Latest revision as of 11:39, 15 August 2022

The Mediant 600 has the capability to transcode a SIP call from e.g. G711µ-law to G729. Here are some examples using the Mediant 600 to transcode G711µ-law to G729.

The examples shown are between:

AlphaCom XE and Cisco Call Manager
AlphaCom XE and AudioCodes MP-114/118
AlphaCom XE and Mobile VoIP GSM Gateway MV-370

Configuration example

1 AlphaCom Configuration
2 Mediant 600 Configuration
3 Configuration of Mediant 600 with SIP Account as End Point
4 AudioCodes MP-114/118 Configuration
- 4.1 SIP Parameters
- 4.2 Audio Codec
5 Cisco Call Manager Configuration
- 5.1 Configure a Region
- 5.2 Configure a SIP Trunk
6 Mobile VoIP GSM Gateway MV-370
- 6.1 SIP settings
- 6.2 Codec settings

AlphaCom Configuration

How to set up a SIP Trunk in the AlphaCom is described here: SIP trunk node - configuration

Mediant 600 Configuration

A license SIP to SIP Mediation and Routing Application (SW/IP2IP/15) is required. With this license, the Mediant600 has 30 active Media Channels that can support 15 transcoding simultane sessions.

Restore to factory default settings

Reset Button

With a paper clip, press and hold down the Reset button (located on the CPU Module) for at least 12 seconds (no more than 25 seconds)
The device restores to factory default settings

Access to the embedded web server

Home Page

The default Network parameter of the Mediant 600:

IP Adress : 10.1.10.10
Subnet Mask : 255.255.0.0
Default Gateway : 0.0.0.0

Connect your PC directly to the device, using an Ethernet Crossover cable

Before the PC can access to the Mediant 600, the IP address of the PC must be changed to match the same subnet (e.g 10.1.10.11)

To access the embedded web server, start your internet browser (e.g. Internet Explorer) and in the address field enter 10.1.10.10

You will be prompted for a username and password (default):

Username: Admin
Password: Admin

The Home Page page should now be displayed

Note: For the rest of configuration used the tree menu in Full mode

IP Configuration

Initial configuration

RETURN TO THIS

IP Settings

In the "IP Settings" page (Configuration tab -> Network Settings menu -> IP Settings page) enter :

IP Adress
Subnet Mask
Default Gateway
Click "Submit" to apply the changes

The IP address is immediately changed when pressing Submit, but it is not permanently stored

Without resetting or powering off the device, you need to log on to the Gateway using its new IP address in order to Burn the new IP address to flash:

Disconnect the PC from the Gateway
Connect 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
Click "Burn" to permanently apply the changes

Add Software Upgrade Key

The license for SIP IP2IP must be uploaded as a Software Upgrade Key.

Software Upgrade Key

In the "Software Upgrade Key Status" page (Management tab -> Software Update menu -> Software Upgrade Key page):

Enter the license key in the "Add a Software Upgrade Key" field
Click "Add Key"
Burn
Reset the device (Device Action -> Reset -> Reset)

Enable IP-to-IP Capabilities

Applications Enabling

In the "Applications Enabling" page (Configuration tab -> Protocol Configuration menu -> Applications Enabling page):

From the "Enable IP2IP Application" drop-down list, select "Enable"
Submit & Burn
Wait the next step for Reset the device

Number of Media Channels

IP Media Settings

In the "IP Media Settings" page (Configuration tab -> Media Settings menu -> IP Media Settings page):

In the "Number of Media Channeles" field, enter the required number of media channels: 30 (maximum capacity of Mediant 600 is 60)
Submit & Burn
Reset the device (Device Action -> Reset -> Reset)

Proxy Sets Table

These "Proxy Sets" are later assigned to "IP Groups" Note that the "Proxy Set" represents the actual destination to which the call is routed

In the "Proxy Sets Table" page (Configuration tab -> Protocol Configuration menu -> Proxies,Registration,IP Groups submenu -> Proxy Sets Table page):

Proxy Set ID1

Proxy Set ID#1 for AlpahCom XE

From the "Proxy Set ID" drop-down list, select "1"
In the "Proxy Address" column, enter the IP address of the AlphaCom
From the "Transport Type" drop-down list corresponding to the IP address entered above, select "UDP"
Submit

Proxy Set ID2

Proxy Set ID#2 for MP-114

From the "Proxy Set ID" drop-down list, select "2"
In the "Proxy Address" column, enter the IP address of the MP-114
From the "Transport Type" drop-down list corresponding to the IP address entered above, select "UDP"
Submit & Burn

IP Group Table

These "IP Groups" are later used by the device for routing calls

In the "IP Group Table" page (Configuration tab -> Protocol Configuration menu -> Proxies,Registration,IP Groups submenu -> IP Group Table page):

IP Group 1

IP Group #1 for AlpahCom XE

From the "Index" drop-down list, select "1"
In the "Description" field, type an arbitrary name for the IP Group (e.g. AlphaCom)
From the "Proxy Set ID" drop-down list, select "1" (For this "IP Group" communicate with the "Proxy Set" of the AlphaCom)
In the "SIP Group Name" field, enter the IP Adress sent in the SIP Request From/To headers for this IP Group (AlphaCom's IP Adress)
Contact User = name that is sent in the SIP Request contact header for this IP Group (e.g. AXE)
From the "IP Profile ID" drop-down list, select "1" (the IP Profile is configured later)
Submit

IP Group 2

IP Group #2 for MP-114

From the "Index" drop-down list, select "2"
In the "Description" field, type an arbitrary name for the IP Group (e.g. MP114)
From the "Proxy Set ID" drop-down list, select "2" (For this "IP Group" communicate with the "Proxy Set" of the MP-114)
In the "SIP Group Name" field, enter the IP Adress sent in the SIP Request From/To headers for this IP Group (MP-114's IP Adress)
Contact User = name that is sent in the SIP Request contact header for this IP Group (e.g. MP114)
From the "IP Profile ID" drop-down list, select "2" (the IP Profile is configured later)
Submit & Burn

IP Profiles for Voice Coders

For use transcoding it's necessary create two IP Profiles for define two types of coders used.

These profiles are later used in the "Inbound IP Routing" and "Outbound IP Routing" tables.

In the "Coder Group Settings" page (Configuration tab -> Protocol Configuration menu -> Coders And Profile Definitions submenu -> Coder Group Settings page):

Coder Group 1

Coder Group ID#1 for AlphaCom XE

From the "Coder Group ID" drop-down list, select "1"
In the "Coder Name" drop-down list, select "G.711U-law" (Coder used by AlphaCom)
Submit

Coder Group 2

Coder Group ID#2 for MP-114

From the "Coder Group ID" drop-down list, select "2"
In the "Coder Name" drop-down list, select "G.729" (Coder used by MP-114)
Submit & Burn

In the "IP Profile Settings" page (Configuration tab -> Protocol Configuration menu -> Coders And Profile Definitions submenu -> IP Profile Settings page):

Profile ID 1

Profile ID#1 for AlpahCom XE

From the "Profile ID" drop-down list, select "1"
From the "Coder Group" drop-down list, select "Coder Group 1"
Submit

Profile ID 2

Profile ID#2 for MP-114

From the "Profile ID" drop-down list, select "2"
In the "Coder Group" drop-down list, select "Coder Group 2"
Submit & Burn

IP Profiles for Voice Coders

For use transcoding it's necessary create two IP Profiles for define two types of coders used.

