View source for AudioCodes MP114/118, v4.8 and higher

Config guide SIP Gateway - AudioCodes MP114/118 here

== AlphaCom Configuration ==
[[image:Configuration guide for AudioCodes MP114 118 - Configuration example.jpg|thumb|250px|Configuration example]]

The configuration of AlphaCom E with SIP gateway includes the
following steps:

<u>'''Using AlphaWeb'''</u>
* Assign IP address to the AlphaCom Ethernet port
* Insert SIP Trunk licenses
* Firewall (filter) settings

<u>'''Using AlphaPro'''</u>
* Create a SIP Trunk Node
* Create AlphaCom / SIP Audio links
* Define the AlphaCom / SIP routing
* Create prefix numbers
* Update the exchange

=== 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.

[[image:Configuration guide for AudioCodes MP114 118 - Assign IP address to the AlphaCom E Ethernet.jpg|thumb|left|250px|]]
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==== Insert SIP Trunk licenses ====
Log on to AlphaWeb and install the SIP Trunk license.

[[image:Configuration guide for AudioCodes MP114 118 - Insert SIP Trunk licenses.jpg|thumb|left|250px|]]
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==== Firewall (filter) settings ====
Enable the SIP protocol on the desired Ethernet port
(default enabled for Ethernet port1).

[[image:Configuration guide for AudioCodes MP114 118 - Firewall (filter) settings.jpg|thumb|left|250px|]]
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=== 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:

[[image:Configuration guide for AudioCodes MP114 118 - Create a SIP Trunk Node.jpg|thumb|left|250px|]]
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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
[http://10.5.2.54/wiki/index.php/Configuration_guide_for_AudioCodes_MP114/118#Define_the_AlphaCom_.2F_SIP_routing 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.

[[image:Configuration guide for AudioCodes MP114 118 - Create AlphaCom SIP Audio links.jpg|thumb|left|250px|]]
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==== 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''.

[[image:Configuration guide for AudioCodes MP114 118 - Define the AlphaCom SIP routing.jpg|thumb|left|250px|]]
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==== Create Prefix number ====
The directory number (prefix) used to access the telephone line must be
programmed in the AlphaCom directory table with ''feature'' 83 and '''Node'''
= SIP Trunk node number (100 in this example). In the example below
the default directory number 0 has been modified to be used as a prefix.

[[image:Configuration guide for AudioCodes MP114 118 - Create Prefix number.jpg|thumb|left|250px|]]
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==== Update the exchange ====
Log on to the exchange and update the exchange by pressing the
SendAll button.

== AudioCodes MP-144/118 Configuration ==
=== Configure Network Parameters ===
[[image:Configuration guide for AudioCodes MP114 118 - Configure Network Parameters.jpg|thumb|150px|]]
The AudioCodes MP-114/118 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 at the rear of the AudioCodes MP-114/118 unit
for minimum 6 seconds.

* The VoIP Gateway will now get the IP address 10.1.10.11, 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 of the 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 <nowiki>'''http://10.1.10.11'''</nowiki> in the URL field.
[[image:Configuration guide for AudioCodes MP114 118 - Configure Network Parameters (2).jpg|thumb|left|250px|]]
Type in user name ''Admin'' and password ''Admin''. (Case-sensitive!)
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The '''‘Quick Setup’''' screen opens.
[[image:Configuration guide for AudioCodes MP114 118 - Configure Network Parameters (3).jpg|thumb|left|250px|]]

<u>'''''IP Configuration'''''</u>
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.
'''The NAT IP Address''' and '''Default Gateway IP Address''' are set to
''0.0.0.0''

<u>'''''SIP Parameters'''''</u>
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 Gateway and PC to the LAN.
The PC and Gateway must be on the same sub-net.

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

[[image:Configuration guide for AudioCodes MP114 118 - Audio Codec.jpg|thumb|left|250px|]]
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* 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, store the data to flash
memory by selecting the '''Save Configuration''' from the main menu.

'''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 ===
There are two ways of selecting a FXO line from the AlphaCom.
* Group Hunt, where a prefix is dialed and you are connected to one
out of several lines
* Direct FXO line selection, where there is one prefix assigned for
each of the FXO lines.
Both methods can be combined.

