This document describes the AudioCodes MediaPack series Voice over IP (VoIP) gateways.
Information contained in this document is believed to be accurate and reliable at the time of printing.
However, due to ongoing product improvements and revisions, AudioCodes cannot guarantee
accuracy of printed material after the Date Published nor can it accept responsibility for errors or
omissions. Updates to this document and other documents can be viewed by registered Technical
Support customers at www.audiocodes.com under Support / Product Documentation.
Figure 1-1: Typical MediaPack VoIP Application .............................................................................................18
Figure 2-1: MP-108 Front Panel.......................................................................................................................23
Figure 2-2: MP-124 Front Panel.......................................................................................................................23
Figure 6-1: ini File Structure ...........................................................................................................................164
Figure 6-2: SIP ini File Example .....................................................................................................................164
Figure 12-3: Example of a Base64-Encoded X.509 Certificate......................................................................215
Figure 12-4: Example of the File clients.conf (FreeRADIUS Client Configuration)........................................217
Figure 12-5: Example of a User Configuration File for FreeRADIUS Using a Plain-Text Password .............217
Figure 14-1: Embedded Web Server CLI Screen...........................................................................................223
Figure 15-1: Example of Entries in a Device ini file Regarding SNMP...........................................................237
Figure 16-1: Call Progress Tone Types..........................................................................................................242
Figure 16-2: Defining a Dial Tone Example....................................................................................................243
Figure 16-3: Examples of Various Ringing Signals........................................................................................245
Figure B-1: Main Screen.................................................................................................................................259
Table 5-1: Protocol Definition, General Parameters (continues on pages 52 to 55)........................................52
Table 5-2: Proxy & Registration Parameters (continues on pages 57 to 60) ...................................................57
Table 5-3: ini File Coder Parameter .................................................................................................................62
Table 5-4: DTMF & Dialing Parameters (continues on pages 63 to 65) ..........................................................63
Table 5-5: Advanced Parameters, General Parameters (continues on pages 67 to 70) .................................67
Table 5-6: Supplementary Services Parameters (continues on pages 72 to 74).............................................72
Table 5-7: Keypad Features Parameters .........................................................................................................75
Table 5-8: Number Manipulation Parameters ..................................................................................................77
Table 5-9: Number Manipulation ini File Parameters (continues on pages 78 to 79)......................................78
Table 5-10: Routing Tables, General Parameters (continues on pages 81 to 82)...........................................81
Table 5-11: Tel to IP Routing Table..................................................................................................................84
Table 5-12: IP to Hunt Group Routing Table....................................................................................................87
Table 5-13: Internal DNS ini File Parameter ....................................................................................................88
Table 5-14: Reasons for Alternative Routing ini File Parameter ......................................................................90
Table 5-15: ini File Coder Group Parameters ..................................................................................................92
Table 5-16: ini File Tel Profile Settings.............................................................................................................94
Table 5-17: ini File IP Profile Settings ..............................................................................................................96
Tip: When viewing this manual on CD, Web site or on any other electronic copy,
all cross-references are hyperlinked. Click on the page or section numbers
(shown in blue) to reach the individual cross-referenced item directly. To
return back to the point from where you accessed the cross-reference, press
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Trademarks
AC logo, Ardito, AudioCoded, AudioCodes, AudioCodes logo, IPmedia, Mediant, MediaPack, MPMLQ, NetCoder, Stretto, TrunkPack, VoicePacketizer and VoIPerfect, are trademarks or
registered trademarks of AudioCodes Limited. All other products or trademarks are property of
their respective owners.
Customer Support
Customer technical support and service are provided by AudioCodes’ Distributors, Partners, and
Resellers from whom the product was purchased. For Customer support for products purchased
directly from AudioCodes, contact support@audiocodes.com
.
Abbreviations and Terminology
Each abbreviation, unless widely used, is spelled out in full when first used. Only industrystandard terms are used throughout this manual. Hexadecimal notation is indicated by 0x
preceding the number.
Related Documentation
Document # Manual Name
LTRT-656xx (e.g., LTRT-65601) MediaPack & Mediant 1000 SIP Analog Gateways Release Notes
LTRT-614xx MP-1xx Fast Track Installation Guide
LTRT-615xx MP-11x Fast Track Installation Guide
LTRT-665xx CPE Configuration Guide for Voice Mail
Note 1:MP-1xx refers to the MP-124 24-port, MP-108 8-port, MP-104 4-port and
MP-102 2-port VoIP gateways having similar functionality except for the
number of channels (the MP-124 and MP-102 support only FXS).
Note 2:MP-11x refers to the MP-118 8-port, MP-114 4-port and MP-112 2-port VoIP
gateways having similar functionality except for the number of channels.
Note 3: MP-10x refers to MP-108 8-port, MP-104 4-port and MP-102 2-port
Note 4: MP-1xx/FXS refers only to the MP-124/FXS, MP-108/FXS, MP-104/FXS and
Note 5: MP-10x/FXO refers only to MP-108/FXO and MP-104/FXO gateways.
Note: In the current version, MP-11x devices only support FXS. References to
gateways.
MP-102/FXS gateways.
FXO only apply to MP-1xx devices.
Note:The MP-112 differs from the MP-114 and MP-118. Its configuration excludes
the RS-232 connector, the Lifeline option and outdoor protection.
Version 4.6 15 June 2005
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MediaPack SIP
Note:Where ‘network’ appears in this manual, it means Local Area Network (LAN),
Wide Area Network (WAN), etc. accessed via the gateway’s Ethernet
Note:FXO (Foreign Exchange Office) is the interface replacing the analog
FXS (Foreign Exchange Station) is the interface replacing the Exchange
Warning: Ensure that you connect FXS ports to analog telephone or to PBX-trunk
interface.
telephone and connects to a Public Switched Telephone Network (PSTN)
line from the Central Office (CO) or to a Private Branch Exchange (PBX).
The FXO is designed to receive line voltage and ringing current, supplied
from the CO or the PBX (just like an analog telephone). An FXO VoIP
gateway interfaces between the CO/PBX line and the Internet.
(i.e., the CO or the PBX) and connects to analog telephones, dial-up
modems, and fax machines. The FXS is designed to supply line voltage and
ringing current to these telephone devices. An FXS VoIP gateway interfaces
between the analog telephone devices and the Internet.
lines only and FXO ports to CO/PBX lines only.
Warning: The MediaPack is supplied as a sealed unit and must only be serviced by
qualified service personnel.
Warning: Disconnect the MediaPack from the mains and from the Telephone Network
This document provides you with the information on installation, configuration and operation of
the MP-124 24-port, MP-108 8-port, MP-104 4-port, MP-102 2-port, MP-118 8-port, MP-114 4port and MP-112 2-port VoIP media gateways. As these units have similar functionality (with the
exception of their number of channels and some minor features), they are collectively referred to
in the manual as the MediaPack.
1.2 Gateway Description
The MediaPack series analog VoIP gateways are cost-effective, cutting edge technology
products. These stand-alone analog VoIP gateways provide superior voice technology for
connecting legacy telephones, fax machines and PBX systems with IP-based telephony
networks, as well as for integration with new IP-based PBX architecture. These products are
designed and tested to be fully interopeable with leading softswitches and SIP servers.
The MediaPack gateways incorporate up to 24 analog ports for connection, either directly to an
enterprise PBX (FXO), to phones, or to fax (FXS), supporting up to 24 simultaneous VoIP calls.
Additionally, the MediaPack units are equipped with a 10/100 Base-TX Ethernet port for
connection to the network.
The MediaPack gateways are best suited for small to medium size enterprises, branch offices or
for residential media gateway solutions.
The MediaPack gateways enable users to make free local or international telephone / fax calls
between the distributed company offices, using their existing telephones / fax. These calls are
routed over the existing network ensuring that voice traffic uses minimum bandwidth.
The MediaPack gateways are very compact devices that can be installed as a desk-top unit, on
the wall or in a 19-inch rack.
The MediaPack gateways support SIP (Session Initiation Protocol) protocol, enabling the
deployment of ‘voice over IP’ solutions in environments where each enterprise or residential
location is provided with a simple media gateway.
This provides the enterprise with a telephone connection (e.g., RJ-11), and the capability to
transmit the voice and telephony signals over a packet network.
Version 4.6 17 June 2005
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MediaPack SIP
The layout diagram (Figure 1-1), illustrates a typical MediaPack VoIP application.
Figure
1-1: Typical MediaPack VoIP Application
1.3 SIP Overview
SIP (Session Initialization Protocol) is an application-layer control (signaling) protocol used on the
MediaPack for creating, modifying, and terminating sessions with one or more participants. These
sessions can include Internet telephone calls, media announcements and conferences.
SIP invitations are used to create sessions and carry session descriptions that enable participants
to agree on a set of compatible media types. SIP uses elements called Proxy servers to help
route requests to the user's current location, authenticate and authorize users for services,
implement provider call-routing policies and provide features to users.
SIP also provides a registration function that enables users to upload their current locations for
use by Proxy servers. SIP, on the MediaPack, complies with the IETF (Internet Engineering Task
Force) RFC 3261 (refer to http://www.ietf.org
• Comprehensive support for supplementary services.
• Web Management for easy configuration and installation.
• EMS for comprehensive management operations (FCAPS).
• Simple Network Management Protocol (SNMP) and Syslog support.
• SMDI support for Voice Mail applications.
• Multiplexes RTP streams from several users together to reduce bandwidth overhead.
• T.38 fax fallback to PCM (or NSE).
• Can be integrated into a Multiple IPs and a VLAN-aware environment.
• Capable of automatically updating its firmware version and configuration.
• Secured Web access (HTTPS) and Telnet access using SSL / TLS.
1.4.2 MP-1xx Hardware Features
•MP-124 19-inch, 1 U rugged enclosure provides up to 24 analog FXS ports, using a single
50 pin Telco connector.
•MP-10x compact, rugged enclosure only one-half of a 19-inch rack unit, 1 U high (1.75" or
44.5 mm).
•Lifeline - provides a wired phone connection to PSTN line when there is no power, or the
network fails (applies to MP-10x FXS gateways).
•LEDs on the front and rear panels that provide information on the operating status of the
media gateway and the network interface.
•Restart button on the Front panel that restarts the MP-1xx gateway, and is also used to
restore the MP-1xx parameters to their factory default values.
1.4.3 MP-11x Hardware Features
• MP-11x compact, rugged enclosure only one-half of a 19-inch rack unit, 1 U high.
• Lifeline - provides a wired phone connection to PSTN line when there is no power, or the
network fails.
•LEDs on the front panel that provide information on the operating status of the media
gateway and the network interface.
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MediaPack SIP
•Restart button on the back panel that restarts the MP-11x gateway, and is also used to
restore the MP-11x parameters to their factory default values.
1.4.4 SIP Features
The MediaPack SIP gateway complies with the IETF RFC 3261 standard.
• Reliable User Datagram Protocol (UDP) transport, with retransmissions.
• Transmission Control Protocol (TCP) Transport layer.
• SIPS using TLS (MP-11x only).
• T.38 real time fax (using SIP).
Note: If the remote side includes the fax maximum rate parameter in the Session Description
Protocol (SDP) body of the INVITE message, the gateway returns the same rate in the
response SDP.
• Works with Proxy or without Proxy, using an internal routing table.
• Fallback to internal routing table if Proxy is not responding.
• Supports up to four Proxy servers. If the primary Proxy fails, the MediaPack automatically
switches to a redundant Proxy.
•Supports domain name resolving using DNS SRV records for Proxy, Registrar and domain
names that appear in the Contact and Record-Route headers.
•Proxy and Registrar Authentication (handling 401 and 407 responses) using Basic or Digest
methods.
• Single gateway Registration or multiple Registration of all gateway endpoints.
• Configuration of authentication username and password per each gateway endpoint, or
¾ INFO method <draft-choudhuri-sip-info-digit-00.txt>.
¾ INFO method, compatible with Cisco gateways.
¾ NOTIFY method <draft-mahy-sipping-signaled-digits-01.txt>.
•SIP URL: sip:”phone number”@IP address (such as 122@10.1.2.4, where “122” is the
phone number of the source or destination phone number) or sip:”phone_number”@”domain
name”, such as 122@myproxy.com. Note that the SIP URI host name can be configured
differently per called number.
• Can negotiate coder from a list of given coders.
•Implementation of Message Waiting Indication (MWI) IETF <draft-ietf-sipping-mwi-04.txt>,
including SUBSCRIBE (to the MWI server). The MediaPack/FXS gateways can accept an
MWI NOTIFY message that indicates waiting messages or indicates that the MWI is cleared.
For more updated information on the gateway’s supported features, refer to the latest MediaPack
SIP Release Notes.
This section provides detailed information on the hardware, the location and functionality of the
LEDs, buttons and connectors on the front and rear panels of the MP-1xx (refer to Section
below) and MP-11x (Section
2.2 on page 27) gateways.
2.1
For detailed information on installing the MediaPack, refer to Section
2.1 MP-1xx Physical Description
2.1.1 MP-1xx Front Panel
Figure 2-1 and Figure 2-2 illustrate the front layout of the MP-108 (almost identical on MP-104
and MP-102) and MP-124 respectively. Refer to Section
buttons; refer to Section
Reset Button
2.1.1.2 for functionality of the front panel LEDs.
Figure
2-1: MP-108 Front Panel
3 on page 29.
2.1.1.1 for meaning of the front panel
Reset Button
Figure
2-2: MP-124 Front Panel
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MediaPack SIP
2.1.1.1 MP-1xx Front Panel Buttons
Table 2-1 lists and describes the front panel buttons on the MP-1xx.
Table
2-1: Front Panel Buttons on the MP-1xx
Type Function Comment
Press the reset button with a paper clip or any other similar
pointed object, until the gateway is reset.
Refer to Section
10.1 on page 201.
Reset button
Reset the MP-1xx
Restore the MP-1xx parameters to
their factory default values
2.1.1.2 MP-1xx Front Panel LEDs
Table 2-2 lists and describes the front panel LEDs on the MP-1xx.
Note: MP-1xx (FXS/FXO) media gateways feature almost identical front panel
LEDs; they only differ in the number of channel LEDs that correspond to the
number of channels.
Table 2-2: Indicator LEDs on the MP-1xx Front Panel
Label Type Color State Function
Device Powered, self-test OK
Software Loading/Initialization
Malfunction
Valid 10/100 Base-TX Ethernet connection
Malfunction
Sending and receiving SIP messages
No traffic
Transmitting RTP (Real-Time Transport Protocol)
Packets
Receiving RTP Packets
No traffic
Offhook / Ringing for FXS Phone Port
FXO Line-Seize/Ringing State for Line Port
There’s an incoming call, before answering
Line Malfunction
Protective earthing screw (mandatory for all installations).
10/100 Base-TX Ethernet connection.
2, 4 or 8 FXS/FXO ports.
FXS / FXO label.
9 pin RS-232 status port (for Cable Wiring of the RS-232 refer to Figure
3-9 on page 35).
Table 2-4: Indicator LEDs on the MP-10x Rear Panel
Label Type Color State Meaning
4
5
6
ETH-1
Ethernet Status
Yellow ON
Red ON
Ethernet port receiving data
Collision
Note that the Ethernet LEDs are located within the RJ-45 socket.
Version 4.6 25 June 2005
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MediaPack SIP
2.1.2.2 MP-124 Rear Panel
Figure 2-4 illustrates the rear panel layout of the MP-124. For descriptions of the MP-124 rear
panel components, refer to Table
to Table
2-6.
2-5. For the functionality of the MP-124 rear panel LEDs, refer
MediaPack SIP User’s Manual 3. Installing the MediaPack
3 Installing the MediaPack
This section provides information on the installation procedure for the MP-1xx (refer to Section
3.1 below) and the MP-11x (refer to Section 3.2 on page 38). For information on how to start
using the gateway, refer to Section
The equipment must only be installed or serviced by qualified service personnel.
3.1 Installing the MP-1xx
¾ To install the MP-1xx, take these 4 steps:
1. Unpack the MP-1xx (refer to Section 3.1.1 below).
4 on page 43.
Caution Electrical Shock
2. Check the package contents (refer to Section
3. Mount the MP-1xx (refer to Section
4. Cable the MP-1xx (refer to Section
After connecting the MP-1xx to the power source, the Ready and LAN LEDs on the front panel
turn to green (after a self-testing period of about 1 minute). Any malfunction changes the Ready
LED to red.
When you have completed the above relevant sections you are then ready to start configuring the
gateway (Section
4 on page 43).
3.1.1 Unpacking
¾ To unpack the MP-1xx, take these 6 steps:
1. Open the carton and remove packing materials.
2. Remove the MP-1xx gateway from the carton.
3. Check that there is no equipment damage.
4. Check, retain and process any documents.
5. Notify AudioCodes or your local supplier of any damage or discrepancies.
6. Retain any diskettes or CDs.
3.1.1.1 below).
3.1.2 on page 30).
3.1.3 on page 33).
3.1.1.1 Package Contents
Ensure that in addition to the MP-1xx, thepackage contains:
• AC power cable for the AC power supply option.
• 3 brackets (2 short, 1 long) and bracket-to-device screws for 19-inch rack installation option
(MP-10x only).
•2 short equal-length brackets and bracket-to-device screws for MP-124 19-inch rack
installation.
• A CD with software and documentation may be included.
• The MP-1xx Fast Track Installation Guide.
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MediaPack SIP
3.1.2 Mounting the MP-1xx
The MP-1xx can be mounted on a desktop or on a wall (only MP-10x), or installed in a standard
19-inch rack. Refer to Section
3.1.2.1 Mounting the MP-1xx on a Desktop
No brackets are required. Simply place the MP-1xx on the desktop in the position you require.
3.1.3 on page 33 for cabling the MP-1xx.
Figure
3-1: Desktop or Shelf Mounting
Rack Mount Safety Instructions (UL)
When installing the chassis in a rack, be sure to implement the following Safety
instructions recommended by Underwriters Laboratories:
•Elevated Operating Ambient - If installed in a closed or multi-unit rack assembly,
the operating ambient temperature of the rack environment may be greater than
room ambient. Therefore, consideration should be given to installing the equipment
in an environment compatible with the maximum ambient temperature (Tma)
specified by the manufacturer.
•Reduced Air Flow - Installation of the equipment in a rack should be such that the
amount of air flow required for safe operation on the equipment is not compromised.
•Mechanical Loading - Mounting of the equipment in the rack should be such that a
hazardous condition is not achieved due to uneven mechanical loading.
•Circuit Overloading - Consideration should be given to the connection of the
equipment to the supply circuit and the effect that overloading of the circuits might
have on overcurrent protection and supply wiring. Appropriate consideration of
equipment nameplate ratings should be used when addressing this concern.
•Reliable Earthing - Reliable earthing of rack-mounted equipment should be
maintained. Particular attention should be given to supply connections other than
direct connections to the branch circuit (e.g., use of power strips.)
3.1.2.2 Installing the MP-10x in a 19-inch Rack
The MP-10x is installed into a standard 19-inch rack by the addition of two supplied brackets (1
short, 1 long). The MP-108 with brackets for rack installation is shown in Figure
3-2.
¾ To install the MP-10x in a 19-inch rack, take these 9 steps:
1. Remove the two screws on one side of the device nearest the front panel.
2. Insert the peg on the short bracket into the third air vent down on the column of air vents
nearest the front panel.
3. Swivel the bracket until the holes in the bracket line up with the two empty screw holes on
MediaPack SIP User’s Manual 3. Installing the MediaPack
4. Use the screws found in the devices’ package to attach the short bracket to the side of the
device.
