one in the list is the
primary route and assigns the second rule to be the alternate route, provided that
alternate routing is enabled.
Note:
Note: The last rule, where Dst Phone Prefix and Src Phone Prefix are set to *, accounts
for the case where a 10-digit Src Phone Prefix (ANI) is not received.
! Important:
Important: Verify that each of these rules is UNLOCKED.
3. Select the Routing Setting tab and set the Enable Alternate Routing field to yes.
4. Set the Alternative Routing Mode to Both or Conn.
5. Set appropriate values for the QoS fields (Max Allowed Delay and Packet Lost) if you set
the Alternative Routing Mode to Both in step 4.
The figure below contains values for the settings applied in the above steps.
6. Select the Alternate Tel to IP tab and add up to four table entries with appropriate SIP
Reason codes (add '503' as one of the reason codes).
Currently, in a single CM SIP trunk group there is a limit of 255 trunk members. Notice that
from step 1 an attempt is made to load balance calls across three SIP signaling groups.
However, there is nothing to prevent Mediant 3000 from receiving more than 255
simultaneous calls where the last digit in the callers ANI ends with 0-3. In this case, all the
trunk group members will be exhausted. So, for the 256th call, CM returns a SIP 503
reason code to Mediant 3000, which indicates that no additional trunk members are
available. Rules in the Alternate Tel to IP tab enable you to instruct Mediant 3000 what to
do when receiving specific reason codes (up to four entries). Since you know that CM is
going to send a SIP 503 reason code, you have instructed Mediant 3000 to invoke the
alternate route before giving up on the call.
! Important:
Important: Verify that each of these rules is UNLOCKED.
7. Create at least one Mediant 3000 trunk group as follows:
a. In the EMS, double-click the TP6310.
b. Select the SIP tab.
c. Select the Trunk Groups tab.
Note:
Note: The value in the Starting Phone Number field is $$.
5. Change the value in the Starting Phone Number field to a numeric value. This has to
be done in case any or all incoming calls from the PSTN to Mediant 3000 do not
contain ANI. Align the value with the routing rules setup in the Tel to IP routing tab.
Note:
Note: In the U.S., this will generally be a 10-digit number.
In this example, you have one DS-3. You have created Trunk Group 1 on Mediant
3000, which consists of all 28 DS-1s in the DS-3. For each DS-1 you have also
indicated that all 24 channels are available (you can include a D-channel used for
signaling in the range as Mediant 3000 will not try to utilize it for voice).
When ANI is blocked or otherwise not present, Mediant 3000 will use the number
100000001 as the starting point for routing calls to CM. As calls come across
B-channels, Mediant 3000 will increment this starting number by 1 so that the call can
be properly load balanced across C-LANs using the rules defined in the Tel to IP tab.
This will also enable Mediant 3000 to construct SIP messages where FROM message
containing only a numeric value.
For example: From: <sip:[email protected]:5061>;tag=1c26112113
! Important:
Important: If you have more than one trunk group for a DS-3, set the Starting Phone Number
in each Trunk Group to values that will not overlap. For example, for Trunk Group
1 you could set the Starting Phone Number to 1000000001 and for Trunk Group 2
you would then set the Starting Phone Number to 2000000001.
! Important:
Important: Verify that each of these rules is UNLOCKED.
Factory production M3K models will be loaded with the following configuration file. This pre-
loaded configuration is helpful to pre-populate administrative settings for the following:
● Voice codec will be set to Mu-law operation
● Ds1 trunks
● DS3 interfaces
● ISDN NFAS groups
● T.38 fax settings
The following textual description describes the exact configuration setting the various features.
Note:
Note: The pre-administrative settings save up to 20 minutes in setting up a new M3K
gateway.
