AudioCodes Mediant 1000

Configuration example

This article describes the setup of the AlphaCom E system and the AudioCodes Mediant 1000 Gateway. The Mediant 1000 supports mixed digital and analog interface configurations, providing up to four E1/T1/J1 spans and up to 24 analog interfaces in various FXO/FXS configurations.

In this article the configuration for digital interface is described, and covers the most common features used in an AlphaCom E/AudioCodes interconnection.

AlphaCom Configuration

AlphaWeb Configuration

Assign IP address to the AlphaCom E Ethernet port(s)

Log on to AlphaWeb and enter a valid IP address on the Ethernet port. In the example below, Ethernet port 1 is used. Consult your network administrator to obtain the IP address.

Insert SIP Trunk licenses

Log on to AlphaWeb and install the SIP Trunk license.

Firewall (filter) settings

Enable the SIP protocol on the desired Ethernet port (default enabled for Ethernet port1).

AlphaPro Configuration

Create a SIP Trunk Node

From the AlphaPro main menu, use the ‘+’ button next to the ‘Select Exchange’ dropdown list to create a new exchange. The exchange type must be set to ‘SIP Node’.

Set the parameters as follows:

The SIP Trunk IP address must be identical to the IP address of the SIP Gateway.

Note: If the AlphaCom is configured with a SIP Registar node in addition to the SIP Trunk node, the SIP Registar node must have a lower node number than the SIP Trunk node.

Create AlphaCom/SIP Audion links

This paragraph is only relevant for AMC software 10.04 or earlier.

From AMC 10.05 the audio links are assigned dynamically whenever needed, and there is no need to specify the links in AlphaPro. Proceed Define the AlphaCom / SIP routing.

However, if you want to reserve VoIP channels for the SIP Gateway, you can do so by following the description below.

In Exchange & System > NetAudio use the Insert button to create one or several audio (VoIP) links between the AlphaCom and the SIP Gateway. The physical number specifies the VoIP channel and must be in the range 605 – 634 (start with 605). Normally the number of audio links will be equal to the number of phone lines connected to the SIP Gateway.

Define the AlphaCom / SIP routing

In Exchange & System > Net Routing use the Insert button to create a route between the AlphaCom and SIP Gateway. Set Preferred codec to G711u and RTP Packet Size to 10 ms.

Prefix and Global numbers

The telephone line can be accessed in different ways:

• Prefix number: Dial Prefix + Phone number. “Phone number” will be called
• Integrated Prefix number: Dial Prefix + Phone number. The prefix will be included as a part of the called telephone number.
• Global number: Dial the phone number without using prefix

Prefix number

The directory number must be programmed in the AlphaCom directory table with feature 81 and Node = SIP Trunk node number (100 in this example). In the field “Collect N more digits (SIP)” you must enter the maximum number of digits in a phone number.

When the prefix is dialed, the AlphaCom will wait for further digits. When the number of digits specified in the “Collect N more digits (SIP)” is collected, a call setup message is sent to the SIP gateway. If fewer digits are entered, the AlphaCom will time out after 4 seconds, and the call setup message will be sent. You can also terminate the digit collection by pressing the M-key. The call setup message will then be sent immediately.

Example of prefix number

In the example to the right the directory number 0 is used as a prefix.

Dialing examples:

0 + 12345678: Telephone number 12345678 will be called

0 + 1234: After a 4 second timeout, telephone number 1234 will be called

0 + 1234 + M: Telephone number 1234 will be called

Integrated Prefix number

The directory number must be programmed in the AlphaCom directory table with feature 83 and Node = SIP Trunk node number (100 in this example). In the field “Collect N more digits (SIP)” you must enter the maximum number of digits in a phone number.

When the prefix is dialed, the AlphaCom will wait for further digits. When the number of digits specified in the “Collect N more digits (SIP)” is collected, a call setup message is sent to the SIP gateway. If fewer digits are entered, the AlphaCom will time out after 4 seconds, and the call setup message will be sent. You can also terminate the digit collection by pressing the M-key. The call setup message will then be sent immediately.