These profiles are later used in the "Inbound IP Routing" and "Outbound IP Routing" tables.

In the "Coder Group Settings" page (Configuration tab -> Protocol Configuration menu -> Coders And Profile Definitions submenu -> Coder Group Settings page):

Coder Group 1

Coder Group ID#1 for AlpahCom XE

From the "Coder Group ID" drop-down list, select "1"
In the "Coder Name" drop-down list, select "G.711U-law" (Coder used by AlphaCom)
Submit

Coder Group 2

Coder Group ID#2 for MP-114

From the "Coder Group ID" drop-down list, select "2"
In the "Coder Name" drop-down list, select "G.729" (Coder used by SIP Server)
Submit & Burn

In the "IP Profile Settings" page (Configuration tab -> Protocol Configuration menu -> Coders And Profile Definitions submenu -> IP Profile Settings page):

Profile ID 1

Profile ID#1 for AlpahCom XE

From the "Profile ID" drop-down list, select "1"
From the "Coder Group" drop-down list, select "Coder Group 1"
Submit

Profile ID 2

Profile ID#2 for MP-114

From the "Profile ID" drop-down list, select "2"
In the "Coder Group" drop-down list, select "Coder Group 2"
Submit & Burn

Inbound IP Routing

The "IP to Trunk Group Routing Table" it used for define the routing inbound IP-to-IP calls.

The table in which this is configured uses the IP Groups that you defined before.

In the "IP to Trunk Group Routing Table" page (Configuration tab -> Protocol Configuration menu -> Routing Tables submenu -> IP to Trunk Group Routing page):

Inbound IP Routing

Index#1 for AlpahCom XE7

Dest Phone Prefix : enter the asterisk (*) symbol to indicate all destinations
Source Phone Prefix : enter the asterisk (*) symbol to indicate all destinations
Source IP Adress : enter the IP adress of the AlphaCom
Trunk Group ID : enter "-1" to indicate that these calls are IP-to-IP calls
IP Profile ID : enter "1" to assign these calls to "ProfileID#1" to use "G.711U-law"
Source IP Groupe ID : enter "1" to assign these calls to the IP Group pertaining to the AlphaCom

Index#2 for MP-114

Dest Phone Prefix : enter the asterisk (*) symbol to indicate all destinations
Source Phone Prefix : enter the asterisk (*) symbol to indicate all destinations
Source IP Adress : enter the IP adress of the MP-114
Trunk Group ID : enter "-1" to indicate that these calls are IP-to-IP calls
IP Profile ID : enter "2" to assign these calls to "ProfileID#2" to use "G.729"
Source IP Groupe ID : enter "2" to assign these calls to the IP Group pertaining to the MP-114
Submit & Burn

Outbound IP Routing

The "Tel to IP Routing Table" it used for define the routing outbound IP-to-IP calls.

The table in which this is configured uses the IP Groups that you defined before.

In the "Tel to IP Routing" page (Configuration tab -> Protocol Configuration menu -> Routing Tables submenu -> Tel to IP Routing page):

Outbound IP Routing

Index#1 from AlpahCom XE7 to MP-114

Src.IPGroupID : select "1" to indicate received (inbound) calls identified as belonging to the IP Group configured for AlphaCom
Dest. Phone Prefix : enter the asterisk (*) symbol to indicate all destinations and callers respectively
Dest.IPGroupID : select "2" to indicate the destination IP Group to where these calls are sent, to the MP-114.
IP Profile ID : enter "2" to indicate the IP Profile configured for "G.729"

Index#2 from MP-114 to AlphaCom XE7

Src.IPGroupID : select "2" to indicate received (inbound) calls identified as belonging to the IP Group configured for MP-114
Dest. Phone Prefix : enter the asterisk (*) symbol to indicate all destinations and callers respectively
Dest.IPGroupID : select "1" to indicate the destination IP Group to where these calls are sent, to the AlphaCom.
IP Profile ID : enter "1" to indicate the IP Profile configured for "G.711µ-law"
Submit & Burn

Configuration of Mediant 600 with SIP Account as End Point

A license SIP to SIP Mediation and Routing Application (SW/IP2IP) is required. This section is additional to section = Mediant 600 Configuration =

Proxy Sets Table

These "Proxy Sets" are later assigned to "IP Groups" Note that the "Proxy Set" represents the actual destination to which the call is routed

In the "Proxy Sets Table" page (Configuration tab -> Protocol Configuration menu -> Proxies,Registration,IP Groups submenu -> Proxy Sets Table page):

Proxy Set ID1

Proxy Set ID#1 for AlpahCom XE

From the "Proxy Set ID" drop-down list, select "1"
In the "Proxy Address" column, enter the IP address of the AlphaCom
From the "Transport Type" drop-down list corresponding to the IP address entered above, select "UDP"
Submit

Proxy Set ID2

Proxy Set ID#2 for SIP Server

From the "Proxy Set ID" drop-down list, select "2"
In the "Proxy Address" column, enter the IP address of the SIP server
From the "Transport Type" drop-down list corresponding to the IP address entered above, select "UDP"
Set Enable Proxy Keep Alive to "Using Register"
Submit & Burn

IP Group Table

These "IP Groups" are later used by the device for routing calls

In the "IP Group Table" page (Configuration tab -> Protocol Configuration menu -> Proxies,Registration,IP Groups submenu -> IP Group Table page):

IP Group 1

IP Group #1 for AlpahCom XE7

From the "Index" drop-down list, select "1"
In the "Description" field, type an arbitrary name for the IP Group (e.g. AlphaCom)
From the "Proxy Set ID" drop-down list, select "1" (For this "IP Group" communicate with the "Proxy Set" of the AlphaCom)
From the "IP Profile ID" drop-down list, select "1" (the IP Profile is configured later)
Submit

IP Group 2

IP Group #2 for SIP-Server

From the "Index" drop-down list, select "2"
In the "Description" field, type an arbitrary name for the IP Group (e.g. 51215382)
From the "Proxy Set ID" drop-down list, select "2" (For this "IP Group" communicate with the "Proxy Set" of the SIPServer)
If more then one Account, repeat the same above with IP Group nr not in use(e.g. IP Group nr 3)
From the "IP Profile ID" drop-down list, select "2". Do the same for all other IP Group associated with SIP Servers IP Groups(the IP Profile is configured later)
Submit & Burn

IP Profiles for Voice Coders

For use transcoding it's necessary create two IP Profiles for define two types of coders used.

These profiles are later used in the "Inbound IP Routing" and "Outbound IP Routing" tables.