5. Remove the two screws on the other side of the device nearest the front panel.
6. Position the long bracket so that the holes in the bracket line up with the two empty screw
holes on the device.
7. Use the screws found in the device’s package to attach the long bracket to the side
device.
8. Position the device in the rack and line up the bracket holes with the rack frame holes.
9. Use four standard rack screws to attach the device to the rack. These screws are not
provided with the device.
Figure
3-2: MP-108 with Brackets for Rack Installation
of the
3.1.2.3 Installing the MP-124 in a 19-inch Rack
The MP-124 is installed into a standard 19-inch rack by the addition of two short (equal-length)
supplied brackets. The MP-124 with brackets for rack installation is shown in Figure
¾ To install the MP-124 in a 19-inch rack, take these 7 steps:
1. Remove the two screws on one side of the device nearest the front panel.
2. Insert the peg on one of the brackets into the third air vent down on the column of air vents
nearest the front panel.
3. Swivel the bracket until the holes in the bracket line up with the two empty screw holes on
the device.
4. Use the screws found in the devices’ package to attach the bracket to the side of the device.
5. Repeat steps 1 to 4 to attach the second bracket to the other side of the device.
6. Position the device in the rack and line up the bracket holes with the rack frame holes.
7. Use four standard rack screws to attach the device to the rack. These screws are not
provided with the device.
3-3.
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MediaPack SIP
Figure 3-3: MP-124 with Brackets for Rack Installation
3.1.2.4 Mounting the MP-10x on a Wall
The MP-10x is mounted on a wall by the addition of two short (equal-length) supplied brackets.
The MP-102 with brackets for wall mount is shown in Figure
3-4.
¾ To mount the MP-10x on a wall, take these 7 steps:
1. Remove the screw on the side of the device that is nearest the bottom and the front panel.
2. Insert the peg on the bracket into the third air vent down on the column of air vents nearest
the front panel.
3. Swivel the bracket so that the side of the bracket is aligned with the base of the device and
the hole in the bracket line up with the empty screw hole.
4. Attach the bracket using one of the screws provided in the device package.
5. Repeat steps 1 to 4 to attach the second bracket to the other side of the device.
6. Position the device on the wall with the base of the device next to the wall.
7. Use four screws to attach the device to the wall. These screws are not provided with the
MediaPack SIP User’s Manual 3. Installing the MediaPack
3.1.3 Cabling the MP-1xx
Verify that you have the cables listed under column ‘Cable’ in Table 3-1 before beginning to cable
the MP-1xx according to the column ‘Cabling Procedure’. For detailed information on the MP-1xx
rear panel connectors, refer to Section
Table
3-1: Cables and Cabling Procedure
Cable Cabling Procedure
Connect the Ethernet connection on the MP-1xx directly to the network using a standard RJ-45
RJ-45 Ethernet
cable
Ethernet cable. For connector’s pinout refer to Figure 3-5 below.
Note that when assigning an IP address to the MP-1xx using HTTP (under step 1 in Section
4.2.1), you may be required to disconnect this cable and re-cable it differently.
Connect the RJ-11 connectors on the rear panel of the MP-10x/FXS to
fax machine, modem, or phones (refer to Figure
RJ-11 two-wire
telephone cords
Connect RJ-11 connectors on the MP-10x/FXO rear panel to telephone
exchange analog lines or PBX extensions (Figure
MP-124/FXS ports are usually distributed using an MDF Adaptor Block (special order option).
Refer to Figure
3-8 for details.
2.1.2 on page 25.
3-6).
3-6).
Ensure that FXS &
FXO are connected to
the correct devices,
otherwise damage can
occur.
Lifeline cable
50-pin Telco cable
(MP-124 devices
only).
An Octopus cable
is not included
with the MP-124
package.
RS-232 serial
cable
Protective
earthing strap
AC Power cable
For detailed information on setting up the Lifeline, refer to the procedure under Section 3.1.3.2 on
page 35.
Refer to the MP-124 Safety Notice below.
1. Wire the 50-pin Telco connectors according to the pinout in Figure 3-7 on page 34, and
Figure 3-8 on page 34.
2. Attach each pair of wires from a 25-pair Octopus cable to its corresponding socket on the
MDF Adaptor Block’s rear.
3. Connect the wire-pairs at the other end of the cable to a male 50-pin Telco connector.
4. Insert and fasten this connector to the female 50-pin Telco connector on the MP-124 rear
panel (labeled Analog Lines 1-24).
5. Connect the telephone lines from the Adaptor Block to a fax machine, modem, or telephones
by inserting each RJ-11 connector on the 2-wire line cords of the POTS phones into the RJ11 sockets on the front of an MDF Adaptor Block as shown in
For detailed information on connecting the MP-1xx RS-232 port to your PC, refer to Section
Figure 3-8 on page 34.
3.1.3.1 on page 35.
Connect an earthed strap to the chassis protective earthing screw and fasten it securely according
to the safety standards.
Connect the MP-1xx power socket to the mains.
MP-124 Safety Notice
To protect against electrical shock and fire, use a 26 AWG min wire to connect analog
FXS lines to the 50-pin Telco connector.
3.1.3.1 Connecting the MP-1xx RS-232 Port to Your PC
Using a standard RS-232 straight cable (not a cross-over cable) with DB-9 connectors, connect
the MP-1xx RS-232 port to either COM1 or COM2 RS-232 communication port on your PC. The
required connector pinout and gender are shown below in Figure
3-9.
For information on establishing a serial communications link with the MP-1xx, refer to Section
10.2 on page 201.
Figure
3-9: MP-1xx RS-232 Cable Wiring
2
3
5
DB-9 female for PC
DB-9 female for PCDB-9 male for MP-100
3.1.3.2 Cabling the Lifeline Phone
The Lifeline provides a wired analog POTS phone connection to any PSTN or PBX FXS port
when there is no power, or when the network connection fails. Users can therefore use the
Lifeline phone even when the MP-1xx is not powered on or not connected to the network. With
the MP-108/FXS and MP-104/FXS the Lifeline connection is provided on port #4 (refer to Figure
3-11). With the MP-102/FXS the Lifeline connection is provided on port #2.
RD
TD
GND
2
3
5
DB-9 male for MP-1xx
Note: The MP-124 and MP-10x/FXO do not support the Lifeline.
The Lifeline’s Splitter connects pins #1 and #4 to another source of an FXS port, and pins #2 and
#3 to the POTS phone. Refer to the Lifeline Splitter pinout in Figure
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3-10.
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MediaPack SIP
Figure 3-10: Lifeline Splitter Pinout & RJ-11 Connector for MP-10x/FXS
1 2 3 4
Lifeline Tip
1 2 -
Tip
3 -
Ring
Lifeline Ring
4 -
¾ To cable the MP-10x/FXS Lifeline phone, take these 3 steps:
1. Connect the Lifeline Splitter to port #4 (on the MP-104/FXS or MP-108/FXS) or to port #2 (on
the MP-102/FXS).
2. Connect the Lifeline phone to Port A on the Lifeline Splitter.
3. Connect an analog PSTN line to Port B on the Lifeline Splitter.
Note: The use of the Lifeline on network failure can be disabled using the
‘LifeLineType’ ini file parameter (described in Table 5-37 on page 128).
B: To PSTN wall port.
Phone to Port 1.
Lifeline to Port 4.
PSTN to Splitter (B).
Phone to Port 1.
Lifeline phone to Splitter (A).
Lifeline phone.
Version 4.6 37 June 2005
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MediaPack SIP
3.2 Installing the MP-11x
¾ To install the MP-11x, take these 3 steps:
1. Unpack the MP-11x (refer to Section 3.2.1 below).
2. Check the package contents (refer to Section
3. Mount the MP-11x (refer to Section
4. Cable the MP-11x (refer to Section
After connecting the MP-11x to the power source, the Ready and Power LEDs on the front panel
turn to green (after a self-testing period of about 2 minutes). Any malfunction in the startup
procedure changes the Fail LED to red and the Ready LED is turned off (refer to Table
page 27 for details on the MP-11x LEDs).
You’re now ready to start configuring the gateway (Section
3.2.1 Unpacking
¾ To unpack the MP-11x, take these 6 steps:
1. Open the carton and remove the packing materials.
2. Remove the MP-11x gateway from the carton.
3. Check that there is no equipment damage.
4. Check, retain and process any documents.
5. Notify AudioCodes or your local supplier of any damage or discrepancies.
6. Retain any diskettes or CDs.
3.2.2 below).
3.2.4 on page 39).
3.2.5 on page 33).
2-7 on
5 on page 47).
3.2.2 Package Contents
Ensure that in addition to the MP-11x, thepackage contains:
• AC power cable.
• Small plastic bag containing four anti-slide bumpers for desktop installation.
• A CD with software and documentation may be included.
MediaPack SIP User’s Manual 3. Installing the MediaPack
3.2.3 19-inch Rack Installation Package
Additional option is available for installing the MP-11x in a 19-inch rack. The 19-inch rack
installation package contains a single shelf (shown in Figure
device screws.
3-12 below) and eight shelf-to-
Figure
3.2.4 Mounting the MP-11x
The MP-11x can be mounted on a desktop (refer to Section 3.2.4.1 below), on a wall (refer to
Section
Figure
3.2.4.2) or installed in a standard 19-inch rack (refer to Section 3.2.4.2).
3-13 below describes the design of the MP-11x base.
Figure
3-13: View of the MP-11x Base
3-12: 19-inch Rack Shelf
3
2
1
3-4: View of the MP-11x Base
Table
Item # Functionality
1 Square slot used to attach anti-slide bumpers (for desktop mounting)
2 Oval notch used to attach the MP-11x to a wall
3 Screw opening used to attach the MP-11x to a 19-inch shelf rack
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MediaPack SIP
3.2.4.1 Mounting the MP-11x on a Desktop
Attach the four (supplied) anti-slide bumpers to the base of the MP-11x (refer to item #1 in Figure
3-13) and place it on the desktop in the position you require.
3.2.4.2 Mounting the MP-11x on a Wall
¾ To mount the MP-11x on a wall, take these 4 steps:
1. Drill four holes according to the following dimensions:
¾ Side-to-side distance 140 mm.
¾ Front-to-back distance 101.4 mm.
2. Insert a wall anchor of the appropriate size into each hole.
3. Fasten a DIN 96 3.5X20 wood screw (not supplied) into each of the wall anchors.
4. Position the four oval notches located on the base of the MP-11x (refer to item #2 in Figure
3-13) over the four screws and hang the MP-11x on them.
3.2.4.3 Installing the MP-11x in a 19-inch Rack
The MP-11x is installed in a standard 19-inch rack by placing it on a shelf preinstalled in the rack.
This shelf can be ordered separately from AudioCodes.
Figure
1
2
Item # Functionality
1 Standard rack holes used to attach the shelf to the rack
3-14: MP-11x Rack Mount
3-5: MP-11x Rack Mount
Table
2 Eight shelf-to-device screws
¾ To install the MP-11x in a 19-inch rack, take these 3 steps:
1. Use the shelf-to-device screws found in the package to attach one or two MP-11x devices to
the shelf.
2. Position the shelf in the rack and line up its side holes with the rack frame holes.
3. Use four standard rack screws to attach the shelf to the rack. These screws are not
MediaPack SIP User’s Manual 3. Installing the MediaPack
3.2.5 Cabling the MP-11x
Cable your MP-11x according to each section of Table 3-6. For detailed information on the MP-
11x rear panel connectors, refer to Table
Table
3-6: Cables and Cabling Procedure
Cable Cabling Procedure
Connect the Ethernet connection on the MP-11x directly to the network using a
standard RJ-45 Ethernet cable. For connector’s pinout refer to Figure 3-15 on page
RJ-45 Ethernet
cable
RJ-11 two-wire
telephone cords
Lifeline
RS-232 serial
cable
AC Power cable
41.
Note that when assigning an IP address to the MP-11x using HTTP (under step 1 in
Section 4.2.1), you may be required to disconnect this cable and re-cable it
differently.
Connect the RJ-11 connectors on the rear
panel of the MP-11x to fax machine, modem,
or phones (refer to Figure 3-6).
For detailed information on setting up the Lifeline, refer to the procedure under
Section 3.2.5.2 on page 42.
For detailed information on connecting the MP-1xx RS-232 port to your PC, refer to
Section 3.2.5.1 on page 41.
Connect the MP-11x power socket to the mains.
2-8 on page 28.
Ensure that the FXS ports are
connected to the correct devices,
otherwise damage can occur.
Figure 3-15: RJ-45 Ethernet Connector Pinout
RJ-45 Connector and Pinout
1 2 3 4 5 6 7 8
1 - Tx+
2 - Tx3 - Rx+
6 - Rx-
4, 5, 7, 8
not
connected
Figure 3-16: RJ-11 Phone Connector Pinout
RJ-11 Connector and Pinout
1 2 3 4
Not connected
1 2 -
Tip
3 -
Ring
Not connected
4 -
3.2.5.1 Connecting the MP-11x RS-232 Port to Your PC
Using a standard RS-232 straight cable (not a cross-over cable) with DB-9 connectors, connect
the MP-11x RS-232 port (using a DB-9 to PS/2 adaptor) to either COM1 or COM2 RS-232
communication port on your PC. The pinout of the PS/2 connector is shown below in Figure
For information on establishing a serial communications link with the MP-11x, refer to Section
10.2 on page 201.
3-17.
Figure 3-17: PS/2 Pinout
PS/2 Female Connector and Pinout
(TD) - Transmit Data
2
(GND) - Ground for Voltage
3
(RD) - Receive Data
6
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MediaPack SIP
3.2.5.2 Cabling the MP-11x Lifeline
The Lifeline (connected to port #1) provides a wired analog POTS phone connection to any PSTN
or PBX FXS port when there is no power, or the when network connection fails. Users can
therefore use the Lifeline phone even when the MP-11x is not powered on or not connected to
the network.
The Lifeline’s Splitter connects pins #1 and #4 to another source of an FXS port, and pins #2 and
#3 to the POTS phone. Refer to the Lifeline Splitter pinout in Figure
3-18.
Figure
3-18: Lifeline Splitter Pinout & RJ-11 Connector
1 2 3 4
Lifeline Tip
1 2 -
Tip
3 -
Ring
Lifeline Ring
4 -
¾ To cable the MP-11x Lifeline, take these 3 steps:
1. Connect the Lifeline Splitter to port #1 on the MP-11x.
2. Connect the Lifeline phone to Port A on the Lifeline Splitter.
3. Connect an analog PSTN line to Port B on the Lifeline Splitter.
Note: The use of the Lifeline on network failure can be disabled using the
‘LifeLineType’ ini file parameter (described in Table 5-37 on page 128).
The MediaPack is supplied with default networking parameters (show in Table 4-1 below) and
with an application software already resident in its flash memory (with factory default parameters).
Before you begin configuring the gateway, change its default IP address to correspond with your
network environment (refer to Section
on the MediaPack (refer to Section
For information on quickly setting up the MediaPack with basic parameters using a standard Web
browser, refer to Section
Table
FXS or FXO Default Value
4.3 on page 45.
4-1: MediaPack Default Networking Parameters
4.2) and learn about the configuration methods available
4.1 below).
FXS
FXO
MediaPack default subnet mask is 255.255.0.0, default gateway IP address is 0.0.0.0
4.1 Configuration Concepts
Users can utilize the MediaPack in a wide variety of applications, enabled by its parameters and
configuration files (e.g., Call Progress Tones (CPT)). The parameters can be configured and
configuration files can be loaded using:
• A standard Web Browser (described and explained in Section
• A configuration file referred to as the ini file. For information on how to use the ini file, refer to
Section
• An SNMP browser software (refer to Section
• The embedded Command Line Interface (refer to Section
• AudioCodes’ Element Management System (EMS) (refer to Section
AudioCodes’ EMS User’s Manual or EMS Product Description).
To upgrade the MediaPack (load new software or configuration files onto the gateway) use the
Software Upgrade wizard, available through the Web Interface (refer to Section
155), or alternatively use the BootP/TFTP configuration utility (refer to Section
166).
6 on page 163.
15 on page 227).
10.1.10.10
10.1.10.11
5 on page 47).
14 on page 223).
15.9 on page 239 and to
5.8.1 on page
7.3.1 on page
For information on the configuration files, refer to Section
6 on page 163.
4.2 Assigning the MediaPack IP Address
To assign an IP address to the MediaPack use one of the following methods:
• HTTP using a Web browser (refer to Section
• BootP (refer to Section
• DHCP (refer to Section
• Embedded command line interface (refer to Section
Use the ‘Reset’ button at any time to restore the MediaPack networking parameters to their
factory default values (refer to Section
Version 4.6 43 June 2005
4.2.2 on page 44).
7.2 on page 165).
10.1 on page 201).
4.2.1 below).
14 on page 223).
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MediaPack SIP
4.2.1 Assigning an IP Address Using HTTP
¾ To assign an IP address using HTTP, take these 8 steps:
1. Disconnect the MediaPack from the network and reconnect it to your PC using one of the
following two methods:
¾Use a standard Ethernet cable to connect the network interface on your PC to a port on
a network hub / switch. Use a second standard Ethernet cable to connect the MediaPack
to another port on the same network hub / switch.
¾Use an Ethernet cross-over cable (for the MP-1xx) or a standard Ethernet cable (for the
MP-11x) to directly connect the network interface on your PC to the MediaPack.
2. Change your PC’s IP address and subnet mask to correspond with the MediaPack factory
default IP address and subnet mask, shown in Table
address and subnet mask of your PC, refer to Windows™ Online Help (Start>Help).
4-1. For details on changing the IP
3. Access the MediaPack Embedded Web Server (refer to Section
4. In the ‘Quick Setup’ screen (shown in Figure
Mask’ and ‘Default Gateway IP Address’ fields under ‘IP Configuration’ to correspond with
your network IP settings. If your network doesn’t feature a default gateway, enter a dummy
value in the ‘Default Gateway IP Address’ field.
5. Click the Reset button and click OK in the prompt; the MediaPack applies the changes and
restarts.
Tip: Record and retain the IP address and subnet mask you assign the
MediaPack. Do the same when defining new username or password. If the
Embedded Web Server is unavailable (for example, if you’ve lost your
username and password), use the BootP/TFTP (Trivial File Transfer
Protocol) configuration utility to access the device, ‘reflash’ the load and
6. Disconnect your PC from the MediaPack or from the hub / switch (depending on the
connection method you used in step
7. Reconnect the MediaPack and your PC (if necessary) to the LAN.
8. Restore your PC’s IP address & subnet mask to what they originally were. If necessary,
restart your PC and re-access the MediaPack via the Embedded Web Server with its new
assigned IP address.
reset the password (refer to Appendix B on page 257 for detailed information
on using a BootP/TFTP configuration utility to access the device).
1).
4-1), set the MediaPack ‘IP Address’, ‘Subnet
5.3 on page 48).
4.2.2 Assigning an IP Address Using BootP
Note: BootP procedure can also be performed using any standard compatible
BootP server.
Tip: You can also use BootP to load the auxiliary files to the MediaPack (refer to
Section 5.8.2.1 on page 160).
¾ To assign an IP address using BootP, take these 3 steps:
1. Open the BootP application (supplied with the MediaPack software package).
2. Add client configuration for the MediaPack, refer to Section B.11.1 on page 263.
3. Use the reset button to physically reset the gateway causing it to use BootP; the MediaPack
changes its network parameters to the values provided by the BootP.