Example of integrated prefix number

In the example to the right the directory number 57 is used as a prefix.

Dialing examples:

57 + 12345678: Telephone number 5712345678 will be called

57 + 1234: After a 4 second timeout, telephone number 571234 will be called

57 + 1234 + M: Telephone number 571234 will be called

Global number

Example of global number

The directory number must be programmed in the AlphaCom directory table with feature 83 and Node = SIP Trunk node number (100 in this example). The field “Collect N more digits (SIP)” must be left blank.

When the global number is dialed, the AlphaCom will immediately send a call setup message to the SIP gateway.

In the example to the right the directory number 12345678 is defined as a global number. When dialing this number a call setup message is sent to the SIP gateway, instructing it to call this phone number.

Update the exchange

Log on to the exchange and update the exchange by pressing the SendAll button.

AudioCodes Mediant 1000 Configuration

Configure Network Parameters

Configure network parameters.jpg

The AudioCodes Mediant 1000 VoIP Gateway comes with default network parameters (factory default parameters).

Before you can set up the gateway in the network, you have to change the default IP address to a fixed IP address in your network environment. The unit is configured from a web browser, e.g. Internet Explorer or Navigator. Consult the network administrator to get the correct IP address.

Follow these steps:

• Load factory network parameters and reset the username and password to its default settings (username: Admin, password: Admin) by pressing the reset button located to the right of ethernet port II, and directly above the RS-232 port, labeled // for minimum 6 seconds.
  • The VoIP Gateway will now get the IP address 10.1.10.10, submask 255.255.0.0.
• Change the IP address of your PC to 10.1.10.12, submask 255.255.0.0.
• Connect the LAN port of the PC to the Ethernet port I of the VoIP Gateway. Use a crossed cable or connect the PC and the VoIP Gateway to a common switch using straight cables.
• Start your Web Browser and type 'http://10.1.10.10' in the URL field. A log-in window appear, type in user name Admin and password Admin. (Case-sensitive!). Now the ‘Quick Setup’ screen opens.

Quick Setup menu - Mediant 1000

IP Configuration
Enter the IP Address and Subnet Mask of the AudioCodes Gateway. This IP address must be identical to the IP address of the SIP Trunk Node created in AlphaPro.

SIP Parameters
Set Working with Proxy to Yes. Enter the IP address of the AlphaCom in the Proxy IP Address field. Set Enable Registration to Disable. Leave Gateway Name and Proxy Name blank.

• Click Reset button and OK button to apply the changes.
• Disconnect the PC from the Gateway.
• Reconnect the Ethernet port I of the VoIP Gateway to the LAN
• Reconnect the PC to the LAN.
• Restore the PC’s IP address and subnet mask to what they originally were, and re-access the Gateway using the new assigned IP address.

Audio Codec

Audio Codec settings

In the Quick Setup screen, select Coders Table. Choose G.711U-law codec, 10 ms packet size and silence suppression Disabled.

• Press Submit to save changes.

About Saving Changes

The Submit button will save the data to the running volatile memory. The changes take effect on-the-fly. The changes will not survive hardware reset or power off.

To permanently save the configuration data you need to store the data to flash memory by selecting Maintenance from the main menu. Click the BURN button. A confirmation message appears when the save is completed successfully.

Note: Parameters proceeded by an exclamation mark (!) is not changeable on-the-fly and require that the device is reset.

Backup and Restore

The configuration of the AudioCodes Gateway can be stored to a file on your PC. Use the Configuration File menu to store or restore the configuration (Advanced Configuration > Configuration File).

AlphaCom to Telephone Network

Trunk Settings

In Advanced Configuration > Trunk Settings you specify the properties of the trunk line. Select the protocol to be used for the trunk (Protocol Type), the trunk clock source (Clock Master), Line Code and physical framing to be used (Framing Method), and other relevant parameters.