In the "Coder Group Settings" page (Configuration tab -> Protocol Configuration menu -> Coders And Profile Definitions submenu -> Coder Group Settings page):

Coder Group 1

Coder Group ID#1 for AlpahCom XE7

From the "Coder Group ID" drop-down list, select "1"
In the "Coder Name" drop-down list, select "G.711U-law" (Coder used by AlphaCom)
Submit

Coder Group 2

Coder Group ID#2 for SIP Server

From the "Coder Group ID" drop-down list, select "2"
In the "Coder Name" drop-down list, select "G.729" (Coder used by SIP Server)
Note: If SIP Provider supports another Codec, please choose from the drop down list.
Submit & Burn

In the "IP Profile Settings" page (Configuration tab -> Protocol Configuration menu -> Coders And Profile Definitions submenu -> IP Profile Settings page):

Profile ID 1

Profile ID#1 for AlpahCom XE7

Enter the Profile name "AlphaCom"
From the "Profile ID" drop-down list, select "1"
From the "Coder Group" drop-down list, select "Coder Group 1"
Submit

Profile ID 2

Profile ID#2 for SIP Server

Enter the Profile name "SIP Server"
From the "Profile ID" drop-down list, select "2"
In the "Coder Group" drop-down list, select "Coder Group 2"
Submit & Burn

Account Table

These "Account Table" should be assigned to "IP Groups" for each account to control which account to call out with.

Note that the "IP Groups" is made for each Account done in earlier step.

In the "Proxy Sets Table" page (Configuration tab -> Control Network menu -> SIP Definitions submenu -> Account Table page):

Account Tables

Account Table for SIP Server

Start line 1 in Add field and Press "Add"
Set Served IP Group to 1
Set Served IP Trunk Group to -1
Set Serving IP Group same as the Serving IP group associated in IP Group used by this account.
Set Username and Password due to information from SIP Provider
From the "Register" drop-down list, select "YES"
Set "Contact User" to SIP ID
From the "Application Type" drop-down list, select "GW\IP2IP"
Submit

Inbound IP Routing

The "IP to Trunk Group Routing Table" it used for define the routing inbound IP-to-IP calls.

The table in which this is configured uses the IP Groups that you defined before.

In the "IP to Trunk Group Routing Table" page (Configuration tab -> Protocol Configuration menu -> Routing Tables submenu -> IP to Trunk Group Routing page):

Inbound IP Routing

Index#1 for AlpahCom XE7

Dest Phone Prefix : enter the asterisk (*) symbol to indicate all destinations
Source Phone Prefix : enter the asterisk (*) symbol to indicate all destinations
Source IP Adress : enter the IP adress of the AlphaCom
Trunk Group ID : enter "-1" to indicate that these calls are IP-to-IP calls
IP Profile ID : enter "1" to assign these calls to "ProfileID#1" to use "G.711U-law"
Source IP Groupe ID : enter "1" to assign these calls to the IP Group pertaining to the AlphaCom

Index#2 for SIP Server

Dest Phone Prefix : enter the asterisk (*) symbol to indicate all destinations
Source Phone Prefix : enter the asterisk (*) symbol to indicate all destinations
Source IP Adress : enter * to receive calls from all URL/IP adresses of the SIP server
Trunk Group ID : enter "-1" to indicate that these calls are IP-to-IP calls
IP Profile ID : enter "2" to assign these calls to "ProfileID#2" to use "G.729"
Source IP Groupe ID : enter "2" to assign these calls to the IP Group pertaining to the SIP Server
Note: Enter Sorce IP Group nr of the every account you register in Accout Table in next step
Submit & Burn

Outbound IP Routing

The "Tel to IP Routing Table" it used for define the routing outbound IP-to-IP calls.

The table in which this is configured uses the IP Groups that you defined before.

In the "Tel to IP Routing" page (Configuration tab -> Protocol Configuration menu -> Routing Tables submenu -> Tel to IP Routing page):

Outbound IP Routing

Index#1 from AlpahCom XE7 to SIPServer

Src.IPGroupID : select "1" to indicate received (inbound) calls identified as belonging to the IP Group configured for AlphaCom
Dest. Phone Prefix : enter the asterisk (*) symbol to indicate all destinations and callers respectively
Dest.IPGroupID : select "2" to indicate the destination IP Group to where these calls are sent, to the SIP-Server
IP Profile ID : enter "2" to indicate the IP Profile configured for "G.729"
Note: Enter a new line with "Source Phone Prefix" set to Extension and "Dest. IP Group ID" to associated SIP account. This to control which SIP Account the each extension is using.

Index#2 from SIP Server to AlphaCom XE7

Src.IPGroupID : select "2" to indicate received (inbound) calls identified as belonging to the IP Group configured for MP-114
Dest. Phone Prefix : enter the asterisk (*) symbol to indicate all destinations and callers respectively
Dest.IPGroupID : select "1" to indicate the destination IP Group to where these calls are sent, to the AlphaCom.
IP Profile ID : enter "1" to indicate the IP Profile configured for "G.711µ-law"
Submit & Burn

Manipulation Table

This step will change the local SIP ID/Dir No to number associated to SIP accounts.

In the "Source Number Tel->IP" page (VoIP tab -> Control Network menu -> GW and IP to IP submenu -> Manipulation page):

Manipulation Tel->IP Calls

Manipulation Table for Outgoing Calls

This step is necessary to route inncoming call to right Extension on AlphaCom side.

In the "Dest Number IP->Tel" page (VoIP tab -> Control Network menu -> GW and IP to IP submenu -> Manipulation page):

Manipulation IP-> Tel Calls

Manipulation Table for route the inncoming call

Submit

AudioCodes MP-114/118 Configuration

All text and pictures refer to version 5.80

For configuration of the MP114/118 please see Configuration guide for AudioCodes MP114/118, v5.4 and higher.

The following highlighted points deviate from the standard MP114/118 configuration and are special for the SIP2SIP transcoding with AlphaCom using AudioCodes Mediant 600:

SIP Parameters

In the Proxy & Registration page (Configuration tab -> Protocol Configuration menu -> Protocol Definition submenu -> Proxy & Registration page item) set the ‘Use Default Proxy’ field to ‘Yes’.

Proxy Name must be blank
Gateway Name must be blank
Click the Proxy Set Table button. In the ‘Proxy Address’ field enter the IP address of the Mediant 600.
Set 'Transport Type' to 'UDP'.

Press Submit to save changes

Audio Codec

Select the voice coder in 'Coders Table' page (Configuration tab -> Protocol Configuration menu -> Protocol Definition submenu -> Coders page item).

From the drop-down list select G.729

Press Submit to save changes

Cisco Call Manager Configuration

All text and pictures refer to version 6.1.2

For configuration of the Cisco Call Manager please see Cisco Call Manager 6 configuration guide.

The following highlighted points deviate from the standard CUCM configuration and are special for the SIP2SIP transcoding with AlphaCom using AudioCodes Mediant 600:

Configure a Region

Region Configuration

System -> Region -> Add New

Name: Zenitel
Press Save
Under Modify Relationships to other Regions select Zenitel.
Select Audio Codec to G.729
Press Save
Press Reset/Restart

Configure a SIP Trunk

Trunk Configuration

Trunk Configuration, continued

Device -> Trunk -> Add New

Trunk Type: SIP Trunk
Device Protocol: SIP
Press Next
Device Name: AlphaCom
Description: AlphaCom
Device Pool: Zenitel
Media Resource Group List: AlphaCom_MRGL
Enable: 'Media Termination Point Required'
Calling Search Space: DefaultUser
Destination Adress: The IP adress of the Mediant 600
SIP Trunk Security Profile: AlphaCom
SIP Profile: Standard SIP Profile
Press Save
Press Reset/Restart

Mobile VoIP GSM Gateway MV-370

All text and pictures refer to version 6.693.t

For configuration of the MV-370 please see Configuration guide for Mobile VoIP.