4.3 Configure the MediaPack Basic Parameters
To configure the MediaPack basic parameters use the Embedded Web Server’s ‘Quick Setup’
screen (shown in Figure
the ‘Quick Setup’ screen.
4-1 below). Refer to Section 5.3 on page 48 for information on accessing
Figure
4-1: Quick Setup Screen
¾ To configure basic SIP parameters, take these 9 steps:
1. If the MediaPack is connected to a router with Network Address Translation (NAT) enabled,
perform the following procedure. If it isn’t, leave the ‘NAT IP Address’ field undefined.
¾Determine the ‘public’ IP address assigned to the router (by using, for instance, router
Web management). Enter this public IP address in the ‘NAT IP Address’ field.
¾Enable the DMZ (Demilitarized Zone) configuration on the residential router for the LAN
port where the MediaPack gateway is connected. This enables unknown packets to be
routed to the DMZ port.
2. Under ‘SIP Parameters’, enter the MediaPack Domain Name in the field ‘Gateway Name’. If
the field is not specified, the MediaPack IP address is used instead (default).
3. When working with a Proxy server, set ‘Working with Proxy’ field to ‘Yes’ and enter the IP
address of the primary Proxy server in the field ‘Proxy IP Address’. When no Proxy is used,
the internal routing table is used to route the calls.
4. Enter the Proxy Name in the field ‘Proxy Name’. If Proxy name is used, it 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.
5. Configure ‘Enable Registration’ to ‘Yes’ or ‘No’:
‘No’ = the MediaPack does not register to a Proxy server/Registrar (default).
‘Yes’ = the MediaPack registers to a Proxy server/Registrar at power up and every
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MediaPack SIP
‘Registration Time’ seconds; The MediaPack sends a REGISTER request according to the
‘Authentication Mode’ parameter. For detailed information on the parameters ‘Registration
Time’ and ‘Authentication Mode’, refer to Table
6. Select the coder (i.e., vocoder) that best suits your VoIP system requirements. The default
coder is: G.7231 30 msec. To program the entire list of coders you want the MediaPack to
use, click the button on the left side of the ‘1
coders appears. Select coders according to your system requirements. Note that coders
higher on the list are preferred and take precedence over coders lower on the list.
Note: The preferred coder is the coder that the MediaPack uses as a first choice
for all connections. If the far end gateway does not use this coder, the
MediaPack negotiates with the far end gateway to select a coder that both
sides can use.
7. To program the Tel to IP Routing Table, press the arrow button next to ‘Tel to IP Routing
Table’. For information on how to configure the Tel to IP Routing Table, refer to Section
5.5.4.2 on page 83.
8. To program the Endpoint Phone Number Table, press the arrow button next to ‘Endpoint
Phone Number’. For information on how to configure the Endpoint Phone Number Table,
refer to Section
5.5.6 on page 97.
5-2 on page 57.
st
Coder’ field; the drop-down list for the 2nd to 5th
9. Click the Reset button and click OK in the prompt; The MediaPack applies the changes and
restarts.
You are now ready to start using the VoIP gateway. To prevent unauthorized access to the
MediaPack, it is recommended that you change the username and password that are used to
access the Web Interface. Refer to Section
username and password.
Tip: Once the gateway is configured correctly back up your settings by making a
copy of the VoIP gateway configuration (ini file) and store it in a directory on
your PC. This saved file can be used to restore configuration settings at a
future time. For information on backing up and restoring the gateway’s
configuration, refer to Section 5.6.3 on page 144.
5.6.5 on page 146 for details on how to change the
MediaPack SIP User’s Manual 5. Configuring the MediaPack
5 Configuring the MediaPack
The Embedded Web Server is used both for gateway configuration, including loading of
configuration files, and for run-time monitoring. The Embedded Web Server can be accessed
from a standard Web browser, such as Microsoft™ Internet Explorer, Netscape™ Navigator, etc.
Specifically, users can employ this facility to set up the gateway configuration parameters. Users
also have the option to remotely reset the gateway and to permanently apply the new set of
parameters.
5.1 Computer Requirements
To use the Embedded Web Server, the following is required:
• A computer capable of running your Web browser.
• A network connection to the VoIP gateway.
• One of the following compatible Web browsers:
¾ Microsoft™ Internet Explorer™ (version 6.0 and higher).
¾ Netscape™ Navigator™ (version 7.2 and higher).
Note: The browser must be Java-script enabled. If java-script is disabled, access to
the Embedded Web Server is denied.
5.2 Protection and Security Mechanisms
Access to the Embedded Web Server is controlled by the following protection and security
mechanisms:
• Dual access level username and password (refer to Section
• Read-only mode (refer to Section
• Disabling access (refer to Section
• Secured HTTP connection (HTTPS) (refer to Section
• Limiting access to a predefined list of IP addresses (refer to Section
• Managed access using a RADIUS server (refer to Section
5.2.2 below).
5.2.3 below).
12.1.2 on page 213) (MP-11x only).
5.2.1 Dual Access Level Username and Password
To prevent unauthorized access to the Embedded Web Server, two levels of security are
available: Administrator (also used for Telnet access) and Monitoring. Each employs a different
username and password. Users can access the Embedded Web Server as either:
5.2.1 below).
12.2 on page 217) (MP-11x only).
5.6.1.4 on page 120).
•Administrator - all Web screens are read-write and can be modified.
•Monitoring - all Web screens are read-only and cannot be modified. In addition, the following
screens cannot be accessed: ’Reset‘, ‘Save Configuration‘, ‘Software Upgrade Wizard’, ‘Load
Auxiliary Files’, ‘Configuration File’ and ‘Regional Settings’. The ’Change Password‘ screen
can only be used to change the monitoring password.
Default username ‘User’.
Default password ‘User’.
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MediaPack SIP
The first time a browser request is made, the user is requested to provide his Administrator or
Monitoring username and password to obtain access. Subsequent requests are negotiated by the
browser on behalf of the user, so that the user doesn’t have to re-enter the username and
password for each request, but the request is still authenticated (the Embedded Web Server uses
the MD5 authentication method supported by the HTTP 1.1 protocol).
For details on changing the Administrator and Monitoring username and password, refer to
Section
sensitive characters.
To reset the Administrator and Monitoring username and password to their defaults, enable the
ini file parameter ‘ResetWebPassword’.
5.6.5 on page 146. Note that the password and username can be a maximum of 19 case-
5.2.2 Limiting the Embedded Web Server to Read-Only Mode
Users can limit access to the Embedded Web Server to read-only mode by changing the ini file
parameter ‘DisableWebConfig’ to 1. In this mode all Web screens, regardless to the access level
used (Administrator or Monitoring), are read-only and cannot be modified. In addition, the
following screens cannot be accessed: ‘Quick Setup’, ‘Change Password’, ’Reset‘, ‘Save
Configuration‘, ‘Software Upgrade Wizard’, ‘Load Auxiliary Files’, ‘Configuration File’ and
‘Regional Settings’.
5.2.3 Disabling the Embedded Web Server
Access to the Embedded Web Server can be disabled by using the ini file parameter
‘DisableWebTask = 1’. The default is access enabled.
5.3 Accessing the Embedded Web Server
¾ To access the Embedded Web Server, take these 4 steps:
1. Open a standard Web-browsing application such as Microsoft™ Internet Explorer™ or
Netscape™ Navigator™.
2. In the Uniform Resource Locator (URL) field, specify the IP address of the MediaPack (e.g.,
http://10.1.10.10); the Embedded Web Server’s ‘Enter Network Password’ screen appears,
shown in Figure
5-1.
Figure
5-1: Embedded Web Server Login Screen
3. In the ‘User Name’ and ‘Password’ fields, enter the username (default: ‘Admin’) and
password (default: ‘Admin’). Note that the username and password are case-sensitive.
4. Click the OK button; the ‘Quick Setup’ screen is accessed (shown in Figure
MediaPack SIP User’s Manual 5. Configuring the MediaPack
r
g
r
5.3.1 Using Internet Explorer to Access the Embedded Web Server
Internet explorer’s security settings may block access to the gateway’s Web browser if they’re
configured incorrectly. In this case, the following message is displayed:
Unauthorized
Correct authorization is required for this area. Either your browser does not perform
authorization or your authorization has failed. RomPager server.
¾ To troubleshoot blocked access to Internet Explorer™, take these 2
steps
1. Delete all cookies from the Temporary Internet files. If this does not clear up the problem, the
security settings may need to be altered (refer to Step 2).
2. In Internet Explorer, Tools, Internet Options select the Security tab, and then select Custom
Level. Scroll down until the Logon options are displayed and change the setting to Prompt
for username and password and then restart the browser. This fixes any issues related to
domain use logon policy.
5.4 Getting Acquainted with the Web Interface
Figure 5-2 shows the general layout of the Web Interface screen.
Main Menu
Ba
Submenu
Ba
Corporate
Lo
Figure
o
5-2: MediaPack Web Interface
Title Bar
Main Action
Frame
Control
Protocol
The Web Interface screen features the following components:
•Title bar - contains three configurable elements: corporate logo, a background image and the
product’s name. For information on how to modify these elements, refer to Section
10.5 on
page 206.
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MediaPack SIP
•Main menu bar - always appears on the left of every screen to quickly access parameters,
submenus, submenu options, functions and operations.
• Submenu bar - appears on the top of screens and contains submenu options.
• Main action frame - the main area of the screen in which information is viewed and
configured.
•Corporate logo – AudioCodes’ corporate logo. For information on how to remove this logo
Section
•Control Protocol – the MediaPack control protocol.
10.5 on page 206.
5.4.1 Main Menu Bar
The main menu bar of the Web Interface is divided into the following 7 menus:
•Quick Setup – Use this menu to configure the gateway’s basic settings; for the full list of
configurable parameters go directly to ‘Protocol Management’ and ‘Advanced Configuration’
menus. An example of the Quick Setup configuration is described in Section
•Protocol Management – Use this menu to configure the gateway’s control protocol
parameters and tables (refer to Section
•Advanced Configuration – Use this menu to set the gateway’s advanced configuration
parameters (for advanced users only) (refer to Section
4.3 on page 45.
5.5 on page 51).
5.6 on page 114).
•Status & Diagnostics – Use this menu to view and monitor the gateway’s channels, Syslog
messages, hardware / software product information, and to assess the gateway’s statistics
and IP connectivity information (refer to Section
•Software Update – Use this menu when you want to load new software or configuration files
onto the gateway (refer to Section
•Save Configuration – Use this menu to save configuration changes to the non-volatile flash
memory (refer to Section
•Reset – Use this menu to remotely reset the gateway. Note that you can choose to save the
gateway configuration to flash memory before reset (refer to Section
When positioning your curser over a parameter name (or a table) for more than 1 second, a short
description of this parameter is displayed. Note that those parameters that are preceded with an
exclamation mark (!) are not changeable on-the-fly and require reset.
5.4.2 Saving Changes
To save changes to the volatile memory (RAM) press the Submit button (changes to parameters
with on-the-fly capabilities are immediately available, other parameter are updated only after a
gateway reset). Parameters that are only saved to the volatile memory revert to their previous
settings after hardware reset. When performing a software reset (i.e., via Web or SNMP) you can
choose to save the changes to the non-volatile memory. To save changes so they are available
after a power fail, you must save the changes to the non-volatile memory (flash). When Save Configuration is performed, all parameters are saved to the flash memory.
5.7 on page 147).
5.8 on page 155).
5.9 on page 161).
5.9 on page 161).
To save the changes to flash, refer to Section
5.9 on page 161.
5.4.3 Entering Phone Numbers in Various Tables
Phone numbers entered into various tables on the gateway, such as the Tel to IP routing table,
must be entered without any formatting characters. For example, if you wish to enter the phone
number 555-1212, it must be entered as 5551212 without the hyphen (-). If the hyphen is entered,
the entry does not work. The hyphen character is used in number entry only, as part of a range
definition. For example, the entry [20-29] means ‘all numbers in the range 20 to 29’.
Note 1: The Supported and Required headers contain the ‘100rel’ parameter.
Note 2: MediaPack sends PRACK message if 180/183 response is received with
‘100rel’ in the Supported or the Required headers.
Port allocation algorithm for IP to Tel calls.
You can select one of the following methods:
•By phone number [0] = Select the gateway port according to the called number
(called number is defined in the ‘Endpoint Phone Number’ table).
•Cyclic Ascending [1] = Select the next available channel in an ascending cycle
order. Always select the next higher channel number in the hunt group. When the
gateway reaches the highest channel number in the hunt group, it selects the
lowest channel number in the hunt group and then starts ascending again.
•Ascending [2] = Select the lowest available channel. Always start at the lowest
channel number in the hunt group and if that channel is not available, select the
next higher channel.
•Cyclic Descending [3] = Select the next available channel in descending cycle
order. Always select the next lower channel number in the hunt group. When the
gateway reaches the lowest channel number in the hunt group, it selects the
highest channel number in the hunt group and then starts descending again.
•Descending [4] = Select the highest available channel. Always start at the highest
channel number in the hunt group and if that channel is not available, select the
next lower channel.
•Number + Cyclic Ascending [5] = First select the gateway port according to the
called number (called number is defined in the ‘Endpoint Phone Number’ table). If
the called number isn’t found, then select the next available channel in ascending
cyclic order. Note that if the called number is found, but the port associated with this
number is busy, the call is released.
The default method is ‘By Phone Number’.
Enable Early Media
[EnableEarlyMedia]
Session-Expires Time
[SIPSessionExpires]
Minimum Session-Expires
[MINSE]
No [0] = Early Media is disabled (default).
Yes [1] = Enable Early Media.
If enabled, the gateway sends 183 Session Progress response with SDP (instead of 180
Ringing), allowing the media stream to be set up prior to the answering of the call.
Note that to send 183 response you must also set the parameter ‘ProgressIndicator2IP’
to 1. If it is equal to 0, 180 Ringing response is sent.
Note: Generally, this parameter is set to 1.
Determines the timeout (in seconds) for keeping a re-INVITE message alive within a SIP
session. The SIP session is refreshed (using INVITE) each time this timer expires.
The default is 0 (not activated).
Defines the time (in seconds) that is used in the Min-SE header field. This field defines
the minimum time that the user agent supports for session refresh.
The valid range is 10 to 100000. The default value is 90.
The Asserted ID mode defines the header that is used in the generated INVITE request.
The header also depends on the calling Privacy: allowed or restricted.
The P-asserted (or P-preferred) headers are used to present the originating party’s
Caller ID. The Caller ID is composed of a Calling Number and (optionally) a Calling
Name.
P-asserted (or P-preferred) headers are used together with the Privacy header. If Caller
ID is restricted the ‘Privacy: id’ is included. Otherwise for allowed Caller ID the ‘Privacy:
none’ is used. If Caller ID is restricted (received from Tel or configured in the gateway),
the From header is set to <anonymous@anonymous.invalid>.
Determines the SIP signaling method used to establish and convey a fax session after a
fax is detected.
No Fax [0] = No fax negotiation using SIP signaling (default).
T.38 Relay [1] = Initiates T.38 fax relay.
G.711 Transport [2] = Initiates fax using the coder G.711 A-law/µ-law with
adaptations (refer to note 1).
Fax Fallback [3] = Initiates T.38 fax relay. If the T.38 negotiation fails, the
gateway re-initiates a fax session using the coder G.711 A-law/µ-law with adaptations
(see note 1).
Note 1: Fax adaptations:
Echo Canceller = On
Silence Compression = Off
Echo Canceller Non-Linear Processor Mode = Off
Dynamic Jitter Buffer Minimum Delay = 40
Dynamic Jitter Buffer Optimization Factor = 13
Note 2: If the gateway initiates a fax session using G.711 (option 2 and possibly 3), a
‘gpmd’ attribute is added to the SDP in the following format:
For A-law: ‘a=gpmd:0 vbd=yes;ecan=on’. For µ-law: ‘a=gpmd:8 vbd=yes;ecan=on’.
Note 3: When ‘IsFaxUsed’ is set to 1, 2 or 3 the parameter ‘FaxTransportMode’ is
ignored.
Initiate T.38 on Preamble [0] = Terminating fax gateway initiates T.38 session on
receiving of HDLC preamble signal from fax (default)
Initiate T.38 on CED [1] = Terminating fax gateway initiates T.38 session on
receiving of CED answer tone from fax.
Note: This parameters is applicable only if ‘IsFaxUsed = 1’.
Determines the default transport layer used for outgoing SIP calls initiated by the
gateway.
UDP [0] (default).
TCP [1].
TLS [2] (SIPS) (MP-11x only).
Note: It is recommended to use TLS to communicate with a SIP Proxy and not for direct
gateway-gateway communication.
Local UDP port used to receive SIP messages.
The default value is 5060.
Local TCP port used to receive SIP messages (MP-11x only).
The default value is 5060.
Local TLS port used to receive SIP messages.
The default value is 5061.
Note: The value of ‘TLSLocalSIPPort’ must be different to the value of
‘TCPLocalSIPPort’.
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Table 5-1: Protocol Definition, General Parameters (continues on pages 52 to 55)
Parameter Description
Enable SIPS
[EnableSIPS]
SIP Destination Port
[SIPDestinationPort]
Use “user=phone” in SIP URL
[IsUserPhone]
Use “user=phone” in From
header
[IsUserPhoneInFrom]
Tel to IP No Answer Timeout
[IPAlertTimeout]
Enable Remote Party ID
[EnableRPIheader]
Add Number Plan and Type to
Remote Party ID Header
[AddTON2RPI]
Use Source Number as
Display Name
[UseSourceNumberAsDispl
ayName]
Use Display Name as Source
Number
[UseDisplayNameAsSource
Number]
Play Ringback Tone to IP
[PlayRBTone2IP]
Play Ringback Tone to Tel
[PlayRBTone2Tel]
Enables secured SIP (SIPS) connections over multiple hops (MP-11x only).
Disable [0] (default).
Enable [1].
When SIPTransportType = 2 (TLS) and EnableSIPS is disabled, TLS is used for the
next network hop only.
When SIPTransportType = 2 (TLS) or 1 (TCP) and EnableSIPS is enabled, TLS is used
through the entire connection (over multiple hops).
Note: If SIPS is enabled and SIPTransportType = UDP, the connection fails.
SIP UDP destination port for sending SIP messages.
The default value is 5060.
No [0] = ‘user=phone’ string isn’t used in SIP URL.
Yes [1] = ‘user=phone’ string is part of the SIP URL (default).
No [0] = Doesn’t use ‘;user=phone’ string in From header (default).
Yes [1] = ‘;user=phone’ string is part of the From header.
Defines the time (in seconds) the gateway waits for a 200 OK response from the called
party (IP side) after sending an INVITE message. If the timer expires, the call is
released.
The valid range is 0 to 3600. The default value is 180.
Enable Remote-Party-ID (RPI) headers for calling and called numbers for TelÆIP calls.
Disable [0] (default).
Enable [1] = RPI headers are generated in SIP INVITE messages for both called and
calling numbers.
No [0] = TON/PLAN parameters aren’t included in the RPID header.
Yes [1] = TON/PLAN parameters are included in the RPID header (default).
If RPID header is enabled (EnableRPIHeader = 1) and ‘AddTON2RPI=1’, it is possible
to configure the calling and called number type and number plan using the Number
Manipulation tables for TelÆIP calls.
No [0] = Interworks the Tel calling name to SIP Display Name (default).
Yes [1] = Set Display Name to Calling Number if not configured.
Applicable to TelÆIP calls. If enabled and calling party name is not defined
(CallerDisplayInfoX = <name> is not specified per gateway’s x port), the calling number
is used instead.