Mediant 1000 - Trunk Settings

Define a Trunk Group

In Protocol Management > Trunk Group you specify which channels to use on the E1 trunk, and assign the trunk to a Trunk Group ID and a Profile ID:

Mediant 1000 - Trunk Group Table

Trunk Group Settings

In Protocol Management > Trunk Group Settings you assign the rules for channel allocation:

Mediant 1000 - Trunk Group settings

IP to Trunk Group routing

In Protocol Management > Routing Tables > IP to Hunt Group Routing you specify the Trunk Group ID to use for call from IP (i.e. calls from AlphaCom E). In the example below all calls from AlphaCom are routed to Trunk Group 1.

IP to Trunk Group Routing Table

Telephone Network to AlphaCom

Because the AlphaCom is defined as a Proxy Server in the SIP Gateway, you do not need to configure any Tel to IP Routing. The calling telephone number is automatically forwarded to the Proxy Server (i.e. the AlphaCom).

Miscellaneous Features

Incoming Calls in Private

Incoming calls from the telephone line can be forced to be in private ringing mode, independent of the private/open switch of the intercom station (available from AlphaPro/AMC 10.05).

AlphaPro 10.05: Check the flag Private Ringing from SIP in Exchange & System > System > Calls and Options.

AlphaPro 10.26: Check the flag Incoming calls from SIP in private ringing mode in Exchange & System > System > VoIP.

Door Opening Feature

During a conversation between a door station and a telephone, the telephone operator can activate the Door Opening feature in the AlphaCom by pressing digit 6.

SIP Gateway Feature

To enable digit actions from the telephone line during conversation, set 1st Tx DTMF Option to INFO(Cisco) in Protocol Management > Protocol Definition > DTMF & Dialing.

AlphaCom configuration

Door Opening when SIP calls the door

The Door Opening feature is programmed in the Event Handler. There are two separate events for the door opening feature, depending on who is the calling side:

• calling from the telephone to the door
• calling from the door to the telephone

Calling from the telephone to the door' The Standard door opening event is used.

Door Opening when door calls out on SIP gateway

Calling from the door to the telephone:

When the phone presses digit 6, the event type Event Trigger Feature (15) is reported, with the digit 6 as sub event. The calling AlphaCom station is Event Owner, and called SIP phone number and node number is Related To. The RCO pulse time is specified as an additional parameter in the RCO action string, i.e. RCO 3 ON 20 means pulse RCO 3 for 2 seconds.

M-key Control from Telephone Network

The ‘*’ and ‘#’ buttons on the telephone can be used to control M-key function (simplex audio) ON or OFF:

• Press the ‘*’-key briefly and the M-key is turned ON
• Press the ‘#’-key briefly and the M-key is turned OFF

This can be useful for group call announcement from the telephone. The feature is enabled by setting 1st Tx DTMF Option to INFO(Cisco) in Protocol Management > Protocol Definition > DTMF & Dialing.

Transmit ‘*’ and ‘#’ from AlphaCom

The DTMF signals ‘*’ and ‘#’ will be transmitted to the line when DAK 0 (*) and DAK 1 (#) is pressed during a telephone conversation. (No programming is required).

Voice Switching in Noisy Environment

If the intercom station is located in a noisy environment, it might be difficult to switch the voice direction from the telephone towards the intercom station. However, there is a setting in the AlphaCom (from AMC 10.05) to overcome this problem. Use the nvram editor in the TST Console to set the parameter:

.ex_profile.flags.DSP_duplex = 1

Contact your local STENTOFON dealer for TST programming help.

When this flag is set, the initial voice direction is forced to be from the intercom towards the telephone. When the phone operator starts to speak, the voice direction will switch towards the intercom station, regardless of the level of the audio signal from the intercom station. As soon as the phone operator stops speaking, the voice direction will switch back to the initial direction.