The following highlighted points deviate from the standard MV-370 configuration and are special for the SIP2SIP transcoding with AlphaCom using AudioCodes Mediant 600:

SIP settings

SIP Settings

In the menu SIP Settings -> Service Domain, enter information for "Realm 1":

Active = ON
User Name = Any text, used for Caller ID. This text will be shown in the display on incoming calls from the GSM network, together with the telephone number
Proxy Server = IP address of the Mediant 600

Status will show Not Registered.

Enable DTMF signalling by SIP INFO method:

SIP Settings > DTMF Setting: Enable Send DTMF SIP Info

Codec settings

Codec Settings

In the menu SIP Settings -> Codec Settings, set codec priority:

Codec Priority 1: = G.729

Leave the rest as is.

Difference between revisions of "SIP2SIP Transcoding using Mediant 600"

From Zenitel Wiki

Latest revision as of 11:39, 15 August 2022

Contents

AlphaCom Configuration

Mediant 600 Configuration

Restore to factory default settings

Access to the embedded web server

IP Configuration

Add Software Upgrade Key

Enable IP-to-IP Capabilities

Number of Media Channels

Proxy Sets Table

IP Group Table

IP Profiles for Voice Coders

IP Profiles for Voice Coders

Inbound IP Routing

Outbound IP Routing

Configuration of Mediant 600 with SIP Account as End Point

Proxy Sets Table

IP Group Table

IP Profiles for Voice Coders

Account Table

Inbound IP Routing

Outbound IP Routing

Manipulation Table

AudioCodes MP-114/118 Configuration

SIP Parameters

Audio Codec

Cisco Call Manager Configuration

Configure a Region

Configure a SIP Trunk

Mobile VoIP GSM Gateway MV-370

SIP settings

Codec settings

@@ Line 1: / Line 1: @@
-The Mediant 600 has the capability to transcode a SIP call from e.g. G711µ-law to G729. In this example the Mediant 600 is used to transcode G711µ-law to G729 between an AlphaCom XE 7 and an MP-114.
+{{A}}
+The Mediant 600 has the capability to transcode a SIP call from e.g. G711µ-law to G729. Here are some examples using the Mediant 600 to transcode G711µ-law to G729.
+The examples shown are between:
+:* AlphaCom XE and Cisco Call Manager
+:* AlphaCom XE and AudioCodes MP-114/118
+:* AlphaCom XE and Mobile VoIP GSM Gateway MV-370
 [[Image:SIP2SIP-M600.PNG|thumb|650px|Configuration example]]
@@ Line 6: / Line 12: @@
 = AlphaCom Configuration =
-== AlphaWeb Configuration ==
-=== Assign IP address to the AlphaCom XE 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:AXE7-IP.PNG|thumb|left|250px|]]
-<br style="clear:both;" />
-=== Insert SIP Trunk licenses ===
-The AlphaCom requires only 1 x SIP Trunk License for this configuration. <br/>
-Log on to AlphaWeb and install the SIP Trunk license.
-[[image:AXE7-InsertLicense.PNG|thumb|left|250px|]]
-<br style="clear:both;" />
-=== Firewall (filter) settings ===
-Enable the SIP protocol on the desired Ethernet port
-(default enabled for Ethernet port1).
-[[image:AXE7-Filters.PNG|thumb|left|250px|]]
-<br style="clear:both;" />
-== 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 shown in the image:
-[[image:Configuration guide for AudioCodes MP114 118 - Create a SIP Trunk Node.jpg|thumb|right|250px|]]
-The SIP Trunk IP address must be identical to the IP address of the SIP
-Gateway (i.e. Mediant 600).
-'''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.
-<br style="clear:both;" />
-=== Create AlphaCom/SIP Audio 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|]]
-<br style="clear:both;" />
-=== 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 ''20 ms''.
-[[image:Configuration guide for AudioCodes MP114 118 - Define the AlphaCom SIP routing.jpg|thumb|left|250px|]]
-<br style="clear:both;" />
-=== 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 “Parameter 2” field (“Collect N more digits (SIP)” in earlier AlphaPro versions) 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 “Parameter 2” field (Collect N more digits (SIP in earlier versions of AlphaPro) are 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.
-[[image:Example of prefix number.jpg|thumb|right|250px|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
-<br style="clear:both;" />
-==== 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 “Parameter 2” field (“Collect N more digits (SIP)” in earlier AlphaPro versions) 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 “Parameter 2” field (“Collect N more digits (SIP)” in earlier AlphaPro versions) are 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.
-[[image:Example of integrated prefix number.jpg|thumb|right|250px|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
-<br style="clear:both;" />
-==== Global number ====
-[[image:Example of global number.jpg|thumb|right|250px|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 “Parameter 2” field (in earlier AlphaPro version “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.
-<br style="clear:both;" />
-=== Update the exchange ===
+How to set up a SIP Trunk in the AlphaCom is described here: [[SIP trunk node - configuration]]
-Log on to the exchange and update the exchange by pressing the
-SendAll button.
 = Mediant 600 Configuration =
@@ Line 134: / Line 20: @@
 == Restore to factory default settings ==
-[[Image:M1K Ext.Comm. ResetButton.png|thumb|350px|Reset Button]]
+[[Image:M1K Ext.Comm. ResetButton.png|left|thumb|500px|Reset Button]]
+<br style="clear:both;" />
 :*With a paper clip, press and hold down the Reset button (located on the CPU Module) for at least 12 seconds (no more than 25 seconds)<br>
 :*The device restores to factory default settings<br>
+== Access to the embedded web server ==
+[[Image:M600-Home.PNG|left|thumb|500px|Home Page]]
 <br style="clear:both;" />
-== Access to the embedded web server ==
-[[Image:M600-Home.PNG|thumb|350px|Home Page]]
 :The default Network parameter of the Mediant 600:
@@ Line 163: / Line 49: @@
 '''''<u>Note:</u>''' For the rest of configuration used the tree menu in '''Full''' mode''
+== IP Configuration ==
+[[Image:AudioCodes Mediant 1000 Configure Network Parameters.jpg|left|thumb|500px|Initial configuration]]
 <br style="clear:both;" />
-== IP Configuration ==
 RETURN TO THIS
-[[Image:AudioCodes Mediant 1000 Configure Network Parameters.jpg|thumb|150px|Initial configuration]]
+[[Image:M1K Ext.Comm. IPConfig.png|left|thumb|500px|IP Settings]]