No [0] = Interworks the IP Source Number to the Tel Source Number (default).
Yes [1] = Sets the Tel Source Number to IP Display Name.
Applicable to IPÆTel calls.
If enabled, the outgoing Source Number is set to the IP Display Name and Presentation
is set to Allowed. If there isn’t a Display Name, the user part of the SIP URI is used as
the Source Number, and the Presentation is set to Restricted.
For example:
When the following is received ’from: 100 <sip:200@201.202.203.204>’, the outgoing
Source Number is set to ’100’, the Display Name is set to ’100’ and the Presentation is
set to Allowed (0).
When the following is received ‘from: <sip:100@101.102.103.104>’, the outgoing
Source Number is set to ‘100’ and the Presentation is set to Restricted (1).
Don’t Play [0] = Ringback tone isn’t played to the IP side of the call (default).
Play [1] = Ringback tone is played to the IP side of the call after SIP 183
session progress response is sent.
Note 1: To enable the gateway to send a 183 response, set ‘EnableEarlyMedia’ to 1.
Note 2: If ‘EnableDigitDelivery = 1’, the gateway doesn’t play a Ringback tone to IP and
doesn’t send a 183 response.
Don’t Play [0] = Ringback Tone isn’t played.
Always Play [1] = Ringback Tone is played to the Tel side of the call when 180/183
response is received.
Play According to PI [3] = N/A.
Play According to 180/183 [2] = Ringback Tone is played to the Tel side of the call if no
SDP is received in 180/183 responses. If 180/183 with SDP message is received, the
gateway cuts through the voice channel and doesn’t play Ringback tone (default).
MediaPack SIP User’s Manual 5. Configuring the MediaPack
Table 5-1: Protocol Definition, General Parameters (continues on pages 52 to 55)
Parameter Description
Retransmission Parameters
SIP T1 Retransmission Timer
[msec]
[SipT1Rtx]
SIP T2 Retransmission Timer
[msec]
[SipT2Rtx]
SIP Maximum Rtx
[SIPMaxRtx]
The time interval (in msec) between the first transmission of a SIP message and the first
retransmission of the same message.
The default is 500.
Note: The time interval between subsequent retransmissions of the same SIP message
starts with SipT1Rtx and is multiplied by two until SipT2Rtx.
For example (assuming that SipT1Rtx = 500 and SipT2Rtx = 4000):
The first retransmission is sent after 500 msec.
The second retransmission is sent after 1000 (2*500) msec.
The third retransmission is sent after 2000 (2*1000) msec.
The fourth retransmission and subsequent retransmissions until SIPMaxRtx are sent
after 4000 (2*2000) msec.
The maximum interval (in msec) between retransmissions of SIP messages.
The default is 4000.
Note: The time interval between subsequent retransmissions of the same SIP message
starts with SipT1Rtx and is multiplied by two until SipT2Rtx.
Number of UDP retransmissions of SIP messages.
The range is 1 to 7.
The default value is 7.
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5.5.1.2 Proxy & Registration Parameters
Use this screen to configure parameters that are associated with Proxy and Registration.
¾ To configure the Proxy & Registration parameters, take these 4 steps:
1. Open the ‘Proxy & Registration’ parameters screen (Protocol Management menu >
MediaPack SIP User’s Manual 5. Configuring the MediaPack
Table 5-2: Proxy & Registration Parameters (continues on pages 57 to 60)
Parameter Description
Enable Proxy
[IsProxyUsed]
Proxy Name
[ProxyName]
Proxy IP Address
[ProxyIP]
Gateway Name
[SIPGatewayName]
Gateway Registration Name
[GWRegistrationName]
First Redundant Proxy IP
Address
[ProxyIP]
Second Redundant Proxy
IP Address
[ProxyIP]
Don’t Use Proxy [0] = Proxy isn’t used, the internal routing table is used instead
(default).
Use Proxy [1] = Proxy is used.
If you are using a Proxy server, enter the IP address of the primary Proxy server in the
Proxy IP address field.
If you are not using a Proxy server, you must configure the Tel to IP Routing table on the
gateway (described in Section
Defines the Home Proxy Domain Name.
If specified, the Proxy Name is used as Request-URI in REGISTER, INVITE and other SIP
messages. If not specified, the Proxy IP address is used instead.
IP address (and optionally port number) of the primary Proxy server you are using.
Enter the IP address as FQDN or in dotted format notation (for example 201.10.8.1).
You can also specify the selected port in the format: <IP Address>:<port>.
This parameter is applicable only if you select ‘Yes’ in the ‘Is Proxy Used’ field.
If you enable Proxy Redundancy (by setting EnableProxyKeepAlive=1), the gateway can
work with up to three Proxy servers. If there is no response from the primary Proxy, the
gateway tries to communicate with the redundant Proxies. When a redundant Proxy is
found, the gateway either continues working with it until the next failure occurs or reverts
to the primary Proxy (refer to the ‘Redundancy Mode’ parameter). If none of the Proxy
servers respond, the gateway goes over the list again.
The gateway also provides real time switching (hotswap mode), between the primary and
redundant proxies (‘IsProxyHotSwap=1’). If the first Proxy doesn’t respond to INVITE
message, the same INVITE message is immediately sent to the second Proxy.
Note 1: If ‘EnableProxyKeepAlive=1’, the gateway monitors the connection with the
Proxies by using keep-alive messages (OPTIONS).
Note 2: To use Proxy Redundancy, you must specify one or more redundant Proxies
using multiple ’ProxyIP= <IP address>’ definitions.
Note 3: When port number is specified (e.g., domain.com:5080), DNS SRV queries aren’t
performed, even if ‘EnableProxySRVQuery’ is set to 1.
Use this parameter to assign a name to the device (For example: ‘gateway1.com’). Ensure
that the name you choose is the one that the Proxy is configured with to identify your
media gateway.
Note: If specified, the gateway Name is used as the host part of the SIP URL, in both ‘To’
and ‘From’ headers. If not specified, the gateway IP address is used instead (default).
Defines the user name that is used in From and To headers of REGISTER messages.
Applicable only to single registration per gateway (’AuthenticationMode = 1).
If ‘GWRegistrationName’ isn’t specified (default), the ’Username’ parameter is used
instead.
Note: If ‘AuthenticationMode=0’, all the gateway’s endpoints are registered with a user
name that equals to the endpoint’s phone number.
IP addresses of the first redundant Proxy you are using.
Enter the IP address as FQDN or in dotted format notation (for example 192.10.1.255).
You can also specify the selected port in the format: <IP Address>:<port>.
Note 1: This parameter is available only if you select ‘Yes’ in the ‘Enable Proxy’ field.
Note 2: When port number is specified, DNS SRV queries aren’t performed, even if
‘EnableProxySRVQuery’ is set to 1.
ini file note: The IP address of the first redundant Proxy is defined by the second
repetition of the ini file parameter ‘ProxyIP’.
IP addresses of the second redundant Proxy you are using.
Enter the IP address as FQDN or in dotted format notation (for example 192.10.1.255).
You can also specify the selected port in the format: <IP Address>:<port>.
Note 1: This parameter is available only if you select ‘Yes’ in the ‘Enable Proxy’ field.
Note 2: When port number is specified, DNS SRV queries aren’t performed, even if
‘EnableProxySRVQuery’ is set to 1.
ini file note: The IP address of the second redundant Proxy is defined by the third
repetition of the ini file parameter ‘ProxyIP’.
5.5.4.2 on page 83).
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Table 5-2: Proxy & Registration Parameters (continues on pages 57 to 60)
Parameter Description
Third Redundant Proxy IP
Address
[ProxyIP]
Enable SRV Queries
[EnableSRVQuery]
Enable Proxy SRV Queries
[EnableProxySRVQuery]
Redundancy Mode
[ProxyRedundancyMode]
Is Proxy Trusted
[IsTrustedProxy]
Enable Registration
[IsRegisterNeeded]
Registrar Name
[RegistrarName]
IP addresses of the third redundant Proxy you are using.
Enter the IP address as FQDN or in dotted format notation (for example 192.10.1.255).
You can also specify the selected port in the format: <IP Address>:<port>.
Note 1: This parameter is available only if you select ‘Yes’ in the ‘Enable Proxy’ field.
Note 2: When port number is specified, DNS SRV queries aren’t performed, even if
‘EnableProxySRVQuery’ is set to 1.
ini file note: The IP addresses of the third redundant Proxy is defined by the forth
repetition of the ini file parameter ‘ProxyIP’.
Enables the use of DNS Service Record (SRV) queries to resolve Proxy and Registrar
servers and to resolve all domain names that appear in the Contact and Record-Route
headers.
Disable [0] (default).
Enable [1].
If enabled and the Proxy / Registrar IP address parameter or the domain name in the
Contact / Record-Route headers contains a domain name without port definition, an SRV
query is performed. The gateway uses the first host name received from the SRV query.
The gateway then performs DNS A-record query for the host name to locate an IP
address.
If the Proxy / Registrar IP address parameter or the domain name in the Contact / RecordRoute headers contains a domain name with port definition, the gateway performs a
regular DNS A-record query.
To enable SRV queries only for Proxy servers, set the parameter ‘EnableProxySRVQuery’
to 1.
Enables the use of DNS Service Record (SRV) queries to discover Proxy servers.
Disable [0] = Disabled (default).
Enable [1] = Enabled.
If enabled and the Proxy IP address parameter contains a domain name without port
definition (e.g., ProxyIP = domain.com), an SRV query is performed. The SRV query
returns up to four Proxy host names and their weights. The gateway then performs DNS
A-record queries for each Proxy host name (according to the received weights) to locate
up to four Proxy IP addresses. Therefore, if the first SRV query returns two domain
names, and the A-record queries return 2 IP addresses each, no more searches are
performed.
If the Proxy IP address parameter contains a domain name with port definition (e.g.,
ProxyIP = domain.com:5080), the gateway performs a regular DNS A-record query.
Note: When enabled, SRV queries are used to discover Proxy servers even if the
parameter ‘EnableSRVQuery’ is disabled.
Parking [0] = Gateway continues working with the last active Proxy until the next failure
(default).
Homing [1] = Gateway always tries to work with the primary Proxy server (switches back
to the main Proxy whenever it is available).
Note: To use Redundancy Mode, enable Keep-alive with Proxy option (Enable Proxy
Keep Alive = Yes).
This parameter isn’t applicable and must always be set to ‘Yes’ [1].
The parameter ‘AssertedIdMode’ should be used instead.
No [0] = Gateway doesn’t register to Proxy / Registrar (default).
Yes [1] = Gateway registers to Proxy / Registrar when the device is powered up and every
RegistrationTime seconds.
Note: The gateway sends a REGISTER request for each channel or for the entire gateway
(according to the AuthenticationMode parameter).
Registrar Domain Name.
If specified, the name is used as Request-URI in REGISTER messages.
If isn’t specified (default), the Registrar IP address or Proxy name or Proxy IP address is
used instead.
Use Routing Table for Host
Names and Profiles
[AlwaysUseRouteTable]
Always Use Proxy
[AlwaysSendToProxy]
IP address and optionally port number of Registrar server.
Enter the IP address in dotted format notation, for example 201.10.8.1:<5080>.
Note 1: If not specified, the REGISTER request is sent to the primary Proxy server (refer
to ‘Proxy IP address’ parameter).
Note 2: When port number is specified, DNS SRV queries aren’t performed, even if
‘EnableSRVQuery’ is set to 1.
Time (in seconds) for which registration to a Proxy server is valid. The value is used in the
‘Expires = ‘ header. Typically a value of 3600 is assigned, for one hour registration.
The gateway resumes registration when half the defined timeout period expires.
The default is 3600 seconds.
Defines the re-registration timing (in percentage). The timing is a percentage of the reregister timing set by the Registration server.
The valid range is 50 to 100. The default value is 50.
For example: If ‘RegistrationTimeDivider = 70’ (%) and Registration Expires time = 3600,
the gateway resends its registration request after 3600 x 70% = 2520 sec.
Defines the time period (in seconds) after which a Registration request is resent if
registration fails with 4xx, or there is no response from the Proxy/Registrar.
The default is 30 seconds. The range is 10 to 3600.
Determines the method the gateway uses to subscribe to an MWI server.
Per Endpoint [0] = Each endpoint subscribes separately. This method is usually used for
FXS gateways (default).
Per Gateway [1] = Single subscription for the entire gateway. This method is usually used
for FXO gateways.
No [0] = Disable (default).
Yes [1] = Keep alive with Proxy is enabled.
If enabled, OPTIONS SIP message is sent every ‘Proxy Keep-Alive Time’.
Note: This parameter must be enabled when Proxy redundancy is used.
Defines the Proxy keep-alive time interval (in seconds) between OPTIONS messages.
The default value is 60 seconds.
No [0] = Use the gateway’s IP address in keep-alive OPTIONS messages (default).
Yes [1] = Use ‘GatewayName’ in keep-alive OPTIONS messages.
The OPTIONS Request-URI host part contains either the gateway’s IP address or a string
defined by the parameter ‘Gatewayname’.
The gateway uses the OPTIONS request as a keep-alive message to its primary and
redundant Proxies.
No [0] = Gateway fallback is not used (default).
Yes [1] = Internal Tel to IP Routing table is used when Proxy servers are not available.
When the gateway falls back to the internal Tel to IP Routing table, the gateway continues
scanning for a Proxy. When the gateway finds an active Proxy, it switches from internal
routing back to Proxy routing.
Note: To enable the redundant Proxies mechanism set ‘EnableProxyKeepAlive’ to 1.
Determines if the local Tel to IP routing table takes precedence over a Proxy for routing
calls.
No [0] = Only Proxy is used to route calls (default).
Yes [1] = The Proxy checks the 'Destination IP Address' field in the 'Tel to IP Routing'
table for a match with the outgoing call. Only if a match is not found, a Proxy is used.
Note: Applicable only if Proxy is not always used (‘AlwaysSendToProxy’ = 0,
‘SendInviteToProxy’ = 0).
Use the internal Tel to IP routing table to obtain the URL Host name and (optionally) an IP
profile (per call), even if Proxy server is used.
No [0] = Don’t use (default).
Yes [1] = Use.
Note: This Domain name is used, instead of Proxy name or Proxy IP address, in the
INVITE SIP URL.
No [0] = Use standard SIP routing rules (default).
Yes [1] = All SIP messages and Responses are sent to Proxy server. Note: Applicable only if Proxy server is used.
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Table 5-2: Proxy & Registration Parameters (continues on pages 57 to 60)
Parameter Description
Send All INVITE to Proxy
[SendInviteToProxy]
Enable Proxy Hot-Swap
[IsProxyHotSwap]
Number of RTX Before HotSwap
[ProxyHotSwapRtx]
User Name
[UserName]
Note: The Authentication
table can be used instead.
Password
[Password]
Cnonce
[Cnonce]
Authentication Mode
[AuthenticationMode]
No [0] = INVITE messages, generated as a result of Transfer or Redirect, are sent
directly to the URL (according to the refer-to header in the REFER message or contact
header in 30x response) (default).
Yes [1] = All INVITE messages, including those generated as a result of Transfer or
Redirect are sent to Proxy.
Note: Applicable only if Proxy server is used and ‘AlwaysSendtoProxy=0’.
Enable Proxy Hot-Swap redundancy mode.
No [0] = Disabled (default).
Yes [1] = Enabled.
If Hot Swap is enabled, SIP INVITE message is first sent to the primary Proxy server. If
there is no response from the primary Proxy server for ‘Number of RTX before Hot-Swap’
retransmissions, the INVITE message is resent to the redundant Proxy server.
Number of retransmitted INVITE messages before call is routed (hot swapped) to another
Proxy.
The range is 1-30. The default is 3.
Note: This parameter is also used for alternative routing using the Tel to IP Routing table.
If a domain name in the routing table is resolved into 2 IP addresses, and if there is no
response for ‘ProxyHotSwapRtx’ retransmissions to the INVITE message that is sent to
the first IP address, the gateway immediately initiates a call to the second IP address.
Username used for Registration and for Basic/Digest authentication process with Proxy /
Registrar.
Parameter doesn’t have a default value (empty string).
Note: Applicable only if single gateway registration is used (‘Authentication Mode =
Authentication Per gateway’).
Password used for Basic/Digest authentication process with Proxy / Registrar. Single
password is used for all gateway ports.
The default is ‘Default_Passwd’.
Note: The Authentication table can be used instead.
String used by the server and client to provide mutual authentication. (Free format i.e.,
‘Cnonce = 0a4f113b’).
The default is ‘Default_Cnonce’.
Per Endpoint [0] = Registration & Authentication separately for each endpoint (default).
Per gateway [1] = Single Registration & Authentication for the gateway.
Per Ch. Select Mode [2] = N/A.
Usually Authentication on a per endpoint basis is used for FXS gateways, in which each
endpoint registers (and authenticates) separately with its own username and password.
Single Registration and Authentication (Authentication Mode=1) is usually defined for FXO
gateways.
MediaPack SIP User’s Manual 5. Configuring the MediaPack
5.5.1.3 Coders
From the Coders screen you can configure the first to fifth preferred coders (and their
corresponding ptimes) for the gateway. The first coder is the highest priority coder and is used by
the gateway whenever possible. If the far end gateway cannot use the coder assigned as the first
coder, the gateway attempts to use the next coder and so forth.
¾ To configure the Gateway’s coders, take these 6 steps:
1. Open the ‘Coders’ screen (Protocol Management menu > Protocol Definition submenu >
Coders option); the ‘Coders’ screen is displayed.
Figure
2. From the coder drop-down list, select the coder you want to use. For the full list of available
coders and their corresponding ptimes, refer to Table
Note: Each coder can appear only once.
3. From the drop-down list to the right of the coder list, select the size of the Voice Packet
(ptime) used with this coder in milliseconds. Selecting the size of the packet determines how
many coder payloads are combined into one RTP (voice) packet.
Note 1: The ptime packetization period depends on the selected coder name.
Note 2: If not specified, the ptime gets a default value.
Note 3: The ptime specifies the maximum packetization time the gateway can receive.
4. Repeat steps 2 and 3 for the second to fifth coders (optional).
5-5: Coders Screen
5-3.
5. Click the Submit button to save your changes.
6. To save the changes so they are available after a power fail, refer to Section
161.
5.9 on page
Note: Only the ptime of the first coder in the defined coder list is declared in INVITE
/ 200 OK SDP, even if multiple coders are defined.
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Table 5-3: ini File Coder Parameter
Parameter Description
CoderName
Enter the coders in the format: CoderName=<Coder>,<ptime>.
For example:
CoderName = g711Alaw64k,20
CoderName = g711Ulaw64k,40
CoderName = g7231,90
Note 1: This parameter (CoderName) can appear up to 10 times.
Note 2: The coder name is case-sensitive.
You can select the following coders:
g711Alaw64k – G.711 A-law.
g711Ulaw64k – G.711 µ-law.
g7231 – G.723.1 6.3 kbps (default).
g7231r53 – G.723.1 5.3 kbps.
g726 – G.726 ADPCM 32 kbps (Payload Type = 2).
g729 – G.729A.
g729_AnnexB – G.729 Annex B.
Note: If the coder G.729 is selected, the gateway includes ‘annexb=no’ in the SDP of the
relevant SIP messages. If G.729 Annex B is selected, ‘annexb=yes’ is included. An
exception to this logic is when the remote gateway is a Cisco device (IsCiscoSCEMode).