Make sure that the Echo Canceler is enabled in the SIP Gateway. (Protocol Management > Profile Definitions > Tel Profile Settings > Echo Canceler = Enable)

Country Settings

The Line Characteristics (AC impedance matching, hybrid balance, Tx & Rx frequency response, Tx & Rx Gains, ring detection threshold, DC characteristics) should be set according to country of origin.

Some of the SIP Gateway parameters are configurable through the ini configuration file only (and not via the Web). The CountryCoefficients parameter that determines the line characteristics must be configured via the ini configuration file.

Procedure to modify the ini file:

• Get the ini file from the gateway using the Embedded Web Server

(Advanced Configuration > Configuration File > Get ini File):

• Open the file (the file is open in Notepad or a Customer-defined text

file editor) and add anywhere in the file the line

CountryCoefficients = xx

• where xx is the country code found below; save and close the file.

The example shows the settings for Norway (46).

• Load the modified ini file back to the gateway (using the Send ini

File).

This method preserves the programming that already exists in the device, including special default values that were preconfigured when the unit was manufactured.

Country codes

The default is 70 (United States).

Argentina	= 0	Finland	= 18	Lebanon	= 36	Russia	= 54
Australia	= 1	France	= 19	Luxembourg	= 37	Saudi_Arabia	= 55
Austria	= 2	Germany	= 20	Macao	= 38	Singapore	= 56
Bahrain	= 3	Greece	= 21	Malaysia	= 39	Slovakia	= 57
Belgium	= 4	Guam	= 22	Malta	= 40	Slovenia	= 58
Brazil	= 5	Hong_Kong	= 23	Mexico	= 41	South_Africa	= 59
Bulgaria	= 6	Hungary	= 24	Morocco	= 42	South_Korea	= 60
Canada	= 7	Iceland	= 25	Netherlands	= 43	Spain	= 61
Chile	= 8	India	= 26	New_Zealand	= 44	Sweden	= 62
China	= 9	Indonesia	= 27	Nigeria	= 45	Switzerland	= 63
Colombia	= 10	Ireland	= 28	Norway	= 46	Syria	= 64
Croatia	= 11	Israel	= 29	Oman	= 47	Taiwan	= 65
Cyprus	= 12	Italy	= 30	Pakistan	= 48	TBR21	= 66
Czech_Republic	= 13	Japan	= 31	Peru	= 49	Thailand	= 67
Denmark	= 14	Jordan	= 32	Philippines	= 50	UAE	= 68
Ecuador	= 15	Kazakhstan	= 33	Poland	= 51	United_Kingdom	= 69
Egypt	= 17	Latvia	= 35	Romania	= 53	Yemen	= 71

Feature Guide

• Make a call from an intercom station: Dial prefix – wait for the dial

tone – dial phone number

• When pressing digits during connection, DTMF digits are sent (Call

center etc.)

• The DTMF signals ‘*’ and ‘#’ will be transmitted when pressing DAK

0 (*) and DAK 1 (#)

• A complete phone number can be stored under a DAK key or a substation call button (From AMC 10.05)
  • Program from station: 784 + <prefix> + <phone number> + M + DAK key Example: 784 + 0 + 40002500 + M + DAK key
  • Program from AlphaPro:

I <prefix> P <phone number> Example: I 0 P 40002500

Call Transfer

• Incoming calls from the line can be transferred to another station
  • From keypad: 2 + <intercom station> + 3
  • From preprogrammed DAK:

D 2 I 104 M M D 3

• Outgoing calls to the line can be transferred to another station
  • From keypad:

DAK8 + 2 + <intercom station> + 3

Call Forwarding (From AMC 10.05)

• An intercom call can be forwarded to a telephone
  • From keypad: 71 + 0 + <phone number> + M
  • From preprogrammed DAK: I 71 I 0 P 400025000

Search List (From AMC 10.05)

• A telephone number can be included in the Search List of a station.
  • Format: I <prefix> P <phone number> Example: I 0 P 40002500