-[[Image:M1K Ext.Comm. IPConfig.png|thumb|350px|IP Settings]]
+<br style="clear:both;" />
 :In the "IP Settings" page ('''Configuration''' tab -> '''Network Settings''' menu -> '''IP Settings''' page) enter :
@@ Line 176: / Line 62: @@
 ::*Default Gateway
 ::* Click "Submit" to apply the changes
-<br>
 :The IP address is immediately changed when pressing Submit, but it is not permanently stored<br>
 :Without resetting or powering off the device, you need to log on to the Gateway using its new IP address in order to Burn the new IP address to flash:
@@ Line 184: / Line 70: @@
 ::* Click "Burn" to permanently apply the changes
+== Add Software Upgrade Key ==
+The license for SIP IP2IP must be uploaded as a Software Upgrade Key.
+[[Image:M600-InsertLicense.JPG|left|thumb|500px|Software Upgrade Key]]
 <br style="clear:both;" />
-== Add Software Upgrade Key ==
-[[Image:M600-InsertLicense.JPG|thumb|350px|Software Upgrade Key]]
-The license for SIP IP2IP must be uploaded as a Software Upgrade Key.
 :In the "Software Upgrade Key Status" page ('''Management''' tab -> '''Software Update''' menu -> '''Software Upgrade Key''' page): <br/>
 ::*Enter the license key in the "Add a Software Upgrade Key" field
@@ Line 195: / Line 81: @@
 ::*Reset the device (Device Action -> Reset -> Reset)
+== Enable IP-to-IP Capabilities ==
+[[Image:M600-EnableIP2IP.PNG|left|thumb|500px|Applications Enabling]]
 <br style="clear:both;" />
-== Enable IP-to-IP Capabilities ==
-[[Image:M600-EnableIP2IP.PNG|thumb|350px|Applications Enabling]]
 :In the "Applications Enabling" page ('''Configuration''' tab -> '''Protocol Configuration''' menu -> '''Applications Enabling''' page):<br>
@@ Line 206: / Line 90: @@
 ::*Wait the next step for Reset the device
+== Number of Media Channels ==
+[[Image:M600-SetNumberOfMediaChannels.PNG|left|thumb|500px|IP Media Settings]]
 <br style="clear:both;" />
-== Number of Media Channels ==
-[[Image:M600-SetNumberOfMediaChannels.PNG|thumb|350px|IP Media Settings]]
 :In the "IP Media Settings" page ('''Configuration''' tab -> '''Media Settings''' menu -> '''IP Media Settings''' page):<br>
@@ Line 215: / Line 98: @@
 ::*Submit & Burn
 ::*Reset the device (Device Action -> Reset -> Reset)
-<br style="clear:both;" />
 == Proxy Sets Table ==
-[[Image:M600-SetProxy1.PNG|thumb|350px|Proxy Set ID1]]
-[[Image:M600-SetProxy2.PNG|thumb|350px|Proxy Set ID2]]
 These "Proxy Sets" are later assigned to "IP Groups"
 Note that the "Proxy Set" represents the actual destination to which the call is routed
-<br>
-<br>
 :In the "Proxy Sets Table" page ('''Configuration''' tab -> '''Protocol Configuration''' menu -> '''Proxies,Registration,IP Groups''' submenu -> '''Proxy Sets Table''' page):<br>
+[[Image:M600-SetProxy1.PNG|left|thumb|500px|Proxy Set ID1]]
+<br style="clear:both;" />
-:*Proxy Set ID#1 for AlpahCom XE7
+:*Proxy Set ID#1 for AlpahCom XE
 ::*From the "Proxy Set ID" drop-down list, select "1"
 ::*In the "Proxy Address" column, enter the IP address of the AlphaCom
 ::*From the "Transport Type" drop-down list corresponding to the IP address entered above, select "UDP"
 ::*Submit
+----
+[[Image:M600-SetProxy2.PNG|left|thumb|500px|Proxy Set ID2]]
+<br style="clear:both;" />
 :*Proxy Set ID#2 for MP-114
@@ Line 244: / Line 125: @@
 == IP Group Table ==
+These "IP Groups" are later used by the device for routing calls
-These "IP Groups" are later used by the device for routing calls
-<br>
-<br>
 :In the "IP Group Table" page ('''Configuration''' tab -> '''Protocol Configuration''' menu -> '''Proxies,Registration,IP Groups''' submenu -> '''IP Group Table''' page):<br>
-[[Image:M600-SetIPGroup1.PNG|thumb|350px|IP Group 1]]
+[[Image:M600-SetIPGroup1.PNG|left|thumb|500px|IP Group 1]]
-:*IP Group #1 for AlpahCom XE7
+<br style="clear:both;" />
+:*IP Group #1 for AlpahCom XE
 ::*From the "Index" drop-down list, select "1"
 ::*In the "Description" field, type an arbitrary name for the IP Group (e.g. AlphaCom)
 ::*From the "Proxy Set ID" drop-down list, select "1" (For this "IP Group" communicate with the "Proxy Set" of the AlphaCom)
 ::*In the "SIP Group Name" field, enter the IP Adress sent in the SIP Request From/To headers for this IP Group (AlphaCom's IP Adress)
-::*Contact User = name that is sent in the SIP Request contact header for this IP Group (e.g. AXE7)
+::*Contact User = name that is sent in the SIP Request contact header for this IP Group (e.g. AXE)
 ::*From the "IP Profile ID" drop-down list, select "1" (the IP Profile is configured later)
 ::*Submit
+----
+[[Image:M600-SetIPGroup2.PNG|left|thumb|500px|IP Group 2]]
 <br style="clear:both;" />
-[[Image:M600-SetIPGroup2.PNG|thumb|350px|IP Group 2]]
 :*IP Group #2 for MP-114
 ::*From the "Index" drop-down list, select "2"
@@ Line 275: / Line 154: @@
 == IP Profiles for Voice Coders ==
+For use transcoding it's necessary create two IP Profiles for define two types of coders used.
-For use transcoding it's necessary create two IP Profiles for define two types of coders used.<br>
 These profiles are later used in the "Inbound IP Routing" and "Outbound IP Routing" tables.
-<br>
-<br>
 :In the "Coder Group Settings" page ('''Configuration''' tab -> '''Protocol Configuration''' menu -> '''Coders And Profile Definitions''' submenu -> '''Coder Group Settings''' page):<br>
-[[Image:M600-CoderGroup1.PNG|thumb|350px|Coder Group 1]]
+[[Image:M600-CoderGroup1.PNG|left|thumb|500px|Coder Group 1]]
-:*Coder Group ID#1 for AlpahCom XE7
+<br style="clear:both;" />
+:*Coder Group ID#1 for AlphaCom XE
 ::*From the "Coder Group ID" drop-down list, select "1"
 ::*In the "Coder Name" drop-down list, select "G.711U-law" (Coder used by AlphaCom)
 ::*Submit
 <br style="clear:both;" />
+----
-[[Image:M600-CoderGroup2.PNG|thumb|350px|Coder Group 2]]
+[[Image:M600-CoderGroup2.PNG|left|thumb|500px|Coder Group 2]]
+<br style="clear:both;" />
 :*Coder Group ID#2 for MP-114
 ::*From the "Coder Group ID" drop-down list, select "2"
@@ Line 296: / Line 177: @@
 <br style="clear:both;" />
+----
+:In the "IP Profile Settings" page ('''Configuration''' tab -> '''Protocol Configuration''' menu -> '''Coders And Profile Definitions''' submenu -> '''IP Profile Settings''' page):<br>
+[[Image:M600-IPProfile1.PNG|left|thumb|500px|Profile ID 1]]
+<br style="clear:both;" />