--------------------------------------------------------------------------------------
;**************
;** Ini File **
;**************
M3KGlobalIpAddr = 0.0.0.0
CallProgressTonesFilename = 'M2K_usa_tones.dat'
V22ModemTransportType = 0
V23ModemTransportType = 0
V32ModemTransportType = 0
V34ModemTransportType = 0
FaxTransportMode = 1
isFaxUsed = 1
RFC2833PayloadType = 96
ENABLEEARLYMEDIA = 1
[ CodersGroup0 ]
FORMAT CodersGroup0_Index = CodersGroup0_Name, CodersGroup0_pTime, CodersGroup0_rate,
CodersGroup0_PayloadType, CodersGroup0_Sce;
CodersGroup0 0 = g711Ulaw64k, 20, 64, 0, 0;
[ \CodersGroup0 ]
PCMLawSelect = 3
;~ PCMLawSelect: Selects the type of PCM companding law in input/output TDM bus (TDM bus
is defined using the TDMBusType parameter). 1 = A-law, 3 = Mu-Law.
TDMBusClockSource = 4
;~ TDMBusClockSource: Selects the clock source on which the board synchronizes. 1 =
Local oscillator, 3 = MVIP, 4 = PSTN Network, 8 = H.110A, 9 = H.110B, 10 = NetRef1, 11
= NetRef2, 12 = SC2M, 13 = SC4M, 14 = SC8M, 15 = BITS, 16 = Network-B, 17 = ATM-OC3, 18
= ATM-OC3-B, 19 = ATM-OC12, 20 = Network-DS3-1, 21 = Network-DS3-2, 22 = Network-DS3-3,
Default = 1, TP-1610 = 3.
[PSTN Params]
ProtocolType = 11
;~ ProtocolType: Used to set the PSTN protocol to be used for this trunk. Relevant only
when TDMBusType=acFRAMERS (2). Either: NONE = 0, E1_EURO_ISDN = 1, T1_CAS = 2,
T1_RAW_CAS = 3, T1_TRANSPARENT = 4, E1_TRANSPARENT_31 = 5, E1_TRANSPARENT_30 = 6,
E1_MFCR2 = 7, E1_CAS = 8, E1_RAW_CAS = 9, T1_NI2_ISDN = 10, T1_4ESS_ISDN = 11,
T1_5ESS_9_ISDN = 12, T1_5ESS_10_ISDN = 13, T1_DMS100_ISDN = 14, J1_TRANSPARENT = 15,
T1_NTT_ISDN = 16, E1_AUSTEL_ISDN = 17, E1_HKT_ISDN = 18, E1_KOR_ISDN = 19, T1_HKT_ISDN
= 20, E1_QSIG = 21, E1_TNZ_ISDN = 22, T1_QSIG = 23, V5_2_AN = 26, T1_IUA = 28, E1_IUA =
29, E1_FRENCH_VN6_ISDN = 30, E1_FRENCH_VN3_ISDN = 31, T1_EURO_ISDN = 34,
T1_DMS100_MERIDIAN_ISDN = 35, T1_NI1_ISDN = 36, E1_DUA = 37, E1_Q931_PACKETS = 38,
T1_Q931_PACKETS = 39, E1_NI2_ISDN = 40.
FramingMethod = D
;~ FramingMethod: Selects the physical framing method used for this trunk. 0 = default
according to protocol type E1 or T1, [E1 default = E1 CRC4 MultiFrame Format extended
G.706B (as c)], [T1 default = T1 Extended SuperFrame with CRC6 (as D)], 1 = T1
SuperFrame Format (as B). a = E1 DoubleFrame Format, b = E1 CRC4 MultiFrame Format, c =
E1 CRC4 MultiFrame Format extended G.706B, A = T1 4-Frame multiframe. B = T1 12-Frame
multiframe (D4). C = T1 Extended SuperFrame without CRC6, D = T1 Extended SuperFrame
with CRC6, E = T1 72-Frame multiframe (SLC96), F = J1 Extended SuperFrame with CRC6
(Japan).