The RTP packetization period (ptime, in msec) depends on the selected coder name, and
can have the following values:
MediaPack SIP User’s Manual 5. Configuring the MediaPack
5.5.1.4 DTMF & Dialing Parameters
Use this screen to configure parameters that are associated with DTMF and dialing.
¾ To configure the dialing parameters, take these 4 steps:
1. Open the ‘DTMF & Dialing’ screen (Protocol Management menu > Protocol Definition
submenu > DTMF & Dialing option); the ‘DTMF & Dialing’ parameters screen is displayed.
Figure
5-6: DTMF & Dialing Parameters Screen
2. Configure the DTMF & Dialing parameters according to Table
3. Click the Submit button to save your changes.
4. To save the changes so they are available after a power fail, refer to Section
161.
Table
5-4: DTMF & Dialing Parameters (continues on pages 63 to 65)
Parameter Description
Max Digits in Phone Num
[MaxDigits]
Note: Digit Mapping Rules
can be used instead.
Inter Digits Timeout [sec]
[TimeBetweenDigits]
Maximum number of digits that can be dialed.
The valid range is 1 to 49.
The default value is 5.
Note: Dialing ends when the maximum number of digits is dialed, the Interdigit Timeout
expires, the '#' key is dialed, or a digit map pattern is matched.
Time in seconds that the gateway waits between digits dialed by the user. When the
Interdigit Timeout expires, the gateway attempts to dial the digits already received.
The valid range is 1 to 10 seconds. The default value is 4 seconds.
5-4.
5.9 on page
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Table 5-4: DTMF & Dialing Parameters (continues on pages 63 to 65)
Parameter Description
Use Out-of-Band DTMF
[IsDTMFUsed]
Out-of-Band DTMF Format
[OutOfBandDTMFFormat]
Declare RFC 2833 in SDP
[RxDTMFOption]
DTMF RFC 2833 Negotiation
[TxDTMFOption]
RFC 2833 Payload Type
[RFC2833PayloadType]
Use INFO for Hook-Flash
[IsHookFlashUsed]
Use out-of-band signaling to relay DTMF digits.
No [0] = DTMF digits are sent in-band (default).
Yes [1] = DTMF digits are sent out-of-band according to the parameter ‘Out-of-band
DTMF format’.
Note: When out-of-band DTMF transfer is used, the parameter ‘DTMF Transport Type’
is automatically set to 0 (erase the DTMF digits from the RTP stream).
The exact method to send out-of-band DTMF digits.
INFO (Nortel) [1] = Sends DTMF digits according with IETF <draft-choudhuri-sipinfo-digit-00>.
INFO (Cisco) [2] = Sends DTMF digits according with Cisco format (default).
NOTIFY (3Com) [3] = NOTIFY format <draft-mahy-sipping-signaled-digits-01.txt>.
Note 1: To use out-of-band DTMF, set ‘IsDTMFUsed=1’.
Note 2: When using out-of-band DTMF, the ‘DTMFTransportType’ parameter is
automatically set to 0, to erase the DTMF digits from the RTP stream.
Defines the supported Receive DTMF negotiation method.
No [0] = Don’t declare RFC 2833 Telephony-event parameter in SDP
Yes [3] = Declare RFC 2833 Telephony-event parameter in SDP (default)
The MediaPack is designed to always be receptive to RFC 2833 DTMF relay packets.
Therefore, it is always correct to include the ‘Telephony-event’ parameter as a default in
the SDP. However some gateways use the absence of the ‘telephony-event’ from the
SDP to decide to send DTMF digits in-band using G.711 coder, if this is the case you
can set ‘RxDTMFOption=0’.
Disable [0] = No negotiation, DTMF digit is sent according to the parameters ‘DTMF
Transport Type’ and ‘RFC2833PayloadType’ (default).
Enable [4] = Enable RFC 2833 payload type (PT) negotiation.
Note 1: This parameter is applicable only if ‘IsDTMFUsed=0’ (out-of-band DTMF is not
used).
Note 2: If enabled, the gateway:
• Negotiates RFC 2833 payload type using local and remote SDPs.
• Sends DTMF packets using RFC 2833 PT according to the PT in the received SDP.
• Expects to receive RFC 2833 packets with the same PT as configured by the
‘RFC2833PayloadType’ parameter.
Note 3: If the remote party doesn’t include the RFC 2833 DTMF relay payload type in
the SDP, the gateway uses the same PT for send and for receive.
Note 4: If TxDTMFOption is set to 0, the RFC 2833 payload type is set according to the
parameter ‘RFC2833PayloadType’ for both transmit and receive.
The RFC 2833 DTMF relay dynamic payload type.
Range: 96 to 99, 106 to 127; Default = 96
The 100, 102 to 105 range is allocated for proprietary usage.
Note 1: Cisco is using payload type 101 for RFC 2833.
Note 2: When RFC 2833 payload type (PT) negotiation is used (TxDTMFOption=4), this
payload type is used for the received DTMF packets. If negotiation isn’t used, this
payload type is used for receive and for transmit.
No [0] = INFO message isn’t sent (default).
Yes [1] = Proprietary INFO message with hook-flash is sent when hook-flash is detected
(FXS). FXO gateways generate a hook-flash signal when INFO message with hookflash is received.
Note: When either of the supplementary services (Hold, Transfer or Call Waiting) is
enabled, hook-flash is used internally, and thus the hook-flash signal isn’t sent via an
INFO message.
MediaPack SIP User’s Manual 5. Configuring the MediaPack
Table 5-4: DTMF & Dialing Parameters (continues on pages 63 to 65)
Parameter Description
Digit Mapping Rules
[DigitMapping]
Dial Tone Duration [sec]
[TimeForDialTone]
Hot Line Dial Tone Duration
[HotLineDialToneDuration]
Enable Special Digits
[IsSpecialDigits]
Default Destination Number
[DefaultNumber]
Digit map pattern. If the digit string (dialed number) has matched one of the patterns in
the digit map, the gateway stops collecting digits and starts to establish a call with the
collected number
The digit map pattern contains up to 52 options separated by a vertical bar (|).
The maximum length of the entire digit pattern is limited to 152 characters.
Available notations:
• [n-m] represents a range of numbers
• ‘.’ (single dot) represents repetition
• ‘x’ represents any single digit
• ‘T’ represents a dial timer (configured by TimeBetweenDigits parameter)
• ‘S’ should be used when a specific rule, that is part of a general rule, is to be
applied immediately. For example, if you enter the general rule x.T and the specific
rule 11x, you should append ‘S’ to the specific rule 11xS.
For example: 11xS|00T|[1-7]xxx|8xxxxxxx|#xxxxxxx|*xx|91xxxxxxxxxx|9011x.T
Time in seconds that the dial tone is played.
The default time is 16 seconds.
FXS gateway ports play the dial tone after phone is picked up; while FXO gateway ports
play the dial tone after port is seized in response to ringing.
Note 1: During play of dial tone, the gateway waits for DTMF digits.
Note 2: ‘TimeForDialTone’ is not applicable when Automatic Dialing is enabled.
Duration (in seconds) of the Hotline dial tone.
If no digits are received during the Hotline dial tone duration, the gateway initiates a call
to a preconfigured number (set in the automatic dialing table).
The valid range is 0 to 60. The default time is 16 seconds.
Applicable to FXS and FXO gateways.
Disable [0] = ‘*’ or ‘#’ terminate number collection (default).
Enable [1] = if you want to allow ‘*’ and ‘#’ to be used for telephone numbers dialed by
a user or entered for the endpoint telephone number.
Note: The # and * can always be used as first digit of a dialed number, even if you
select ‘Disable’ for this parameter.
Defines the telephone number that the gateway uses if the parameters ‘TrunkGroup_x’
or ’ChannelList‘ doesn’t include a phone number. The parameter is used as a starting
number for the list of channels comprising all hunt groups in the gateway.
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5.5.2 Configuring the Advanced Parameters
Use this submenu to configure the gateway’s advanced control protocol parameters.
5.5.2.1 General Parameters
Use this screen to configure general control protocol parameters.
¾ To configure the general parameters under Advanced Parameters, take
these 4 steps:
1. Open the ‘General Parameters’ screen (Protocol Management menu > Advanced
Parameters submenu > General Parameters option); the ‘General Parameters’ screen is
displayed.
Figure
5-7: Advanced Parameters, General Parameters Screen
MediaPack SIP User’s Manual 5. Configuring the MediaPack
2. Configure the general parameters under ‘Advanced Parameters’ according to Table 5-5.
3. Click the Submit button to save your changes.
4. To save the changes so they are available after a power fail, refer to Section
161.
Table
5-5: Advanced Parameters, General Parameters (continues on pages 67 to 70)
Parameter Description
Signaling DiffServ
[ControlIPDiffServ]
IP Security
[SecureCallsFromIP]
Filter Calls to IP
[FilterCalls2IP]
Enable Digit Delivery to IP
[EnableDigitDelivery2IP]
Enable Digit Delivery to Tel
[EnableDigitDelivery]
Defines the value of the 'DiffServ' field in the IP header for SIP messages.
The valid range is 0 to 63. The default value is 0.
No [0] = Gateway accepts all SIP calls (default).
Yes [1] = Gateway accepts SIP calls only from IP addresses defined in the Tel to IP
routing table. The gateway rejects all calls from unknown IP addresses.
For detailed information on the Tel to IP Routing table, refer to Section
83.
Note: Specifying the IP address of a Proxy server in the Tel to IP Routing table enables
the gateway to only accept calls originating in the Proxy server and rejects all other
calls.
If the filter calls to IP feature is enabled, then when a Proxy is used, the gateway first
checks the TelÆIP routing table before making a call through the Proxy. If the number is
not allowed (number isn’t listed or a Call Restriction routing rule, IP=0.0.0.0, is applied),
the call is released.
Disable [0] = Disabled (default).
Enable [1]= Enable digit delivery to IP.
The digit delivery feature enables sending of DTMF digits to the destination IP address
after the TelÆIP call was answered.
To enable this feature, modify the called number to include at least one ’p’ character.
The gateway uses the digits before the ‘p’ character in the initial INVITE message. After
the call was answered the gateway waits for the required time (# of ‘p’ * 1.5 seconds)
and then sends the rest of the DTMF digits using the method chosen (in-band, out-ofband).
Note: The called number can include several ‘p’ characters (1.5 seconds pause).
For example, the called number can be as follows: pp699, p9p300.
The digit delivery feature enables sending of DTMF digits to the gateway’s port after the
line is offhooked (FXS) or seized (FXO). For IPÆTel calls, after the line is offhooked /
seized, the MediaPack plays the DTMF digits (of the called number) towards the phone
line.
Note 1: The called number can also include the characters ‘p’ (1.5 seconds pause) and
‘d’ (detection of dial tone). If the character ‘d’ is used, it must be the first ‘digit’ in the
called number. The character ‘p’ can be used several times.
For example, the called number can be as follows: d1005, dpp699, p9p300.
To add the ‘d’ and ‘p’ digits, use the usual number manipulation rules.
Note 2: To use this feature with FXO gateways, configure the gateway to work in one
stage dialing mode.
Note 3: If the parameter ‘EnableDigitDelivery’ is enabled, it is possible to configure the
gateway to wait for dial tone per destination phone number (before or during dialing of
destination phone number), therefore the parameter ‘IsWaitForDialTone’ (that is
configurable for the entire gateway) is ignored.
Note 4: The FXS gateway sends 200 OK messages only after it finishes playing the
DTMF digits to the phone line.
5.9 on page
5.5.4.2 on page
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Table 5-5: Advanced Parameters, General Parameters (continues on pages 67 to 70)
Parameter Description
Enable DID Wink
[EnableDIDWink]
Reanswer Time
[RegretTime]
Disconnect and Answer Supervision
Enable Polarity Reversal
[EnableReversalPolarity]
Enable Current Disconnect
[EnableCurrentDisconnect]
Disconnect on Broken
Connection
[DisconnectOnBrokenConn
ection]
Broken Connection Timeout
[BrokenConnectionEventTi
meout]
Disconnect Call on Silence
Detection
[EnableSilenceDisconnect]
Silence Detection Period [sec]
[FarEndDisconnectSilenceP
eriod]
Disable [0] = DID is disabled (default).
Enable [1] = Enable DID.
If enabled, the MediaPack can be used for connection to EIA/TIA-464B DID Loop Start
lines. Both FXO (detection) and FXS (generation) are supported.
An FXO gateway dials DTMF digits after a Wink signal is detected (instead of a Dial
tone).
An FXS gateway generates the Wink signal after the detection of offhook (instead of
playing a Dial tone).
The time period (in seconds) after user hangs up the phone and before call is
disconnected (FXS). Also called regret time.
The default time is 0 seconds.
Disable [0] = Disable the polarity reversal service (default).
Enable [1] = Enable the polarity reversal service.
If the polarity reversal service is enabled, then the FXS gateway changes the line
polarity on call answer and changes it back on call release.
The FXO gateway sends a 200 OK response when polarity reversal signal is detected,
and releases a call when a second polarity reversal signal is detected.
Disable [0] = Disable the current disconnect service (default).
Enable [1] = Enable the current disconnect service.
If the current disconnect service is enabled, the FXO gateway releases a call when
current disconnect signal is detected on its port, while the FXS gateway generates a
‘Current Disconnect Pulse’ after a call is released from IP.
The current disconnect duration is determined by the parameter
‘CurrentDisconnectDuration’. The current disconnect threshold (FXO only) is determined
by the parameter ‘CurrentDisconnectDefaultThreshold’. The frequency at which the
analog line voltage is sampled is determined by the parameter
‘TimeToSampleAnalogLineVoltage’.
No [0] = Don’t release the call.
Yes [1] = Call is released if RTP packets are not received for a predefined timeout
(default).
Note 1: If enabled, the timeout is set by the parameter
‘BrokenConnectionEventTimeout’, in 100 msec resolution. The default timeout is 10
seconds: (BrokenConnectionEventTimeout =100).
Note 2: This feature is applicable only if RTP session is used without Silence
Compression. If Silence Compression is enabled, the gateway doesn’t detect that the
RTP connection is broken.
Note 3: During a call, if the source IP address (from where the RTP packets were sent)
is changed without notifying the gateway, the gateway filters these RTP packets. To
overcome this issue, set ‘DisconnectOnBrokenConnection=0’; the gateway doesn’t
detect RTP packets arriving from the original source IP address, and switches (after 300
msec) to the RTP packets arriving from the new source IP address.
The amount of time (in 100 msec units) an RTP packet isn’t received, after which a call
is disconnected.
The valid range is 1 to 1000. The default value is 100 (10 seconds).
Note 1: Applicable only if ‘DisconnectOnBrokenConnection = 1’.
Note 2: Currently this feature works only if Silence Suppression is disabled.
Yes [1] = The FXO gateway disconnect calls in which silence occurs in both (call)
directions for more than 120 seconds.
No [0] = Call is not disconnected when silence is detected (default).
The silence duration can be set by the ‘FarEndDisconnectSilencePeriod’ parameter
(default 120).
Note: To activate this feature set DSP Template to 2 or 3.
Duration of silence period (in seconds) prior to call disconnection.
The range is 10 to 28800 (8 hours). The default is 120 seconds.
Applicable to gateways, that use DSP templates 2 or 3.
MediaPack SIP User’s Manual 5. Configuring the MediaPack
Table 5-5: Advanced Parameters, General Parameters (continues on pages 67 to 70)
Parameter Description
Silence Detection Method
[FarEndDisconnectSilenceM
ethod]
CDR and Debug
CDR Server IP Address
[CDRSyslogServerIP]
CDR Report Level
[CDRReportLevel]
Debug Level
[GwDebugLevel]
Misc. Parameters
Progress Indicator to IP
[ProgressIndicator2IP]
Enable Busy Out
[EnableBusyOut]
Default Release Cause
[DefaultReleaseCause]
Delay After Reset [sec]
[GWAppDelayTime]
Silence detection method.
None [0] = Silence detection option is disabled.
Packets Count [1] = According to packet count.
Voice/Energy Detectors [2] = According to energy and voice detectors (default).
All [3] = According to packet count and energy / voice detectors.
Defines the destination IP address for CDR logs.
The default value is a null string that causes the CDR messages to be sent with all
Syslog messages.
Note: The CDR messages are sent to UDP port 514 (default Syslog port).
None [0] = Call Detail Recording (CDR) information isn’t sent to the Syslog server
(default).
End Call [1] = CDR information is sent to the Syslog server at end of each Call.
Start & End Call [2] = CDR information is sent to the Syslog server at the start and at
the end of each Call.
The CDR Syslog message complies with RFC 3161 and is identified by:
Facility = 17 (local1) and Severity = 6 (Informational).
Syslog logging level. One of the following debug levels can be selected:
0 [0] = Debug is disabled (default)
1 [1] = Flow debugging is enabled
2 [2] = Flow and device interface debugging are enabled
3 [3] = Flow, device interface and stack interface debugging are enabled
4 [4] = Flow, device interface, stack interface and session manager debugging are
enabled
5 [5] = Flow, device interface, stack interface, session manager and device interface
expanded debugging are enabled.
Note: Usually set to 5 if debug traces are needed.
No PI [0] = For IPÆTel calls, the gateway sends ‘180 Ringing’ SIP response to IP after
placing a call to phone (FXS) or to PBX (FXO).
PI = 1, PI = 8 [1], [8] = For IPÆTel calls, if ‘EnableEarlyMedia=1’, the gateway sends
‘183 session in progress’ message + SDP, immediately after a call is placed to
Phone/PBX. This is used to cut through the voice path, before remote party answers the
call, enabling the originating party to listen to network Call Progress Tones (such as
Ringback tone or other network announcements).
Not Configured [-1] = Default values are used.
The default for FXO gateways is 1; The default for FXS gateways is 0.
No [0] = ‘Busy out’ feature is not used (default).
Yes [1] = The MediaPack/FXS gateway plays a reorder tone when the phone is
offhooked and one of the following occurs:
There is a network problem.
Proxy servers do not respond and the internal routing table is not configured.
Default Release Cause (to IP) for IPÆTel calls, used when the gateway initiates a call
release, and if an explicit matching cause for this release isn’t found, a default release
cause can be configured:
The default release cause is: NO_ROUTE_TO_DESTINATION (3).
Other common values are: NO_CIRCUIT_AVAILABLE (34),
DESTINATION_OUT_OF_ORDER (27), etc.
Note: The default release cause is described in the Q.931 notation, and is translated to
corresponding SIP 40x or 50x value. For example: 404 for 3, 503 for 34 and 502 for 27.
Defines the amount of time (in seconds) the gateway’s operation is delayed after a reset
cycle.
The valid range is 0 to 600. The default value is 5 seconds.
Note: This feature helps to overcome connection problems caused by some LAN
routers or IP configuration parameters change by a DHCP Server.
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Table 5-5: Advanced Parameters, General Parameters (continues on pages 67 to 70)
Parameter Description
Max Number of Active Calls
[MaxActiveCalls]
Max Call Duration (sec)
[MaxCallDuration]
Enable LAN Watchdog
[EnableLanWatchDog]
Enable Calls Cut Through
[CutThrough]
Defines the maximum number of calls that the gateway can have active at the same
time. If the maximum number of calls is reached, new calls are not established.
The default value is max available channels (no restriction on the maximum number of
calls). The valid range is 1 to max number of channels.
Defines the maximum call duration in seconds. If this time expires, both sides of the call
are released (IP and Tel).
The valid range is 0 to 120. The default is 0 (no limitation).
Disable [0] = Disable LAN Watch-Dog (default).