==== Group Hunt ====
Dial a prefix and get connected to a free FXO line.

<u>'''''Hunt Group Settings'''''</u>

In the '''Hunt Group Settings''', specify a ''Hunt Group ID'' = 1,

In '''Channel Select Mode,''' select ''Cyclic Ascending''.
'''(Protocol Management > Hunt Group Settings)'''.

[[image:Configuration guide for AudioCodes MP114 118 - Group Hunt.jpg|thumb|left|250px|]]
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<u>'''''IP to Hunt Group Routing'''''</u>

When dialing the prefix from AlphaCom, the call needs to be routed to
the right Hunt Group.

In the example below the call is routed to group hunt ID 1. '''(Protocol
Management > Routing Tables > IP to Hunt Group Routing)'''.

[[image:Configuration guide for AudioCodes MP114 118 - IP to Hunt Group Routing.jpg|thumb|left|250px|]]
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<u>'''''Endpoint Phone Number'''''</u>
In the '''Endpoint Phone Number Table''' the FXO lines are linked to the
prefix in AlphaCom and to the hunt group ID.
'''Protocol Management > Endpoint Phone Numbers)'''.

In the example below all four FXO lines belong to Hunt Group ID 1.
When dialing 0 on a station the first available line will be granted.
Directory number 0 must be programmed in the '''AlphaCom directory
table''' with ''feature 83/<node>''. See [http://10.5.2.54/wiki/index.php/Configuration_guide_for_AudioCodes_MP114/118#Create_Prefix_number Create Prefix Number]

[[image:Configuration guide for AudioCodes MP114 118 - Endpoint Phone Number.jpg|thumb|left|250px|]]
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If there are unused lines, leave the fields for that line blank.

==== FXO Line Select ====
In installations with different types of equipment connected to the
various FXO lines the user must be able to select which FXO port to
use. On a ship, for instance, there could be a mix of shore lines, GSM
interface and SatCom lines. Line selection is achieved by assigning
each port a '''Phone Number''' in the '''Endpoint Phone Number Table'''.
These directory numbers must be programmed in the AlphaCom
'''directory table''' ''with feature 83/<node>''. See [http://10.5.2.54/wiki/index.php/Configuration_guide_for_AudioCodes_MP114/118#Create_Prefix_number Create Prefix Number]

Replace “0” with four lines, “41” to “44”.

[[image:Configuration guide for AudioCodes MP114 118 - FXO Line Select.jpg|thumb|left|250px|]]
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In this table the four FXO lines are selected by dialing 41 – 44

If there are unused lines, leave all fields for that line blank.

Group hunt is not used in this call mode, and the '''IP to Hunt Routing
Table''' must be empty.

[[image:Configuration guide for AudioCodes MP114 118 - FXO Line Select (2).jpg|thumb|left|250px|]]
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=== Telephone Network to AlphaCom ===
You can choose between three different ways of handling an incoming
call from the telephone line:

* Selective Dialing
* Automatic Dialing
* Delayed Automatic Dialing

==== Selective Dialing ====
A second dial tone will be presented when calling in, and the user can
dial the desired intercom number. The fields in the '''Automatic Dialing'''
table must be left blank (Default setting).
('''Protocol Management > Endpoint Settings > Automatic Dialing''').

[[image:Configuration guide for AudioCodes MP114 118 - Selective Dialing.jpg|thumb|left|250px|]]
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In this mode the gateway collects digits from the line, and sets up the
call towards the AlphaCom when a predefined number of digits are
collected and no more digits are received within a preset time (default 4
seconds), or when the ‘#’ key is dialed.

In '''Protocol Management > Protocol Definition > DTMF & Dialing'',
set '''Max Digits In Phone Num''' equal to the number of digits used on the
AlphaCom stations, normally 3 or 4. The parameter '''Inter Digit Timeout'''
specifies the waiting time for more digits before setting up the call.

[[image:Configuration guide for AudioCodes MP114 118 - Selective Dialing (3).jpg|thumb|left|250px|]]
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==== Automatic Dialing (Call to Switchboard) ====
When calling in, the call will automatically be connected to a predefined
intercom number.