+:*Profile ID#1 for AlpahCom XE
+::*From the "Profile ID" drop-down list, select "1"
+::*From the "Coder Group" drop-down list, select "Coder Group 1"
+::*Submit
+<br style="clear:both;" />
+----
+[[Image:M600-IPProfile2.PNG|left|thumb|500px|Profile ID 2]]
+<br style="clear:both;" />
+:*Profile ID#2 for MP-114
+::*From the "Profile ID" drop-down list, select "2"
+::*In the "Coder Group" drop-down list, select "Coder Group 2"
+::*Submit & Burn
+<br style="clear:both;" />
+== IP Profiles for Voice Coders ==
+For use transcoding it's necessary create two IP Profiles for define two types of coders used.
+These profiles are later used in the "Inbound IP Routing" and "Outbound IP Routing" tables.
+:In the "Coder Group Settings" page ('''Configuration''' tab -> '''Protocol Configuration''' menu -> '''Coders And Profile Definitions''' submenu -> '''Coder Group Settings''' page):<br>
+[[Image:CoderGroupAlphaCom.jpg|left|thumb|500px|Coder Group 1]]
+<br style="clear:both;" />
+:*Coder Group ID#1 for AlpahCom XE
+::*From the "Coder Group ID" drop-down list, select "1"
+::*In the "Coder Name" drop-down list, select "G.711U-law" (Coder used by AlphaCom)
+::*Submit
+<br style="clear:both;" />
+----
+[[Image:CoderGroupSIPServer.jpg|left|thumb|500px|Coder Group 2]]
+<br style="clear:both;" />
+:*Coder Group ID#2 for MP-114
+::*From the "Coder Group ID" drop-down list, select "2"
+::*In the "Coder Name" drop-down list, select "G.729" (Coder used by SIP Server)
+::*Submit & Burn
+<br style="clear:both;" />
+----
 :In the "IP Profile Settings" page ('''Configuration''' tab -> '''Protocol Configuration''' menu -> '''Coders And Profile Definitions''' submenu -> '''IP Profile Settings''' page):<br>
-[[Image:M600-IPProfile1.PNG|thumb|350px|Profile ID 1]]
+[[Image:M600-IPProfile1.PNG|left|thumb|500px|Profile ID 1]]
-:*Profile ID#1 for AlpahCom XE7
+<br style="clear:both;" />
+:*Profile ID#1 for AlpahCom XE
 ::*From the "Profile ID" drop-down list, select "1"
 ::*From the "Coder Group" drop-down list, select "Coder Group 1"
@@ Line 306: / Line 233: @@
 <br style="clear:both;" />
+----
-[[Image:M600-IPProfile2.PNG|thumb|350px|Profile ID 2]]
+[[Image:M600-IPProfile2.PNG|left|thumb|500px|Profile ID 2]]
+<br style="clear:both;" />
 :*Profile ID#2 for MP-114
 ::*From the "Profile ID" drop-down list, select "2"
@@ Line 316: / Line 244: @@
 == Inbound IP Routing ==
+The "IP to Trunk Group Routing Table" it used for define the routing inbound IP-to-IP calls.
-The "IP to Trunk Group Routing Table" it used for define the routing inbound IP-to-IP calls.<br>
 The table in which this is configured uses the IP Groups that you defined before.
-<br>
-<br>
 :In the "IP to Trunk Group Routing Table" page ('''Configuration''' tab -> '''Protocol Configuration''' menu -> '''Routing Tables''' submenu -> '''IP to Trunk Group Routing''' page):<br>
-[[Image:M600-RoutingIPTrunkGroup.PNG|thumb|350px|Inbound IP Routing]]
+[[Image:M600-RoutingIPTrunkGroup.PNG|left|thumb|500px|Inbound IP Routing]]
+<br style="clear:both;" />
 :*Index#1 for AlpahCom XE7
@@ Line 345: / Line 274: @@
 == Outbound IP Routing ==
+The "Tel to IP Routing Table" it used for define the routing outbound IP-to-IP calls.
-The "Tel to IP Routing Table" it used for define the routing outbound IP-to-IP calls.<br>
 The table in which this is configured uses the IP Groups that you defined before.
-<br>
-<br>
 :In the "Tel to IP Routing" page ('''Configuration''' tab -> '''Protocol Configuration''' menu -> '''Routing Tables''' submenu -> '''Tel to IP Routing''' page):<br>
-[[Image:M600-RoutingTelToIP.PNG|thumb|350px|Outbound IP Routing]]
+[[Image:M600-RoutingTelToIP.PNG|left|thumb|500px|Outbound IP Routing]]
+<br style="clear:both;" />
 :*Index#1 from AlpahCom XE7 to MP-114
@@ Line 370: / Line 300: @@
 <br style="clear:both;" />
-= AudioCodes MP-114 Configuration =
+= Configuration of Mediant 600 with SIP Account as End Point =
+A license SIP to SIP Mediation and Routing Application (SW/IP2IP) is required.
+This section is additional to section = Mediant 600 Configuration =
-== Configure Network Parameters ==
+== Proxy Sets Table ==
-The AudioCodes MP-114/118 VoIP Gateway comes with default network parameters (factory default parameters).
+These "Proxy Sets" are later assigned to "IP Groups"
+Note that the "Proxy Set" represents the actual destination to which the call is routed
-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. Consult your network administrator to obtain a valid IP address.
+:In the "Proxy Sets Table" page ('''Configuration''' tab -> '''Protocol Configuration''' menu -> '''Proxies,Registration,IP Groups''' submenu -> '''Proxy Sets Table''' page):
+[[Image:ProxySet1.JPG|left|thumb|500px|Proxy Set ID1]]
+<br style="clear:both;" />
-The Home page of the Web Interface:
+:*Proxy Set ID#1 for AlpahCom XE
-[[Image:Configure Network Parameters 3.jpg|left|700px]]
+::*From the "Proxy Set ID" drop-down list, select "1"
+::*In the "Proxy Address" column, enter the IP address of the AlphaCom
+::*From the "Transport Type" drop-down list corresponding to the IP address entered above, select "UDP"
+::*Submit
+----
+[[Image:ProxySet2.JPG|left|thumb|500px|Proxy Set ID2]]
+<br style="clear:both;" />
+:*Proxy Set ID#2 for SIP Server
+::*From the "Proxy Set ID" drop-down list, select "2"
+::*In the "Proxy Address" column, enter the IP address of the SIP server
+::*From the "Transport Type" drop-down list corresponding to the IP address entered above, select "UDP"
+::*Set Enable Proxy Keep Alive to "Using Register"
+::*Submit & Burn
 <br style="clear:both;" />
-=== Restoring factory defaults ===
+== IP Group Table ==
-Follow these steps:
+These "IP Groups" are later used by the device for routing calls
-[[Image:Configure Network Parameters 1.jpg|100px|thumb]]
-[[Image:Configure Network Parameters 2.jpg|100px|thumb]]
-:*Load factory network parameters and reset the username and password to its default settings:
-::*Disconnect the Ethernet cable from the device
-::*With a paper clip or any other similar pointed object, press and hold down the Reset button (located on the rear panel) for about six seconds; the Fail LED turns red and the device restores to factory default settings
-::*When the Fail LED turns off, reconnect the Ethernet cable to the device