DCHConfig_0 = 0
;~ DCHConfig: Defines D-channel configuration. This setting is only applicable to ISDN
PRI protocols that support NFAS and/or D-channel backup procedures. 0 = D-channel is
Primary, 1 = Backup, 2 = NFAS.
DCHConfig_1 = 1
DCHConfig_2 = 2
DCHConfig_3 = 2
DCHConfig_4 = 2
DCHConfig_5 = 2
DCHConfig_6 = 2
DCHConfig_7 = 2
DCHConfig_8 = 2
DCHConfig_9 = 2
DCHConfig_11 = 1
DCHConfig_12 = 2
DCHConfig_13 = 2
DCHConfig_14 = 2
DCHConfig_15 = 2
DCHConfig_16 = 2
DCHConfig_17 = 2
DCHConfig_18 = 2
DCHConfig_19 = 2
DCHConfig_20 = 0
;~ DCHConfig: Defines D-channel configuration. This setting is only applicable to ISDN
PRI protocols that support NFAS and/or D-channel backup procedures. 0 = D-channel is
Primary, 1 = Backup, 2 = NFAS.
DCHConfig_21 = 1
DCHConfig_22 = 2
DCHConfig_23 = 2
DCHConfig_24 = 2
DCHConfig_25 = 2
DCHConfig_26 = 2
DCHConfig_27 = 2
ISDNIBehavior = 2560
;~ ISDNIBehavior: Bit-field used to determine several behavior options, which influence
how the Q.931 protocol behaves. Refer to the VoPLib documentation (ISDN Flexible
Behavior).
ISDNNFASInterfaceID_0 = 0
;~ ISDNNFASInterfaceID: Defines the Interface ID. Works with NS_EXPLICIT_INTERFACE_ID.
Refer to the VoPLib documentation(ISDN Flexible Behavior). Default = (unsigned char)-1.
Range = 0 to 255.
ISDNNFASInterfaceID_1 = 1
ISDNNFASInterfaceID_2 = 2
ISDNNFASInterfaceID_3 = 3
ISDNNFASInterfaceID_4 = 4
ISDNNFASInterfaceID_5 = 5
ISDNNFASInterfaceID_6 = 6
ISDNNFASInterfaceID_7 = 7
ISDNNFASInterfaceID_8 = 8
ISDNNFASInterfaceID_9 = 9
ISDNNFASInterfaceID_10 = 0
;~ ISDNNFASInterfaceID: Defines the Interface ID. Works with NS_EXPLICIT_INTERFACE_ID.
Refer to the VoPLib documentation(ISDN Flexible Behavior). Default = (unsigned char)-1.
Range = 0 to 255.
ISDNNFASInterfaceID_11 = 1
ISDNNFASInterfaceID_12 = 2
ISDNNFASInterfaceID_13 = 3
ISDNNFASInterfaceID_14 = 4
ISDNNFASInterfaceID_15 = 5
ISDNNFASInterfaceID_16 = 6
ISDNNFASInterfaceID_17 = 7
ISDNNFASInterfaceID_18 = 8
ISDNNFASInterfaceID_19 = 9
ISDNNFASInterfaceID_20 = 0
;~ ISDNNFASInterfaceID: Defines the Interface ID. Works with NS_EXPLICIT_INTERFACE_ID.
Refer to the VoPLib documentation(ISDN Flexible Behavior). Default = (unsigned char)-1.
Range = 0 to 255.
ISDNNFASInterfaceID_21 = 1
ISDNNFASInterfaceID_22 = 2
ISDNNFASInterfaceID_23 = 3
ISDNNFASInterfaceID_24 = 4
ISDNNFASInterfaceID_25 = 5
ISDNNFASInterfaceID_26 = 6
ISDNNFASInterfaceID_27 = 7
NFASGroupNumber_0 = 1
;~ NFASGroupNumber: Relevant only to T1 ISDN NFAS trunks, this parameter indicates the
group number of the NFAS group. Valid NFAS group numbers are only 1 to 4. 0 indicates
that this trunk is not NFAS (in this case the parameters ISDN NFAS Interface ID and Dch
Config are ignored). Range = 0 to 9.