Enter the intercom number in the '''Destination Phone Number field''' in
the '''Automatic Dialing''' table. Set '''Auto Dial Status''' to Enable.
'''(Protocol Management > Endpoint Settings > Automatic Dialing)'''.

[[image:Configuration guide for AudioCodes MP114 118 - Automatic Dialing (Call to Switchboard).jpg|thumb|left|250px|]]
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In the example above, calls on line 1, 3 and 4 are routed to station 103,
and incoming calls on line 2 are routed to station 101.

==== Delayed Automatic Dialing ( Delayed Call to Switchboard) ====
If '''Auto Dial Status''' is set to ''Hotline'', a second dial tone will be
presented when calling in, allowing the user to dial a number. But if no
digits are pressed within the '''Hotline Dial Tone Duration''' time, the
number in the '''Destination Phone Number''' is automatically dialed.

[[image:Configuration guide for AudioCodes MP114 118 - Delayed Automatic Dialing.jpg|thumb|left|250px|]]
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The '''Hotline Dial Tone Duration''' can be changed. The default value is
16 seconds. ('''Protocol Management > Protocol Definition > DTMF &
Dialing > Hotline Dial Tone Duration'''').

==== Caller ID ====
Use the '''Caller Display Information''' table to send display information to
the intercom station that receives the call. ('''Protocol Management >
Endpoint Settings > Caller ID''').

[[image:Configuration guide for AudioCodes MP114 118 - Caller ID.jpg|thumb|left|250px|]]
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The prefix code entered in the '''End Point Phone Number Table''' will be
shown together with the text in '''Caller ID/Name'''.

If Caller ID name is detected on the FXO line, this will be used instead
of the Caller ID name in the table above. Caller ID from FXO line must
be enabled in '''Protocol Management > Advanced Parameters >
Supplementary Services'''.

Set '''Enable Caller ID''' to Enable and choose the CID protocol used by
the PSTN supplier in '''Caller ID Type'''.
Check with the local telephone company to find the CID protocol used.

[[image:Configuration guide for AudioCodes MP114 118 - Caller ID (2).jpg|thumb|left|250px|]]
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== Far End Disconnect (FED) ==
Far End Disconnect refers to methods for detecting that a remote party
has hung up. The far end disconnect signal is not mandatory and this
could create problems. If the Far End Disconnect signal is not sent to or
properly detected by the SIP Gateway, the connection will not be
released by the unit, thus freezing the FXO line in the off hook state.

=== Call Termination options in the SIP Gateway ===
The following methods for call termination are supported by the
AudioCodes MP-114/118. Note that the used disconnection methods
must be supported by the CO (Central Office) or to PBX (Private Branch
Exchange).

* Detection of polarity reversal / current disconnect
* Detection of Busy / Dial tones
* Detection of silence
* Timeout of Conversation

==== Detection of polarity reversal / current disconnect ====
This is the recommended method. The call is immediately disconnected
after polarity reversal or current disconnect is detected on the Tel side
(assuming the PBX / CO produces this signal).

Open the General Parameters screen ('''Protocol Management >
Advanced Parameters > General Parameters''') and enable the
relevant detection method.

[[image:Configuration guide for AudioCodes MP114 118 - Detection of polarity reversal current disconnect.jpg|thumb|left|250px|]]
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==== Detection of Busy / Dial tones ====
The call is immediately disconnected after Busy or Dial tone is detected
on the Tel side (assuming the PBX / CO produces this tone). This
method requires the correct tone frequencies and cadence to be defined
in the Call Progress Tones (CPT) file of the SIP Gateway. If these
frequencies are not known, define them in the CPT file (the tone
produced by the PBX / CO must be recorded and its frequencies
analyzed). This method is slightly less reliable than the previous one.

A file with the most common tone patterns can be downloaded from
http://www.zenitel.com/stentofon/support.

Open the '''FXO Settings''' screen ('''Protocol Management > FXO
Settings > FXO Settings''') and enable the relevant detection method.