Enable [1] = Enable LAN Watch-Dog.
If LAN Watch-Dog is enabled, the gateway restarts when a network failure is detected.
Enables users to receive incoming IP calls while the port is in an offhooked state.
Disable [0] = Disabled (default).
Enable [1] = Enabled.
If enabled, FXS gateways answer the call and ‘cut through’ the voice channel, if there is
no other active call on that port, even if the port is in offhooked state.
When the call is terminated (by the remote party), the gateway plays a reorder tone for
‘TimeForReorderTone’ seconds and is then ready to answer the next incoming call,
without onhooking the phone.
The waiting call is automatically answered by the gateway when the current call is
terminated (EnableCallWaiting=1).
Note: This option is applicable only to FXS gateways.
MediaPack SIP User’s Manual 5. Configuring the MediaPack
5.5.2.2 Supplementary Services
Use this screen to configure parameters that are associated with supplementary services. For
detailed information on the supplementary services, refer to Section
8.1 on page 169.
¾ To configure the supplementary services’ parameters, take these 4
steps:
1. Open the ‘Supplementary Services’ screen (Protocol Management menu > Advanced
Parameters submenu > Supplementary Services option); the ‘Supplementary Services’
screen is displayed.
Figure
5-8: Supplementary Services Parameters Screen
2. Configure the supplementary services parameters according to Table
3. Click the Submit button to save your changes, or click the Subscribe for MWI or Un-
Subscribe for MWI buttons to save your changes and to subscribe / unsubscribe to the MWI
server.
4. To save the changes so they are available after a power fail, refer to Section
161.
Version 4.6 71 June 2005
5-6.
5.9 on page
Page 72
MediaPack SIP
Table 5-6: Supplementary Services Parameters (continues on pages 72 to 74)
Parameter Description
Enable Hold
[EnableHold]
Hold Format
[HoldFormat]
Enable Transfer
[EnableTransfer]
Transfer Prefix
[xferPrefix]
Enable Call Forward
[EnableForward]
Enable Call Waiting
[EnableCallWaiting]
Number of Call Waiting
Indications
[NumberOfWaitingIndication
s]
Time Between Call Waiting
Indications
[TimeBetweenWaitingIndica
tions]
Time before Waiting Indication
[TimeBeforeWaitingIndicatio
n]
[Waiting Beep Duration]
WaitingBeepDuration
No [0] = Disable the Hold service (default).
Yes [1] = Enable the Hold service.
If the Hold service is enabled, a user can activate Hold (or Unhold) using the hook-flash.
On receiving a Hold request, the remote party is put on-hold and hears the hold tone.
Note: To use this service, the gateways at both ends must support this option.
Determines the format of the hold request.
0.0.0.0 [0] = The connection IP address in SDP is 0.0.0.0 (default).
Send Only [1] = The last attribute of the SDP contains the following ‘a=sendonly’.
No [0] = Disable the Call Transfer service (default).
Yes [1] = Enable the Call Transfer service (using REFER).
If the Transfer service is enabled, the user can activate Transfer using hook-flash
signaling. If this service is enabled, the remote party performs the call transfer.
Note 1: To use this service, the gateways at both ends must support this option.
Note 2: To use this service, set the parameter ‘Enable Hold’ to ‘Yes’.
Defined string that is added, as a prefix, to the transferred / forwarded called number,
when Refer / Redirect message is received.
Note 1: The number manipulation rules apply to the user part of the ‘REFER-TO /
Contact’ URL before it is sent in the INVITE message.
Note 2: The ‘xferprefix’ parameter can be used to apply different manipulation rules to
differentiate the transferred / forwarded number from the original dialed number.
No [0] = Disable the Call Forward service (default).
Yes [1] = Enable Call Forward service (using REFER).
For FXS gateways a Call Forward table must be defined to use the Call Forward
service.
To define the Call Forward table, refer to Section
Note: To use this service, the gateways at both ends must support this option.
No [0] = Disable the Call Waiting service (default).
Yes [1] = Enable the Call Waiting service.
If enabled, when an FXS gateway receives a call on a busy endpoint, it responds with a
182 response (and not with a 486 busy). The gateway plays a call waiting indication
signal. When hook-flash is detected, the gateway switches to the waiting call.
The gateway that initiated the waiting call plays a Call Waiting Ringback tone to the
calling party after a 182 response is received.
Note 1: The gateway’s Call Progress Tones file must include a ‘call waiting Ringback’
tone (caller side) and a ‘call waiting’ tone (called side, FXS only).
Note 2: The ‘Enable Hold’ parameter must be enabled on both the calling and the called
sides.
For information on the Call Waiting feature, refer to Section
For information on the Call Progress Tones file, refer to Section
Number of waiting indications that are played to the receiving side of the call (FXS only)
for Call Waiting.
The default value is 2.
Difference (in seconds) between call waiting indications (FXS only) for call waiting.
The default value is 10 seconds.
Defines the interval (in seconds) before a call waiting indication is played to the port that
is currently in a call (FXS only).
The valid range is 0 to 100. The default time is 0 seconds.
Duration (in msec) of waiting indications that are played to the receiving side of the call
(FXS only) for Call Waiting.
The default value is 300.
MediaPack SIP User’s Manual 5. Configuring the MediaPack
Table 5-6: Supplementary Services Parameters (continues on pages 72 to 74)
Parameter Description
Enable Caller ID
[EnableCallerID]
Caller ID Type
[CallerIDType]
MWI Parameters
Enable MWI
[EnableMWI]
MWI Analog Lamp
[MWIAnalogLamp]
MWI Display
[MWIDisplay]
Subscribe to MWI
[EnableMWISubscription]
MWI Server IP Address
[MWIServerIP]
MWI Subscribe Expiration
Time
[MWIExpirationTime]
MWI Subscribe Retry Time
[SubscribeRetryTime]
No [0] = Disable the Caller ID service (default).
Yes [1] = Enable the Caller ID service.
If the Caller ID service is enabled, then, for FXS gateways, calling number and Display
text are sent to gateway port.
For FXO gateways, the Caller ID signal is detected and is sent to IP in SIP INVITE
message (as ‘Display’ element).
For information on the Caller ID table, refer to Section
To disable/enable caller ID generation per port, refer to Section 5.5.8.4 on page 104.
Defines one of the following standards for detection (FXO) and generation (FXS) of
Caller ID and detection (FXO) of MWI (when specified) signals.
Bellcore [0] (Caller ID and MWI) (default).
ETSI [1] (Caller ID and MWI)
NTT [2]
British [4]
DTMF ETSI [16]
Denmark [17] (Caller ID and MWI)
India [18]
Brazil [19] Note 1: The Caller ID signals are generated/detected between the first and the second
rings.
Note 2: To select the Bellcore Caller ID sub standard, use the parameter
‘BellcoreCallerIDTypeOneSubStandard’. To select the ETSI Caller ID sub standard, use
the parameter ‘ETSICallerIDTypeOneSubStandard’.
Note 3: To select the Bellcore MWI sub standard, use the parameter
‘BellcoreVMWITypeOneStandard’. To select the ETSI MWI sub standard, use the
parameter ‘ETSIVMWITypeOneStandard’.
Enable MWI (message waiting indication).
Disable [0] = Disabled (default).
Enable [1] = MWI service is enabled.
This parameter is applicable only to FXS gateways.
Note: The MediaPack only supports reception of MWI.
For detailed information on MWI, refer to Section
Disable [0] = Disable (default).
Enable [1] = Enable visual Message Waiting Indication, supplies line voltage of
approximately 100 VDC to activate the phone’s lamp.
This parameter is applicable only to FXS gateways.
Disable [0] = MWI information isn’t sent to display (default).
Enable [1] = MWI information is sent to display.
If enabled, the gateway generates an MWI FSK message that is displayed on the MWI
display.
This parameter is applicable only to FXS gateways.
Disable [0] = Disable MWI subscription (default).
Enable [1] = Enable subscription to MWI (to MWIServerIP address).
Note: Use the parameter ‘SubscriptionMode’ (described in Table 5-27 on page 111) to
determine whether the gateway subscribes separately per endpoint of for the entire
gateway.
MWI server IP address. If provided, the gateway subscribes to this IP address.
Can be configured as a numerical IP address or as a domain name. If not configured,
the Proxy IP address is used instead.
MWI subscription expiration time in seconds.
The default is 7200 seconds. The range is 10 to 72000.
Subscription retry time in seconds.
The default is 120 seconds. The range is 10 to 7200.
5.5.8.3 on page 103.
8.1.6 on page 171.
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MediaPack SIP
Table 5-6: Supplementary Services Parameters (continues on pages 72 to 74)
Parameter Description
Stutter Tone Duration
[StutterToneDuration]
Duration (in msec) of the played stutter dial tone that indicates waiting message(s).
The default is 2000 (2 seconds). The range is 1000 to 60000.
The Stutter tone is played (instead of a regular Dial tone) when a MWI is received. The
tone is composed of a ‘Confirmation tone’ that is played for ‘StutterToneDuration’
followed by a ‘Stutter tone’. Both tones are defined in the CPT file.
Note: This parameter is applicable only to FXS gateways.
For detailed information on Message Waiting Indication (MWI), refer to Section
page 171.
5.5.2.3 Keypad Features
The Keypad Features screen (applicable only to FXS gateways) enables you to activate /
deactivate the following features directly from the connected telephone’s keypad:
• Call Forward (refer to Section
• Caller ID Restriction (refer to Section
• Hotline (refer to Section
¾ To configure the keypad features, take these 4 steps:
1. Open the ‘Keypad Features’ screen (Protocol Management menu > Advanced
Parameters submenu > Keypad Features option); the ‘Keypad Features’ screen is
displayed.
8.1.6 on
5.5.8.4 on page 104).
5.5.8.3 on page 103).
5.5.8.2 on page 102).
Figure
5-9: Keypad Features Screen
2. Configure the Keypad Features according to Table
MediaPack SIP User’s Manual 5. Configuring the MediaPack
4. To save the changes so they are available after a power fail, refer to Section 5.9 on page
161.
Note: The method used by the gateway to collect dialed numbers is identical to the
method used during a regular call (i.e., max digits, interdigit timeout, digit
Parameter Description
Forward
map, etc.).
Table 5-7: Keypad Features Parameters
Unconditional
[KeyCFUnCond]
No Answer
[KeyCFNoAnswer]
On Busy
[KeyCFBusy]
On Busy or No Answer
[KeyCFBusyOrNoAnswer]
Do Not Disturb
[KeyCFDoNotDisturb]
To activate the required forward method from the telephone:
Keypad sequence that activates the immediate forward option.
Keypad sequence that activates the forward on no answer option.
Keypad sequence that activates the forward on busy option.
Keypad sequence that activates the forward on ‘busy or no answer’ option.
Keypad sequence that activates the Do Not Disturb option.
• Dial the preconfigured sequence number on the keypad; a dial tone is heard.
• Dial the telephone number to which the call is forwarded (terminate the number with #); a confirmation tone is
heard.
Deactivate
[KeyCFDeact]
Caller ID Restriction
Activate
[KeyCLIR]
Deactivate
[KeyCLIRDeact]
Hotline
Keypad sequence that deactivates any of the forward options.
After the sequence is pressed a confirmation tone is heard.
Keypad sequence that activates the restricted Caller ID option.
After the sequence is pressed a confirmation tone is heard.
Keypad sequence that deactivates the restricted Caller ID option.
After the sequence is pressed a confirmation tone is heard.
Activate
[KeyHotLine]
Keypad sequence that activates the delayed hotline option.
To activate the delayed hotline option from the telephone:
• Dial the preconfigured sequence number on the keypad; a dial tone is heard.
• Dial the telephone number to which the phone automatically dials after a
configurable delay (terminate the number with #); a confirmation tone is heard.
Deactivate
[KeyHotLineDeact]
Keypad sequence that deactivates the delayed hotline option.
After the sequence is pressed a confirmation tone is heard.
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5.5.3 Configuring the Number Manipulation Tables
The VoIP gateway provides four Number Manipulation tables for incoming and outgoing calls.
These tables are used to modify the destination and source telephone numbers so that the calls
can be routed correctly.
The Manipulation Tables are:
• Destination Phone Number Manipulation Table for IPÆTel calls
• Destination Phone Number Manipulation Table for TelÆIP call
• Source Phone Number Manipulation Table for IPÆTel calls
• Source Phone Number Manipulation Table for TelÆIP calls
Note: Number manipulation can occur either before or after a routing decision is
made. For example, you can route a call to a specific hunt group according
to its original number, and then you can remove / add a prefix to that number
before it is routed. To control when number manipulation is done, set the IP
Possible uses for number manipulation can be as follows:
to Tel Routing Mode (described in Table 5-12) and the Tel to IP Routing
Mode (described in Table 5-11) parameters.
•To strip/add dialing plan digits from/to the number. For example, a user could dial 9 in front
of each number in order to indicate an external line. This number (9) can be removed here
before the call is setup.
•Allow / disallow Caller ID information to be sent according to destination / source prefixes.
For detailed information on Caller ID, refer to Section
5.5.8.3 on page 103.
¾ To configure the Number Manipulation tables, take these 5 steps:
1. Open the Number Manipulation screen you want to configure (Protocol Management menu
> Manipulation Tables submenu); the relevant Manipulation table screen is displayed.
Figure
Figure
5-10 shows the ‘Source Phone Number Manipulation Table for TelÆIP calls’.
5-10: Source Phone Number Manipulation Table for TelÆIP calls
2. In the ‘Table Index’ drop-down list, select the range of entries that you want to edit (up to 20
entries can be configured for Source Number Manipulation and 50 entries for Destination
Number Manipulation).
3. Configure the Number Manipulation table according to Table
MediaPack SIP User’s Manual 5. Configuring the MediaPack
4. Click the Submit button to save your changes.
5. To save the changes so they are available after a power fail, refer to Section
5.9 on page
161.
Table 5-8: Number Manipulation Parameters
Parameter Description
Destination Prefix Each entry in the Destination Prefix fields represents a destination telephone number
prefix. An asterisk (*) represents any number.
Source Prefix Each entry in the Source Prefix fields represents a source telephone number prefix.
An asterisk (*) represents any number.
Source IP Each entry in the Source IP fields represents the source IP address of the call
(obtained from the Contact header in the INVITE message).
This column only applies to the ‘Destination Phone Number Manipulation Table for
IP to Tel’.
Note: The source IP address can include the ‘x’ wildcard to represent single
For example: 10.8.8.xx represents all the addresses between 10.8.8.10 to 10.8.8.99.
The manipulation rules are applied to any incoming call whose:
digits.
• Destination number prefix matches the prefix defined in the ‘Destination Number’ field.
• Source number prefix matches the prefix defined in the ‘Source Prefix’ field.
• Source IP address matches the IP address defined in the ‘Source IP’ field (if applicable).
Note that number manipulation can be performed using a combination of each of the above criteria, or using each
criterion independently.
Note: For available notations that represent multiple numbers, refer to Section 5.5.3.1 on page 79.
Num of stripped digits
•Enter the number of digits that you want to remove from the left of the telephone
number prefix. For example, if you enter 3 and the phone number is 5551234,
the new phone number is 1234.
•Enter the number of digits (in brackets) that you want to remove from the right of
the telephone number prefix.
Note: A combination of the two options is allowed (e.g., 2(3)).
Prefix / Suffix to add
•Prefix - Enter the number / string you want to add to the front of the phone
number. For example, if you enter 9 and the phone number is 1234, the new
number is 91234.
•Suffix - Enter the number / string (in brackets) you want to add to the end of the
phone number. For example, if you enter (00) and the phone number is 1234,
the new number is 123400.
Note: You can enter a prefix and a suffix in the same field (e.g., 9(00)).
Number of digits to leave Enter the number of digits that you want to leave from the right.
Note: The manipulation rules are executed in the following order:
1. Num of stripped digits
2. Number of digits to leave
3. Prefix / suffix to add
5-10 on the previous page exemplifies the use of these manipulation rules in the ‘Source Phone Number
Figure
Manipulation Table for TelÆIP Calls’:
•When destination number equals 035000 and source number equals 20155, the source number is changed to
97220155.
• When source number equals 1001876, it is changed to 587623.
• Source number 1234510012001 is changed to 20018.
• Source number 3122 is changed to 2312.
Presentation Select ‘Allowed’ to send Caller ID information when a call is made using these
destination / source prefixes.
Select ‘Restricted’ if you want to restrict Caller ID information for these prefixes.
When set to ‘Not Configured’, the privacy is determined according to the Caller ID
table (refer to Section
Note: If ‘Presentation’ is set to ‘Restricted’ and ‘Asserted Identity Mode’ is set to ‘PAsserted’, the From header in INVITE message is: From: ‘anonymous’ <sip:
anonymous@anonymous.invalid> and ‘privacy: id’ header is included in the INVITE
message.
5.5.8.3 on page 103).
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Table 5-9: Number Manipulation ini File Parameters (continues on pages 78 to 79)
Parameter Description
NumberMapTel2IP
NumberMapIP2Tel
Manipulates the destination number for Tel to IP calls.
NumberMapTel2IP = a,b,c,d,e,f,g
a = Destination number prefix
b = Number of stripped digits from the left, or (if brackets are used) from the right.
A combination of both options is allowed.
c = String to add as prefix, or (if brackets are used) as suffix. A combination of
both options is allowed.
d = Number of remaining digits from the right
e = Number Plan used in RPID header
f = Number Type used in RPID header
g = Source number prefix
The ‘b’ to ‘f’ manipulation rules are applied if the called and calling numbers match
the ‘a’ and ‘g’ conditions.
The manipulation rules are executed in the following order: ‘b’, ‘d’ and ‘c’.
Parameters can be skipped by using the sign ‘$$’, for example:
NumberMapTel2IP=01,2,972,$$,0,0,$$
NumberMaPTel2IP=03,(2),667,$$,0,0,22
Note: Number Plan & Type can optionally be used in Remote Party ID (RPID)
header by using the ‘EnableRPIHeader’ and ‘AddTON2RPI’ parameters.
Manipulate the destination number for IP to Tel calls.
NumberMapIP2Tel = a,b,c,d,e,f,g,h,i
a = Destination number prefix.
b = Number of stripped digits from the left, or (if brackets are used) from the right.
A combination of both options is allowed.
c = String to add as prefix, or (if brackets are used) as suffix. A combination of
both options is allowed.
d = Number of remaining digits from the right.
e = Not applicable, set to $$.
f = Not applicable, set to $$.
g = Source number prefix.
h = Not applicable, set to $$.
i = Source IP address (obtained from the Contact header in the INVITE
message).
The ‘b’ to ‘d’ manipulation rules are applied if the called and calling numbers match
the ‘a’, ‘g’ and ‘i’ conditions.
The manipulation rules are executed in the following order: ‘b’, ‘d’ and ‘c’.
Parameters can be skipped by using the sign ‘$$’, for example:
NumberMapIP2Tel =01,2,972,$$,$$,$$,034,$$,10.13.77.8
NumberMapIP2Tel =03,(2),667,$$,$$,$$,22
Note: The Source IP address can include the ‘x’ wildcard to represent single
For example: 10.8.8.xx represents all the addresses between 10.8.8.10 to 10.8.8.99.