-:*The VoIP Gateway will now have the IP address 10.1.10.11, subnet mask 255.255.0.0.
+:In the "IP Group Table" page ('''Configuration''' tab -> '''Protocol Configuration''' menu -> '''Proxies,Registration,IP Groups''' submenu -> '''IP Group Table''' page):
-:*Change the IP address of your PC to 10.1.10.12, subnet mask 255.255.0.0.
-:*Connect the LAN port of the PC to the Ethernet port of the Gateway. Use a crossover cable or connect the PC and the VoIP Gateway to a common switch using straight cables
-::*Start your Web Browser and type 10.1.10.11 in the URL field
-::*Type in user name '''Admin''' and password '''Admin''' (Case-sensitive!)
+[[Image:IPGrouptable-AlphaCom.jpg|left|thumb|500px|IP Group 1]]
 <br style="clear:both;" />
+:*IP Group #1 for AlpahCom XE7
+::*From the "Index" drop-down list, select "1"
+::*In the "Description" field, type an arbitrary name for the IP Group (e.g. AlphaCom)
+::*From the "Proxy Set ID" drop-down list, select "1" (For this "IP Group" communicate with the "Proxy Set" of the AlphaCom)
+::*From the "IP Profile ID" drop-down list, select "1" (the IP Profile is configured later)
+::*Submit
-=== IP Configuration ===
+<br style="clear:both;" />
-:In the 'IP Settings' page ('''Configuration''' tab > '''Network Settings''' menu > '''IP Settings''' page item):
+----
-:*Enter the IP Address, Subnet Mask and optionally the Default Gateway Address of the AudioCodes Gateway
+[[Image:IPGrouptable-SIPserverAccount.jpg|left|thumb|500px|IP Group 2]]
-:*Click Submit to apply the changes
+<br style="clear:both;" />
+:*IP Group #2 for SIP-Server
+::*From the "Index" drop-down list, select "2"
+::*In the "Description" field, type an arbitrary name for the IP Group (e.g. 51215382)
+::*From the "Proxy Set ID" drop-down list, select "2" (For this "IP Group" communicate with the "Proxy Set" of the SIPServer)
+::*If more then one Account, repeat the same above with IP Group nr not in use(e.g. IP Group nr 3)
+::*From the "IP Profile ID" drop-down list, select "2". Do the same for all other IP Group associated with SIP Servers IP Groups(the IP Profile is configured later)
+::*Submit & Burn
-[[Image:IP Configuration.jpg|600px]]
+<br style="clear:both;" />
-'''Note''': The IP address is immediately changed when pressing Submit, but it is not permanently stored. Without resetting or powering off the device, you need to log on to the Gateway using its new IP address in order to Burn the new IP address to flash:
+== IP Profiles for Voice Coders ==
+For use transcoding it's necessary create two IP Profiles for define two types of coders used.
-:*Disconnect the PC from the Gateway.
+These profiles are later used in the "Inbound IP Routing" and "Outbound IP Routing" tables.
-:*Reconnect the Gateway and PC to the LAN. The PC and Gateway must be on the same subnet.
-:*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.
-:*Click '''Burn''' to permanently apply the changes.
-=== SIP Parameters ===
+:In the "Coder Group Settings" page ('''Configuration''' tab -> '''Protocol Configuration''' menu -> '''Coders And Profile Definitions''' submenu -> '''Coder Group Settings''' page):
-:In the 'Proxy & Registration' page ('''Configuration''' tab > '''Protocol Configuration''' menu > '''Protocol Definition''' submenu > '''Proxy & Registration''' page item) set the ‘Use Default Proxy’ field to ‘Yes’.
+[[Image:CoderGroupAlphaCom.jpg|left|thumb|500px|Coder Group 1]]
-:*'Proxy Name' '''''must''''' be blank
+<br style="clear:both;" />
-:*'Gateway Name' '''''must''''' be blank
+:*Coder Group ID#1 for AlpahCom XE7
-:*Click the Proxy Set Table button. In the ‘Proxy Address’ field enter the IP address of the Mediant 600.
+::*From the "Coder Group ID" drop-down list, select "1"
-:*Set 'Transport Type' to 'UDP'.
+::*In the "Coder Name" drop-down list, select "G.711U-law" (Coder used by AlphaCom)
+::*Submit
+<br style="clear:both;" />
+----
+[[Image:CoderGroupSIPServer.jpg|left|thumb|500px|Coder Group 2]]
+<br style="clear:both;" />
+:*Coder Group ID#2 for SIP Server
+::*From the "Coder Group ID" drop-down list, select "2"
+::*In the "Coder Name" drop-down list, select "G.729" (Coder used by SIP Server)
+::*Note: If SIP Provider supports another Codec, please choose from the drop down list.
+::*Submit & Burn
+<br style="clear:both;" />
+----
+:In the "IP Profile Settings" page ('''Configuration''' tab -> '''Protocol Configuration''' menu -> '''Coders And Profile Definitions''' submenu -> '''IP Profile Settings''' page):<br>
-[[Image:SIP Parameters.jpg|600px]]
+[[Image:IPProfileAlphaCom.jpg|left|thumb|500px|Profile ID 1]]
+<br style="clear:both;" />
+:*Profile ID#1 for AlpahCom XE7
+::*Enter the Profile name "AlphaCom"
+::*From the "Profile ID" drop-down list, select "1"
+::*From the "Coder Group" drop-down list, select "Coder Group 1"
+::*Submit
+<br style="clear:both;" />
+----
+[[Image:IPProfileSIPServer.jpg|left|thumb|500px|Profile ID 2]]
+<br style="clear:both;" />
+:*Profile ID#2 for SIP Server
+::*Enter the Profile name "SIP Server"
+::*From the "Profile ID" drop-down list, select "2"
+::*In the "Coder Group" drop-down list, select "Coder Group 2"
+::*Submit & Burn
+<br style="clear:both;" />
-:*Press Submit to save changes
+== Account Table ==
+These "Account Table" should be assigned to "IP Groups" for each account to control which account to call out with.
-=== Audio Codec ===
+Note that the "IP Groups" is made for each Account done in earlier step.
-:Select the voice coder in 'Coders Table' page ('''Configuration''' tab > '''Protocol Configuration''' menu > '''Protocol Definition''' submenu > Coders page item).
-:*From the drop-down list select G.729
-[[Image:MP114-Coders.JPG|600px]]
+:In the "Proxy Sets Table" page ('''Configuration''' tab -> '''Control Network''' menu -> '''SIP Definitions''' submenu -> '''Account Table''' page):
-:*Press Submit to save changes
+[[Image:AccountTable.jpg|left|thumb|500px|Account Tables]]
+<br style="clear:both;" />
+:*Account Table for SIP Server
+::*Start line 1 in Add field and Press "Add"
+::*Set Served IP Group to 1
+::*Set Served IP Trunk Group to -1
+::*Set Serving IP Group same as the Serving IP group associated in IP Group used by this account.
+::*Set Username and Password due to information from SIP Provider
+::*From the "Register" drop-down list, select "YES"
+::*Set "Contact User" to SIP ID