[[image:Configuration guide for AudioCodes MP114 118 - Detection of Busy Dial tones.jpg|thumb|left|250px|]]
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<u>'''''Call Progress Tones (CPT)'''''</u>
The Detection of Busy / Dial tones method requires the correct tone
frequencies and cadence (on/off sequence) to be defined in the Call
Progress Tones (CPT) file of the SIP gateway. These tones are region
specific and telephone exchange dependent.

The Call Progress Tones (CPT) configuration file is a binary file (with the
extension '''.dat''').

Users can either use one of the supplied configuration (dat) files found
on the CD provided with the gateway, or construct their own file.

Either:
* Modify the supplied usa_tone.ini file (in any standard text editor) to suit the specific requirements, and convert the modified ini file into binary format using the '''TrunkPack Downloadable Conversion Utility'''.
Or:
* Use the Call Progress Tones Wizard.
The '''Call Progress Tones Wizard''' (CPTWizard) is an application
designed to detect the Call Progress Tones generated by your PBX (or
telephone exchange) to create a basic Call Progress Tones ''ini'' file
containing definitions for all relevant Call Progress Tones. This provides
a good starting point when configuring the SIP gateway. This ''ini'' file can
then be converted to a ''dat'' file using the '''TrunkPack Downloadable
Conversion utility'''.

Both the '''TrunkPack Downloadable Conversion Utility'''
(''DConvert.exe'') and the '''Call Progress Tones Wizard''' (''CPTWizard.exe'')
are provided with the AudioCodes CD.

Load a Call Progress Tones (''dat'') file to the SIP gateway:

* Open the Regional Settings screen ('''Advanced Configuration > Regional Settings''').

[[image:Configuration guide for AudioCodes MP114 118 - Detection of Busy Dial tones (2).jpg|thumb|left|250px|]]
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* Click the Browse button and navigate to the folder that contains the file you want to load
* Click the file and click the Open button; the name and path of the file appear in the field beside the Browse button
* Click the Send File button
* Save configuration so the file can be available after a power failure
* Reset the SIP Gateway for the changes to take effect

For more detailed information regarding Call Progress Tones please
refer to the following sections in the AudioCodes ‘LTRT-65406
MediaPack SIP User's Manual Ver 4.8’ on the CD supplied with the unit:

* Section 15.1 – on how to configure the CPT ini file
* Section D.1.1 – on how to convert CPT ini file to a binary dat file
* Section D.2 – on how to use the Call Progress Tones Wizard (CPTWizard) application

==== Detection of Silence ====
The call is disconnected after silence is detected on both call directions
for a specific (configurable) amount of time. This method should only be
used as a backup.

Open the '''General Parameters screen (Protocol Management >
Advanced Parameters > General Parameters''').

[[image:Configuration guide for AudioCodes MP114 118 - Detection of Silence.jpg|thumb|left|250px|]]
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==== Timeout of Conversation ====
As an additional safety to prevent lines from accidentally locking up, it is
recommended to enable a timeout of conversation.

The '''Max Call Duration''' defines the maximum call duration in minutes. 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). ('''Protocol
Management > Advanced Parameters > General Parameters: Max
Call Duration''').

[[image:Configuration guide for AudioCodes MP114 118 - Timeout of Conversation.jpg|thumb|left|250px|]]

== Miscellaneous Features ==
[[image:Configuration guide for AudioCodes MP114 118 - Incoming Calls in Private.jpg|thumb|left|250px|]]
=== 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 (''from AlphaPro/AMC 10.05'').

Check the flag '''Private Ringing from SIP''' in AlphaPro, ('''Exchange &
System > System > Calls and Options''').

=== 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'''.

[[image:Configuration guide for AudioCodes MP114 118 - SIP Gateway configuration.jpg|thumb|left|250px|]]

==== AlphaCom configuration ====
[[image:Configuration guide for AudioCodes MP114 118 - AlphaCom configuration.jpg|thumb|left|250px|]]
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

<u>'''''Calling from the telephone to the door''''</u>
The Standard door opening event is used.

[[image:Configuration guide for AudioCodes MP114 118 - AlphaCom configuration (2).jpg|thumb|left|250px|]]