NFASGroupNumber_1 = 1
NFASGroupNumber_2 = 1
NFASGroupNumber_3 = 1
NFASGroupNumber_4 = 1
NFASGroupNumber_5 = 1
NFASGroupNumber_6 = 1
NFASGroupNumber_7 = 1
NFASGroupNumber_8 = 1
NFASGroupNumber_9 = 1
NFASGroupNumber_10 = 2
;~ NFASGroupNumber: Relevant only to T1 ISDN NFAS trunks, this parameter indicates the
group number of the NFAS group. Valid NFAS group numbers are only 1 to 4. 0 indicates
that this trunk is not NFAS (in this case the parameters ISDN NFAS Interface ID and Dch
Config are ignored). Range = 0 to 9.
NFASGroupNumber_11 = 2
NFASGroupNumber_12 = 2
NFASGroupNumber_13 = 2
NFASGroupNumber_14 = 2
NFASGroupNumber_15 = 2
NFASGroupNumber_16 = 2
NFASGroupNumber_17 = 2
NFASGroupNumber_18 = 2
NFASGroupNumber_19 = 2
NFASGroupNumber_20 = 3
NFASGroupNumber_21 = 3
NFASGroupNumber_22 = 3
NFASGroupNumber_23 = 3
NFASGroupNumber_24 = 3
NFASGroupNumber_25 = 3
NFASGroupNumber_26 = 3
NFASGroupNumber_27 = 3
CASProtocolEnable = 1
;~ CASProtocolEnable: This parameter enables or disables the possibility CAS protocol
configuration. At TP6310 When this parameter is enable the conference will disabled in
case it's relevant. 0 = Disable, 1 = Enable.
PSTNTransmissionType = 2
;~ PSTNTransmissionType: Sets the PSTN Transmission type for the board. Relevant only
when TDMBusType=acFRAMERS (2). Values are: 0 = NONE, PSTN Transmission type is not
defined, 1 = Optical SONET or SDH Transmission type, 2 = Copper DS3 (T3) Transmission
type, 3 = Copper E1 or DS1(T1) Transmission type, Default is NONE for TP6310 and Copper
E1 or DS1 for rest of boards.
DTMFTransportType = 2
;~ DTMFTransportType: Defines the type of DTMF transport. 0 = Erase DTMFs from voice
transport not relayed to remote, 2 = DTMFs not erased are not relayed to remote, 3 =
DTMFs are muted from the voice stream and relayed according to RFC 2833, 7 = DTMFs are
sent according to RFC 2833 and muted when received.
[SIP Params]
ADDIEINSETUP = 200200e3
;~ ADDIEINSETUP: Additional information element to send in ISDN Setup message.
SENDIEONTG = '1'
;~ SENDIEONTG: Configure trunk groups on which to send additional IE.
USERTOUSERHEADERFORMAT = 2
;
; *** TABLE DS3CONFIG ***
;
;
[ DS3CONFIG ]
FORMAT DS3CONFIG_Index = DS3CONFIG_FramingMethod, DS3CONFIG_ClockSource,
DS3CONFIG_LineBuildOut, DS3CONFIG_CircuitIdentifier, DS3CONFIG_TrapEnable,
DS3CONFIG_PmOnOff, DS3CONFIG_TappingEnable, DS3CONFIG_AdminState;
DS3CONFIG 0 = 0, 0, 4, , 1, 1, 0, 1;
DS3CONFIG 1 = 0, 0, 4, , 1, 1, 0, 1;
DS3CONFIG 2 = 0, 0, 4, , 1, 1, 0, 1;
[ \DS3CONFIG ]
------------------------------------------------------------------------------------