MediaPack SIP User’s Manual 5. Configuring the MediaPack
Parameter Description
SourceNumberMapTel2IP
SourceNumberMapIP2Tel
SourceNumberMapTel2IP = a,b,c,d,e,f,g,h
a = Source number prefix
b = Number of stripped digits from the left, or (if in brackets are used) from right. A
combination of both options is allowed.
c = String to add as prefix, or (if in brackets are used) as suffix. A combination of
both options is allowed.
d = Number of remaining digits from the right
e = Number Plan used in RPID header
f = Number Type used in RPID header
g = Destination number prefix
h = Calling number presentation (0 to allow presentation, 1 to restrict
presentation)
The ‘b’ to ‘f’ and ‘h’ manipulation rules are applied if the called and calling numbers
match the ‘a’ and ‘g’ conditions.
The manipulation rules are executed in the following order: ‘b’, ‘d’ and ‘c’.
Parameters can be skipped by using the sign ‘$$’, for example:
SourceNumberMapTel2IP=01,2,972,$$,0,0,$$,1
SourceNumberMapTel2IP=03,(2),667,$$,0,0,22
Note 1: ‘Presentation’ is set to ‘Restricted’ only if ‘Asserted Identity Mode’ is set to
‘P-Asserted’.
Note 2: Number Plan & Type can optionally be used in Remote Party ID (RPID)
header by using the ‘EnableRPIHeader’ and ‘AddTON2RPI’ parameters.
Manipulate the destination number for IP to Tel calls.
NumberMapIP2Tel = a,b,c,d,e,f,g
a = Source number prefix
b = Number of stripped digits from the left, or (if brackets are used) from the right.
A combination of both options is allowed.
c = String to add as prefix, or (if brackets are used) as suffix. A combination of
both options is allowed.
d = Number of remaining digits from the right
e = Not in use, should be set to $$
f = Not in use, should be set to $$
g = Destination number prefix
The ‘b’ to ‘d’ manipulation rules are applied if the called and calling numbers match
the ‘a’ and ‘g’ conditions.
The manipulation rules are executed in the following order: ‘b’, ‘d’ and ‘c’.
Parameters can be skipped by using the sign ‘$$’, for example:
NumberMapIP2Tel =01,2,972,$$,$$,$$,034
NumberMapIP2Tel =03,(2),667,$$,$$,$$,22
5.5.3.1 Dialing Plan Notation
The dialing plan notation applies, in addition to the four Manipulation tables, also to TelÆIP
Routing table and to IPÆHunt Group Routing table.
When entering a number in the destination and source ‘Prefix’ columns, you can create an entry
that represents multiple numbers using the following notation:
• [n-m] represents a range of numbers
• [n,m] represents multiple numbers. Note that this notation only supports single digit numbers.
• x represents any single digit
• # (that terminates the number) represents the end of a number
• A single asterisk (*) represents any number
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For example:
• [5551200-5551300]# represents all numbers from 5551200 to 5551300
• [2,3,4]xxx# represents four-digit numbers that start with 2, 3 or 4
• 54324 represents any number that starts with 54324
• 54324xx# represents a 7 digit number that starts with 54324
• 123[100-200]# represents all numbers from 123100 to 123200.
The VoIP gateway matches the rules starting at the top of the table. For this reason, enter more
specific rules above more generic rules. For example, if you enter 551 in entry 1 and 55 in entry
2, the VoIP gateway applies rule 1 to numbers that starts with 551 and applies rule 2 to numbers
that start with 550, 552, 553, 554, 555, 556, 557, 558 and 559. However if you enter 55 in entry 1
and 551 in entry 2, the VoIP gateway applies rule 1 to all numbers that start with 55 including
numbers that start with 551.
MediaPack SIP User’s Manual 5. Configuring the MediaPack
5.5.4 Configuring the Routing Tables
Use this submenu to configure the gateway’s IPÆTel and TelÆIP routing tables and their
associated parameters.
5.5.4.1 General Parameters
Use this screen to configure the gateway’s IPÆTel and TelÆIP routing parameters.
¾ To configure the general parameters under Routing Tables, take these 4
steps:
1. Open the ‘General Parameters’ screen (Protocol Management menu > Routing Tables
submenu > General option); the ‘General Parameters’ screen is displayed.
Figure
5-11: Routing Tables, General Parameters Screen
2. Configure the general parameters under ‘Routing Tables’ according to Table
3. Click the Submit button to save your changes.
4. To save the changes so they are available after a power fail, refer to Section
161.
Table
5-10: Routing Tables, General Parameters (continues on pages 81 to 82)
5-10.
5.9 on page
Parameter Description
Add Hunt Group ID as Prefix
[AddTrunkGroupAsPrefix]
Add Port Number as Prefix
[AddPortAsPrefix]
No [0] = Don’t add hunt group ID as prefix (default).
Yes [1] = Add hunt group ID as prefix to called number.
If enabled, then the hunt group ID is added as a prefix to the destination phone number
for TelÆIP calls.
Note 1: This option can be used to define various routing rules.
Note 2: To use this feature you must configure the hunt group IDs.
No [0] = Disable the add port as prefix service (default).
Yes [1] = Enable the add port as prefix service.
If enabled, then the gateway’s port number (single digit in the range 1 to 8 for 8-port
gateways, two digits in the range 01 to 24 in MP-124) is added as a prefix to the
destination phone number for TelÆIP calls.
Note: This option can be used to define various routing rules.
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Table 5-10: Routing Tables, General Parameters (continues on pages 81 to 82)
Parameter Description
IP to Tel Remove Routing
Table Prefix
[RemovePrefix]
Enable Alt Routing Tel to IP
[AltRoutingTel2IPEnable]
Alt Routing Tel to IP Mode
[AltRoutingTel2IPMode]
Max Allowed Packet Loss for
Alt Routing [%]
[IPConnQoSMaxAllowedPL]
Max Allowed Delay for Alt
Routing [msec]
[IPConnQoSMaxAllowedDel
ay]
No [0] = Don't remove prefix (default)
Yes [1] = Remove the prefix (defined in the IP to Hunt Group Routing table) from a
telephone number for an IPÆTel call, before forwarding it to Tel.
For example:
To route an incoming IPÆTel Call with destination number 21100, the IP to Hunt Group
Routing table is scanned for a matching prefix. If such prefix is found, 21 for instance,
then before the call is routed to the corresponding hunt group the prefix (21) is removed
from the original number, so that only 100 is left.
Note 1: Applicable only if number manipulation is performed after call routing for IPÆTel
calls (refer to ‘IP to Tel Routing Mode’ parameter).
Note 2: Similar operation (of removing the prefix) is also achieved by using the usual
number manipulation rules.
No [0] = Disable the Alternative Routing feature (default).
Yes [1] = Enable the Alternative Routing feature.
Status Only [2] = The Alternative Routing feature is disabled. A read only information
on the quality of service of the destination IP addresses is provided.
For information on the Alternative Routing feature, refer to Section
None [0] = Alternative routing is not used.
Conn [1] = Alternative routing is performed if ping to initial destination failed.
QoS [2] = Alternative routing is performed if poor quality of service was detected.
Both [3] = Alternative routing is performed if, either ping to initial destination failed,
or poor quality of service was detected, or DNS host name is not resolved (default).
Note: QoS (Quality of Service) is quantified according to delay and packet loss,
calculated according to previous calls. QoS statistics are reset if no new data is received
for two minutes.
For information on the Alternative Routing feature, refer to
Packet loss percentage at which the IP connection is considered a failure.
The range is 1% to 20%. The default value is 20%.
Transmission delay (in msec) at which the IP connection is considered a failure.
The range is 100 to 1000. The default value is 250 msec.
MediaPack SIP User’s Manual 5. Configuring the MediaPack
5.5.4.2 Tel to IP Routing Table
The Tel to IP Routing Table is used to route incoming Tel calls to IP addresses. This routing table
associates a called / calling telephone number’s prefixes with a destination IP address or with an
FQDN (Fully Qualified Domain Name). When a call is routed through the VoIP gateway (Proxy
isn’t used), the called and calling numbers are compared to the list of prefixes on the IP Routing
Table (up to 50 prefixes can be configured); Calls that match these prefixes are sent to the
corresponding IP address. If the number dialed does not match these prefixes, the call is not
made.
When using a Proxy server, you do not need to configure the Tel to IP Routing Table. However, if
you want to use fallback routing when communication with Proxy servers is lost, or to use the
‘Filter Calls to IP’ and ‘IP Security’ features, or to obtain different SIP URI host names (per called
number) or to assign IP profiles, you need to configure the IP Routing Table.
Note that for the Tel to IP Routing table to take precedence over a Proxy for routing calls, set the
parameter ‘PreferRouteTable’ to 1. The gateway checks the 'Destination IP Address' field in the
'Tel to IP Routing' table for a match with the outgoing call. Only if a match is not found, a Proxy is
used.
Possible uses for Tel to IP Routing can be as follows:
• Can fallback to internal routing table if there is no communication with the Proxy servers.
• Call Restriction – (when Proxy isn’t used), reject all outgoing TelÆIP calls that are
associated with the destination IP address: 0.0.0.0.
•IP Security – When the IP Security feature is enabled (SecureCallFromIP = 1), the VoIP
gateway accepts only those IPÆTel calls with a source IP address identical to one of the IP
addresses entered in the Tel to IP Routing Table.
•Filter Calls to IP – When a Proxy is used, the gateway checks the TelÆIP routing table
before a telephone number is routed to the Proxy. If the number is not allowed (number isn’t
listed or a Call Restriction routing rule was applied), the call is released.
•Always Use Routing Table – When this feature is enabled (AlwaysUseRouteTable = 1), even
if a Proxy server is used, the SIP URI host name in the sent INVITE message is obtained
from this table. Using this feature users are able to assign a different SIP URI host name for
different called and/or calling numbers.
• Assign Profiles to destination address (also when a Proxy is used).
• Alternative Routing – (When Proxy isn’t used) an alternative IP destination for telephone
number prefixes is available. To associate an alternative IP address to called telephone
number prefix, assign it with an additional entry (with a different IP address), or use an
FQDN that resolves to two IP addresses. Call is sent to the alternative destination when one
of the following occurs:
¾No ping to the initial destination is available, or when poor QoS (delay or packet loss,
calculated according to previous calls) is detected, or when a DNS host name is not
resolved. For detailed information on Alternative Routing, refer to Section 8.7 on page
179.
¾When a release reason that is defined in the ‘Reasons for Alternative Tel to IP Routing’
table is received. For detailed information on the ‘Reasons for Alternative Routing
Tables’, refer to Section 5.5.4.5 on page 89.
Alternative routing (using this table) is commonly implemented when there is no response to
an INVITE message (after INVITE retransmissions). The gateway then issues an internal
408 ‘No Response’ implicit release reason. If this reason is included in the ‘Reasons for
Alternative Routing’ table, the gateway immediately initiates a call to the redundant
destination using the next matched entry in the ‘Tel to IP Routing’ table. Note that if a domain
name in this table is resolved to two IP addresses, the timeout for INVITE retransmissions
can be reduced by using the parameter ‘Number of RTX Before Hotswap’.
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Tip: Tel to IP routing can be performed either before or after applying the number
manipulation rules. To control when number manipulation is done, set the
¾ To configure the Tel to IP Routing table, take these 6 steps:
1. Open the ‘Tel to IP Routing’ screen (Protocol Management menu > Routing Tables
submenu > Tel to IP Routing option); the ‘Tel to IP Routing’ screen is displayed (shown in
Figure
5-12).
Tel to IP Routing Mode parameter (described in Table 5-11).
2. In the ‘Tel to IP Routing Mode’ field, select the Tel to IP routing mode (refer to Table
3. In the ‘Routing Index’ drop-down list, select the range of entries that you want to edit.
4. Configure the Tel to IP Routing table according to Table
5. Click the Submit button to save your changes.
6. To save the changes so they are available after a power fail, refer to Section
161.
Figure
5-12: Tel to IP Routing Table Screen
5-11.
5.9 on page
5-11).
5-11: Tel to IP Routing Table
Table
Parameter Description
Tel to IP Routing Mode
[RouteModeTel2IP]
Destination Phone Prefix Each entry in the Destination Phone Prefix fields represents a called telephone number
Source Phone Prefix Each entry in the Source Phone Prefix fields represents a calling telephone number
Any telephone number whose destination number matches the prefix defined in the ‘Destination Phone Prefix’ field and
its source number matches the prefix defined in the adjacent ‘Source Phone Prefix‘ field, is sent to the IP address
entered in the ‘IP Address’ field.
Note that Tel to IP routing can be performed according to a combination of source and destination phone prefixes, or
using each independently.
Note 1: An additional entry of the same prefixes can be assigned to enable alternative routing.
Note 2: For available notations that represent multiple numbers, refer to Section 5.5.3.1 on page 79.
Route calls before manipulation [0] = TelÆIP calls are routed before the number
manipulation rules are applied (default).
Route calls after manipulation [1] = TelÆIP calls are routed after the number
manipulation rules are applied.
Note: Not applicable if Proxy routing is used.
prefix. The prefix can be 1 to 19 digits long. An asterisk (*) represents all numbers.
prefix. The prefix can be 1 to 19 digits long. An asterisk (*) represents all numbers.
MediaPack SIP User’s Manual 5. Configuring the MediaPack
Table 5-11: Tel to IP Routing Table
Parameter Description
Destination IP Address In each of the IP Address fields, enter the IP address (and optionally port number) that is
assigned to these prefixes. Domain names, such as domain.com, can be used instead of
IP addresses.
For example: <IP Address>:<Port>
To discard outgoing IP calls, enter 0.0.0.0 in this field.
Note: When using domain names, you must enter a DNS server IP address, or
alternatively define these names in the ‘Internal DNS Table’.
Profile ID Enter the number of the IP profile that is assigned to the destination IP address defined in
the ‘Destination IP Address’ field.
Status A read only field representing the quality of service of the destination IP address.
N/A = Alternative Routing feature is disabled.
OK = IP route is available
Ping Error = No ping to IP destination, route is not available
QoS Low = Bad QoS of IP destination, route is not available
DNS Error = No DNS resolution (only when domain name is used instead of an IP
address).
Parameter Name in ini File Parameter Format
Prefix
Prefix = <Destination Phone Prefix>,<Destination IP Address>,<Source Phone
Prefix>,<Profile ID>
For example:
Prefix = 20,10.2.10.2,202,1
Prefix = 10[340-451]xxx#,10.2.10.6,*,1
Prefix = *,gateway.domain.com,*
Note 1: <destination / source phone prefix> can be single number or a range of numbers.
For available notations, refer to Section 5.5.3.1 on page 79.
Note 2: This parameter can appear up to 50 times.
Note 3: Parameters can be skipped by using the sign ‘$$’, for example:
Prefix = $$,10.2.10.2,202,1
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5.5.4.3 IP to Hunt Group Routing
The IP to Hunt Group Routing Table is used to route incoming IP calls to groups of channels
called hunt groups. Calls are assigned to hunt groups according to any combination of the
following three options (or using each independently):
• Destination phone prefix
• Source phone prefix
• Source IP address
The call is then sent to the VoIP gateway channels assigned to that hunt group. The specific
channel, within a hunt group, that is assigned to accept the call is determined according to the
hunt group’s channel selection mode which is defined in the Hunt Group Settings table (Section
5.5.7 on page 99) or according to the global parameter ‘ChannelSelectMode’ (refer to Table 5-5
on page 67). Hunt groups can be used on both FXO and FXS gateways; however, usually they
are used with FXO gateways.
Note: When a release reason that is defined in the ‘Reasons for Alternative IP to Tel Routing’
table is received for a specific IPÆTel call, an alternative hunt group for that call is available. To
associate an alternative hunt group to an incoming IP call, assign it with an additional entry in the
‘IP to Hunt Group Routing’ table (repeat the same routing rules with a different hunt group ID).
For detailed information on the ‘Reasons for Alternative Routing Tables’, refer to Section
on page 89.
5.5.4.5
To use hunt groups you must also do the following.
•You must assign a hunt group ID to the VoIP gateway channels on the Endpoint Phone
Number screen. For information on how to assign a hunt group ID to a channel, refer to
Section
•You can configure the Hunt Group Settings table to determine the method in which new calls
are assigned to channels within the hunt groups (a different method for each hunt group can
be configured). For information on how to enable this option, refer to Section
99. If a Channel Select Mode for a specific hunt group isn’t specified, then the global
‘Channel Select Mode’ parameter (defined in ‘General Parameters’ screen under ‘Advanced
Parameters’) applies.
5.5.6 on page 97.
5.5.7 on page
¾ To configure the IP to Hunt Group Routing table, take these 6 steps:
1. Open the ‘IP to Hunt Group Routing’ screen (Protocol Management menu > Routing
Tables submenu > IP to Hunt Group Routing option); the ‘IP to Hunt Group Routing’ table
MediaPack SIP User’s Manual 5. Configuring the MediaPack
2. In the ‘IP to Tel Routing Mode’ field, select the IP to Tel routing mode (refer to Table 5-12).
3. In the ‘Routing Index’ drop-down list, select the range of entries that you want to edit (up to
24 entries can be configured).
4. Configure the IP to Hunt Group Routing table according to Table
5-12.
5. Click the Submit button to save your changes.
6. To save the changes so they are available after a power fail, refer to Section
5.9 on page
161.
Table 5-12: IP to Hunt Group Routing Table
Parameter Description
IP to Tel Routing Mode
[RouteModeIP2Tel]
Destination Phone Prefix Each entry in the Destination Phone Prefix fields represents a called telephone number
Source Phone Prefix Each entry in the Source Phone Prefix fields represents a calling telephone number
Source IP Address Each entry in the Source IP Address fields represents the source IP address of an
Any SIP incoming call whose destination number matches the prefix defined in the ‘Destination Phone Prefix’ field and
its source number matches the prefix defined in the adjacent ‘Source Phone Prefix‘ field and its source IP address
matches the address defined in the ‘Source IP Address’ field, is assigned to the hunt group entered in the field to the
right of these fields.
Note that IP to hunt group routing can be performed according to any combination of source / destination phone
prefixes and source IP address, or using each independently.
Note: For available notations that represent multiple numbers (used in the prefix columns), refer to Section
page 79.
Hunt Group ID In each of the Hunt Group ID fields, enter the hunt group ID to which calls that match
Profile ID Enter the number of the IP profile that is assigned to the routing rule.
Route calls before manipulation [0] = IPÆTel calls are routed before the number
manipulation rules are applied (default).
Route calls after manipulation [1] = IPÆTel calls are routed after the number
manipulation rules are applied.
prefix. The prefix can be 1 to 49 digits long. An asterisk (*) represents all numbers.
prefix. The prefix can be 1 to 49 digits long. An asterisk (*) represents all numbers.
IPÆTel call (obtained from the Contact header in the INVITE message).
Note: The source IP address can include the ‘x’ wildcard to represent single
example: 10.8.8.xx represents all the addresses between 10.8.8.10 to 10.8.8.99.
digits. For
5.5.3.1 on
these prefixes are assigned.
Parameter Name in ini
File
PSTNPrefix
Parameter Format
PSTNPrefix = a,b,c,d,e
a = Destination Number Prefix
b = Hunt Group ID
c = Source Number Prefix
d = Source IP address (obtained from the Contact header in the INVITE message)
e = IP Profile ID
Selection of hunt groups (for IP to Tel calls) is according to destination number, source
number and source IP address.
Note 1: To support the ‘in call alternative routing’ feature, users can use two entries that
support the same call, but assigned it with a different hunt groups. The second entree
functions as an alternative selection if the first rule fails as a result of one of the release
reasons listed in the AltRouteCauseIP2Tel table.
Note 2: An optional IP ProfileID (1 to 4) can be applied to each routing rule.
Note 3: The Source IP Address can include the ‘x’ wildcard to represent single digits.
For example: 10.8.8.xx represents all IP addresses between 10.8.8.10 to 10.8.8.99.
Note 4: For available notations that represent multiple numbers, refer to Section 5.5.3.1
on page 79. Note 5: This parameter can appear up to 24 times.
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5.5.4.4 Internal DNS Table
The internal DNS table, similar to a DNS resolution, translates hostnames into IP addresses. This
table is used when hostname translation is required (e.g., ‘Tel to IP Routing’ table). Two different
IP addresses can be assigned to the same hostname. If the hostname isn’t found in this table, the
gateway communicates with an external DNS server.
Assigning two IP addresses to hostname can be used for alternative routing (using the ‘Tel to IP
Routing’ table).
¾ To configure the internal DNS table, take these 7 steps:
1. Open the ‘Internal DNS Table’ screen (Protocol Management menu > Routing Tables
submenu > Internal DNS Table option); the ‘Internal DNS Table’ screen is displayed.
Figure
2. In the ‘DNS Name’ field, enter the hostname to be translated. You can enter a string up to 31
characters long.
3. In the ‘First IP Address’ field, enter the first IP address that the hostname is translated to.
4. In the ‘Second IP Address’ field, enter the second IP address that the hostname is translated
to.
5. Repeat steps 2 to 4, for each Internal DNS Table entry.
6. Click the Submit button to save your changes.
7. To save the changes so they are available after a power fail, refer to Section
DNS2IP = <Hostname>, <first IP address>, <second IP address>
For example:
DNS2IP = Domainname.com, 10.8.21.4, 10.13.2.95
Note: This parameter can appear up to 10 times.
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MediaPack SIP User’s Manual 5. Configuring the MediaPack
5.5.4.5 Reasons for Alternative Routing
The Reasons for Alternative Routing screen includes two tables (TelÆIP and IPÆTel). Each table
enables you to define up to 4 different release reasons. If a call is released as a result of one of
these reasons, the gateway tries to find an alternative route to that call. The release reason for
IPÆTel calls is provided in Q.931 notation. The release reason for TelÆIP calls is provided in SIP
4xx, 5xx and 6xx response codes. For TelÆIP calls an alternative IP address, for IPÆTel calls an
alternative hunt group.
Refer to ‘Tel to IP Routing’ on page 83 for information on defining an alternative IP address. Refer
to the ‘IP to Hunt Group Routing’ on page 86 for information on defining an alternative hunt group.
You can use this table for example:
For TelÆIP calls, when there is no response to an INVITE message (after INVITE
retransmissions), and the gateway then issues an internal 408 ‘No Response’ implicit release
reason.
For IPÆTel calls, when the destination is busy, and release reason #17 is issued or for other call
releases that issue the default release reason (#3). Refer to ‘DefaultReleaseCause’ in Table
Note: The reasons for alternative routing option for TelÆIP calls only applies when Proxy isn’t
used.
5-5.
¾ To configure the reasons for alternative routing, take these 5 steps:
1. Open the ‘Reasons for Alternative Routing’ screen (Protocol Management menu > Routing
Tables submenu > Reasons for Alternative Routing option); the ‘Reasons for Alternative
Routing’ screen is displayed.
Figure
5-15: Reasons for Alternative Routing Screen
2. In the ‘IP to Tel Reasons’ table, from the drop-down list select up to 4 different call failure
reasons that invoke an alternative IP to Tel routing.
3. In the ‘Tel to IP Reasons’ table, from the drop-down list select up to 4 different call failure
reasons that invoke an alternative Tel to IP routing.
4. Click the Submit button to save your changes.
5. To save the changes so they are available after a power fail, refer to Section
161.
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Table 5-14: Reasons for Alternative Routing ini File Parameter
Parameter Name in ini File Parameter Format
AltRouteCauseTel2IP
AltRouteCauseIP2Tel
AltRouteCauseTel2IP = <SIP Call failure reason from IP>
For example:
AltRouteCauseTel2IP = 408 (Response timeout).
AltRouteCauseTel2IP = 486 (User is busy).
Note: This parameter can appear up to 4 times.
AltRouteCauseIP2Tel = <Call failure reason from Tel>
For example:
AltRouteCauseIP2Tel = 3 (No route to destination).
AltRouteCauseIP2Tel = 17 (Busy here).
MediaPack SIP User’s Manual 5. Configuring the MediaPack
5.5.5 Configuring the Profile Definitions
Utilizing the Profiles feature, the MediaPack provides high-level adaptation when connected to a
variety of equipment (from both Tel and IP sides) and protocols, each of which require a different
system behavior. Using Profiles, users can assign different Profiles (behavior) on a per-call basis,
using the Tel to IP and IP to Hunt Group Routing tables, or associate different Profiles to the
gateway’s endpoint(s). The Profiles contain parameters such as Coders, T.38 Relay, Voice and
DTMF Gains, Silence Suppression, Echo Canceler, RTP DiffServ, Current Disconnect and more.
The Profiles feature allows users to tune these parameters or turn them on or off, per source or
destination routing and/or the specific gateway or its ports. For example, specific ports can be
designated to have a profile which always uses G.711.
Each call can be associated with one or two Profiles, Tel Profile and (or) IP Profile. If both IP and
Tel profiles apply to the same call, the coders and other common parameters of the preferred
Profile (determined by the Preference option) are applied to that call. If the Preference of the Tel
and IP Profiles is identical, the Tel Profile parameters are applied.
Note: The default values of the parameters in the Tel and IP Profiles are identical
to the Web/ini file parameter values. If a value of a parameter is changed in
the Web/ini file, it is automatically updated in the Profiles correspondingly.
After any parameter in the Profile is modified by the user, modifications to
parameters in the Web/ini file no longer impact that Profile.
5.5.5.1 Coder Group Settings
Use the Coders Group Settings screen to define up to four different coder groups. These coder
groups are used in the Tel and IP Profile Settings screens to assign different coders to Profiles.
¾ To configure the coder group settings, take these 8 steps:
1. Open the ‘Coder Group Settings’ screen (Protocol Management menu > Profile
Definitions submenu > Coder Group Settings option); the ‘Coder Group Settings’ screen is
displayed.
Figure
5-16: Coder Group Settings Screen
2. In the ‘Coder Group ID’ drop-down list, select the coder group you want to edit (up to four
coder groups can be configured).
3. From the coder drop-down list, select the coder you want to use. For the full list of available
coders and their corresponding ptimes, refer to Table
Note: Each coder can appear only once.
4. From the drop-down list to the right of the coder list, select the size of the Voice Packet
(ptime) used with this coder in milliseconds. Selecting the size of the packet determines how
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many coder payloads are combined into one RTP (voice) packet.
Note 1: The ptime packetization period depends on the selected coder name.
Note 2: If not specified, the ptime gets a default value.
Note 3: The ptime specifies the maximum packetization time the gateway can receive.
5. Repeat steps 3 and 4 for the second to fifth coders (optional).
6. Repeat steps 2 to 5 for the second to forth coder groups (optional).
7. Click the Submit button to save your changes.
8. To save the changes so they are available after a power fail, refer to Section
161.
Note: In the current version, only the ptime of the first coder is sent in the SDP
section of the INVITE message.
Table 5-15: ini File Coder Group Parameters
Parameter Description
CoderName_ID
Coder list for Profiles (up to five coders in each group).
The CoderName_ID parameter (ID from 1 to 4) provides groups of coders that can be
associated with IP or Tel profiles.
You can select the following coders:
g711Alaw64k – G.711 A-law.
g711Ulaw64k – G.711 µ-law.
g7231 – G.723.1 6.3 kbps (default).
g7231r53 – G.723.1 5.3 kbps.
g726 – G.726 ADPCM 32 kbps (Payload Type = 2).
g729 – G.729A.
g729_AnnexB – G.729 Annex B.
The RTP packetization period (ptime, in msec) depends on the selected Coder name,
and can have the following values:
Note: If the coder G.729 is selected, the gateway includes ‘annexb=no’ in the SDP of the
relevant SIP messages. If G.729 Annex B is selected, ‘annexb=yes’ is included. An
exception to this logic is when the remote gateway is a Cisco device (IsCiscoSCEMode).
ini file note 1: This parameter (CoderName_ID) can appear up to 20 times (five coders
in four coder groups).
ini file note 2: The coder name is case-sensitive.
ini file note 3: Enter in the format: Coder,ptime.
For example, the following three coders belong to coder group with ID=1:
CoderName_1 = g711Alaw64k,20
CoderName_1 = g711Ulaw64k,40
CoderName_1 = g7231,90
MediaPack SIP User’s Manual 5. Configuring the MediaPack
5.5.5.2 Tel Profile Settings
Use the Tel Profile Settings screen to define up to four different Tel Profiles. These Profiles are
used in the ‘Endpoint Phone Number’ table to associate different Profiles to gateway’s endpoints,
thereby applying different behavior to different MediaPack ports.
¾ To configure the Tel Profile settings, take these 9 steps:
1. Open the ‘Tel Profile Settings’ screen (Protocol Management menu > Profile Definitions
submenu > Tel Profile Settings option); the ‘Tel Profile Settings’ screen is displayed.
Figure
5-17: Tel Profile Settings Screen
2. In the ‘Profile ID’ drop-down list, select the Tel Profile you want to edit (up to four Tel Profiles
can be configured).
3. In the ‘Profile Name’ field, enter a name that enables you to identify the Profile intuitively and
easily.
4. In the ‘Profile Preference’ drop-down list, select the preference (1-10) of the current Profile.
The preference option is used to determine the priority of the Profile. If both IP and Tel
profiles apply to the same call, the coders and other common parameters of the preferred
Profile are applied to that call. If the Preference of the Tel and IP Profiles is identical, the Tel
Profile parameters are applied.
Note: If the coder lists of both IP and Tel Profiles apply to the same call, an intersection of
the coders is performed (i.e., only common coders remain). The order of the coders is
determined by the preference.
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5. Configure the Profile’s parameters according to your requirements. For detailed information
on each parameter, refer to the description of the screen in which it is configured as an
individual parameter.
6. In the ‘Coder Group’ drop-down list, select the coder group you want to assign to that Profile.
You can select the gateway’s default coders (refer to Section
the coder groups you defined in the Coder Group Settings screen (refer to Section
page 91).
7. Repeat steps 2 to 6 for the second to fifth Tel Profiles (optional).
8. Click the Submit button to save your changes.
5.5.1.3 on page 61) or one of
5.5.5.1 on
9. To save the changes so they are available after a power fail, refer to Section
MediaPack SIP User’s Manual 5. Configuring the MediaPack
5.5.5.3 IP Profile Settings
Use the IP Profile Settings screen to define up to four different IP Profiles. These Profiles are
used in the Tel to IP and IP to Hunt Group Routing tables to associate different Profiles to routing
rules. IP Profiles can also be used when working with Proxy server (set ‘AlwaysUseRouteTable’
to 1).
¾ To configure the IP Profile settings, take these 9 steps:
1. Open the ‘IP Profile Settings’ screen (Protocol Management menu > Profile Definitions
submenu > IP Profile Settings option); the ‘IP Profile Settings’ screen is displayed.
Figure
5-18: IP Profile Settings Screen
2. In the ‘Profile ID’ drop-down list, select the IP Profile you want to edit (up to four IP Profiles
can be configured).
3. In the ‘Profile Name’ field, enter a name that enables you to identify the Profile intuitively and
easily.
4. In the ‘Profile Preference’ drop-down list, select the preference (1-10) of the current Profile.
The preference option is used to determine the priority of the Profile. If both IP and Tel
profiles apply to the same call, the coders and other common parameters of the preferred
Profile are applied to that call. If the Preference of the Tel and IP Profiles is identical, the Tel
Profile parameters are applied.
Note: If the coder lists of both IP and Tel Profiles apply to the same call, an intersection of
the coders is performed (i.e., only common coders remain). The order of the coders is
determined by the preference.
5. Configure the Profile’s parameters according to your requirements. For detailed information
on each parameter, refer to the description of the screen in which it is configured as an
individual parameter.
6. In the ‘Coder Group’ drop-down list, select the coder group you want to assign to that Profile.
You can select the gateway’s default coders (refer to Section
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the coder groups you defined in the Coder Group Settings screen (refer to Section 5.5.5.1 on
page 91).
7. Repeat steps 2 to 6 for the second to fifth IP Profiles (optional).
8. Click the Submit button to save your changes.
9. To save the changes so they are available after a power fail, refer to Section
MediaPack SIP User’s Manual 5. Configuring the MediaPack
5.5.6 Configuring the Endpoint Phone Numbers
From the Endpoint Phone Numbers screen you can enable and assign telephone numbers, hunt
groups (optional) and profiles to the VoIP gateway ports.
¾ To configure the Endpoint Phone Numbers table, take these 4 steps:
1. Open the ‘Endpoint Phone Numbers Table’ screen (Protocol Management menu >
Endpoint Phone Numbers); the ‘Endpoint Phone Numbers Table’ screen is displayed.
Figure
2. Configure the Endpoint Phone Numbers according to Table
in the Phone Number fields for each port that you want to use.
3. Click the Submit button to save your changes, or click the Register or Un-Register buttons
to save your changes and to register / unregister to a Proxy / Registrar.
4. To save the changes so they are available after a power fail, refer to Section
161.
5-19: Endpoint Phone Number Table Screen
5-18. You must enter a number
5.9 on page
Table
5-18: Endpoint Phone Numbers Table
Parameter Description
Channel(s) The numbers (1-8) in the Channel(s) fields represent the ports on the back of the VoIP
gateway.
To enable a VoIP gateway channel, you must enter the port number on this screen.
[n-m] represents a range of ports. For example, enter [1-4] to specify the ports from 1 to
4.
Phone Number In each of the Phone Number fields, enter the telephone number that is assigned to that
channel.
For a range of channels enter the first number in an ordered sequence.
These numbers are also used for port allocation for IP to Tel calls, if the hunt group’s
‘Channel Select Mode’ is set to ‘By Phone Number’.
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Table 5-18: Endpoint Phone Numbers Table
Parameter Description
Hunt Group ID In each of the Hunt Group ID fields, enter the hunt group ID (1-99) assigned to the
channel(s). The same hunt group ID can be used for more than one channel and in
more than one field.
The hunt group ID is an optional field that is used to define a group of common behavior
channels that are used for routing IP to Tel calls. If an IP to Tel call is assigned to a hunt
group, the call is routed to the channel or channels that correspond to the hunt group ID.
You can configure the Hunt Group Settings table to determine the method in which new
calls are assigned to channels within the hunt groups (refer to Section
99).
Note: If you enter a hunt group ID, you must configure the IP to Hunt Group Routing
Table (assigns incoming IP calls to the appropriate hunt group). If you do not configure
the IP to Hunt Group Routing Table, calls don’t complete.
For information on how to configure this table, refer to Section
Profile ID Enter the number of the Tel profile that is assigned to the endpoints defined in the
‘Channel(s)’ field.
Parameter Name in ini File Parameter Format
TrunkGroup_x
For example:
TrunkGroup_1 = 1-4,100
TrunkGroup_2 = 5-8,200,1
Note 1: The numbering of channels starts with 1.
Note 2: ‘Hunt Group ID’ can be set to any number in the range 1 to 99.
Note 3: When ‘x’ (Hunt Group ID) is omitted, the functionality of the TrunkGroup
parameter is similar to the functionality of ChannelList and Channel2Phone parameters.
Note 4: This parameter can appear up to 8 times for 8-port gateways and up to 24 times
for MP-124 gateways.
Note 5: An optional Tel ProfileID (1 to 4) can be applied to each group of channels.
List of phone numbers for MediaPack channels
a, b, c, d
a = first channel.
b = number of channels starting from ‘a’.
c = the phone number of the first channel.
d = Tel Profile ID assigned to the group of channels.
For example: ChannelList = 0,8,101, defines phone numbers 101 to 108 for up to 8
channels.
Note 1: The ini file can include up to 24 ‘ChannelList‘ entries.
Note 2: The ‘ChannelList’ can be used instead of, or in addition to, Channel2Phone
parameter.
Phone number of channel.
Its format: Channel2Phone = ‘<channel>, <number>’
<channel> is 0...23.
Example: ‘Channel2Phone = 0, 1002’
Appears once for each channel: 8 times for 8-port gateways, or 4 times for 4-port
gateways and twice for 2-port gateways.
For 8-port and 24-port gateways it is suggested to use ‘TrunkGroup’ parameter, where
in a single line, all gateway’s phone numbers can be defined.
Note: When ‘Channel2Phone’ is used to define an endpoint, hunt group and profile can’t
be assigned to that endpoint.
MediaPack SIP User’s Manual 5. Configuring the MediaPack
5.5.7 Configuring the Hunt Group Settings
The Hunt Group Settings Table is used to determine the method in which new calls are assigned
to channels within each hunt group. If such a rule doesn’t exist (for a specific hunt group), the
global rule, defined by the Channel Select Mode parameter (Protocol Definition > General
Parameters), applies.
¾ To configure the Hunt Group Settings table, take these 7 steps:
1. Open the ‘Hunt Group Settings’ screen (Protocol Management menu > Hunt Group
Settings); the ‘Hunt Group Settings’ screen is displayed.
Figure
2. In the Routing Index drop-down list, select the range of entries that you want to edit (up to
24 entries can be configured).
5-20: Hunt Group Settings screen
3. In the Hunt Group ID field, enter the hunt group ID number.
4. In the Channel Select Mode drop-down list, select the Channel Select Mode that
determines the method in which new calls are assigned to channels within the hunt groups
entered in the field to the right of this field. For information on available Channel Select
Modes, refer to Table
5. Repeat steps 4 and 5, for each defined hunt group.
6. Click the Submit button to save your changes.
7. To save the changes so they are available after a power fail, refer to Section
161.
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Table 5-19: Channel Select Modes
Mode Description
By phone number Select the gateway port according to the called number (refer to the note
below).
Cyclic Ascending Select the next available channel in ascending cycle order. Always select the
next higher channel number in the hunt group. When the gateway reaches the
highest channel number in the hunt group, it selects the lowest channel
number in the hunt group and then starts ascending again.
Ascending Select the lowest available channel. Always start at the lowest channel number
in the hunt group and if that channel is not available, select the next higher
channel.
Cyclic Descending Select the next available channel in descending cycle order. Always select the
next lower channel number in the hunt group. When the gateway reaches the
lowest channel number in the hunt group, it selects the highest channel
number in the hunt group and then start descending again.
Descending Select the highest available channel. Always start at the highest channel
number in the hunt group and if that channel is not available, select the next
lower channel.
Number + Cyclic Ascending First select the gateway port according to the called number (refer to the note
below). If the called number isn’t found, then select the next available channel
in ascending cyclic order. Note that if the called number is found, but the port
associated with this number is busy, the call is released.
Parameter Name in ini File
TrunkGroupSettings
Parameter Format
TrunkGroupSettings = <Hunt group ID>, <Channel select Mode>
For example:
TrunkGroupSettings = 1,5
<Channel Select Mode> can accept the following values:
• 0 = By Phone Number
• 1 = Cyclic Ascending
• 2 = Ascending
• 3 = Cyclic Descending
• 4 = Descending
• 5 = Number + Cyclic Ascending
Note: This parameter can appear up to 24 times.
Note: The gateway’s port numbers are defined in the ‘Endpoint Phone Numbers’
table under the ‘Phone Number’ column. For detailed information on the
‘Endpoint Phone Numbers’ table, refer to Section 5.5.6 on page 97).