+::*From the "Application Type" drop-down list, select "GW\IP2IP"
+::*Submit
+<br style="clear:both;" />
+== Inbound IP Routing ==
+The "IP to Trunk Group Routing Table" it used for define the routing inbound IP-to-IP calls.
-=== About Saving Changes ===
+The table in which this is configured uses the IP Groups that you defined before.
-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.
+:In the "IP to Trunk Group Routing Table" page ('''Configuration''' tab -> '''Protocol Configuration''' menu -> '''Routing Tables''' submenu -> '''IP to Trunk Group Routing''' page):
+[[Image:IP-to-Tel-Routing.jpg|left|thumb|500px|Inbound IP Routing]]
+<br style="clear:both;" />
-To permanently save the configuration data, store the data to flash memory by selecting '''Burn''' from the Tool Bar.
+:*Index#1 for AlpahCom XE7
+::*Dest Phone Prefix : enter the asterisk (*) symbol to indicate all destinations
+::*Source Phone Prefix : enter the asterisk (*) symbol to indicate all destinations
+::*Source IP Adress : enter the IP adress of the AlphaCom
+::*Trunk Group ID : enter "-1" to indicate that these calls are IP-to-IP calls
+::*IP Profile ID : enter "1" to assign these calls to "ProfileID#1" to use "G.711U-law"
+::*Source IP Groupe ID : enter "1" to assign these calls to the IP Group pertaining to the AlphaCom
-'''Note''': Parameters proceeded by a yellow lightning symbol is not changeable on-the-fly and require that the device is reset.
+:*Index#2 for SIP Server
+::*Dest Phone Prefix : enter the asterisk (*) symbol to indicate all destinations
+::*Source Phone Prefix : enter the asterisk (*) symbol to indicate all destinations
+::*Source IP Adress : enter * to receive calls from all URL/IP adresses of the SIP server
+::*Trunk Group ID : enter "-1" to indicate that these calls are IP-to-IP calls
+::*IP Profile ID : enter "2" to assign these calls to "ProfileID#2" to use "G.729"
+::*Source IP Groupe ID : enter "2" to assign these calls to the IP Group pertaining to the SIP Server
+::*Note: Enter Sorce IP Group nr of the every account you register in Accout Table in next step
+::*Submit & Burn
+<br style="clear:both;" />
-=== Backup and Restore ===
+== Outbound IP Routing ==
+The "Tel to IP Routing Table" it used for define the routing outbound IP-to-IP calls.
-The configuration of the AudioCodes Gateway can be stored to a file on your PC. <br>
+The table in which this is configured uses the IP Groups that you defined before.
-From the Tool Bar select '''Device Actions -> Save Configuration File.''' Select '''Save INI File''' to save the configuration to the PC, and select '''Load INI File''' to upload a configuration file to the SIP Gateway.
-=== AlphaCom to Telephone Network ===
+:In the "Tel to IP Routing" page ('''Configuration''' tab -> '''Protocol Configuration''' menu -> '''Routing Tables''' submenu -> '''Tel to IP Routing''' page):
+[[Image:Tel-to-IP-Routing.jpg|left|thumb|500px|Outbound IP Routing]]
+<br style="clear:both;" />
+:*Index#1 from AlpahCom XE7 to SIPServer
+::*Src.IPGroupID : select "1" to indicate received (inbound) calls identified as belonging to the IP Group configured for AlphaCom
+::*Dest. Phone Prefix : enter the asterisk (*) symbol to indicate all destinations and callers respectively
+::*Dest.IPGroupID : select "2" to indicate the destination IP Group to where these calls are sent, to the SIP-Server
+::*IP Profile ID : enter "2" to indicate the IP Profile configured for "G.729"
+::*Note: Enter a new line with "Source Phone Prefix" set to Extension and "Dest. IP Group ID" to associated SIP account. This to control which SIP Account the each extension is using.
-There are two ways of selecting a FXO line from the AlphaCom.
+:*Index#2 from SIP Server to AlphaCom XE7
-* '''Group Hunt''', where a prefix is dialed and you are connected to one out of several lines
+::*Src.IPGroupID : select "2" to indicate received (inbound) calls identified as belonging to the IP Group configured for MP-114
-* '''Direct FXO line selection''', where there is one prefix assigned for each of the FXO lines.
+::*Dest. Phone Prefix : enter the asterisk (*) symbol to indicate all destinations and callers respectively
-The two methods can be combined.
+::*Dest.IPGroupID : select "1" to indicate the destination IP Group to where these calls are sent, to the AlphaCom.
+::*IP Profile ID : enter "1" to indicate the IP Profile configured for "G.711µ-law"
+::*Submit & Burn
-==== Group Hunt ====
+<br style="clear:both;" />
-Dial a prefix and get connected to a free FXO line.<br>
+== Manipulation Table ==
-'''''Hunt Group Settings'''''
+This step will change the local SIP ID/Dir No to number associated to SIP accounts.
-:'''Configuration''' tab -> '''Protocol Configuration''' menu > '''Hunt/IP Group''' submenu -> '''Hunt Group Settings''' page item
-:*In the 'Hunt Group Settings' page set the ‘Hunt Group ID’ field to ‘1’ and ‘Channel Select Mode’ to ‘Cyclic Ascending’
-[[Image:Group Hunt.jpg|600px]]
+:In the "Source Number Tel->IP" page ('''VoIP''' tab -> '''Control Network''' menu -> '''GW and IP to IP''' submenu -> '''Manipulation''' page):
+[[Image:Manipulation-table-Outgoing-call.jpg|left|thumb|500px|Manipulation Tel->IP Calls]]
+<br style="clear:both;" />
+:*Manipulation Table for Outgoing Calls
-'''''IP to Hunt Group Routing''''' <br/>
+----
-:When dialing the prefix from AlphaCom, the call needs to be routed to the appropriate Hunt Group ID associated with the FXO ports
-:'''Configuration''' tab -> '''Protocol Configuration''' menu -> '''Routing Tables''' submenu -> '''IP to Trunk Group Routing''' page item
-:*In the example below the call is routed to group hunt ID 1
-[[Image:IP to Hunt Group Routing.jpg|600px]]
-'''''Endpoint Phone Number'''''<br/>
+This step is necessary to route inncoming call to right Extension on AlphaCom side.
-:'''Configuration''' tab -> '''Protocol Configuration''' menu -> '''Endpoint Number''' submenu -> '''EndPoint Phone Number''' page item
-:*In the ‘Endpoint Phone Number Table’ page the FXO lines are linked to the prefix in AlphaCom and to the hunt group ID
+:In the "Dest Number IP->Tel" page ('''VoIP''' tab -> '''Control Network''' menu -> '''GW and IP to IP''' submenu -> '''Manipulation''' page):<br>
+[[Image:Manipulation-table-inncoming-call.jpg|left|thumb|500px|Manipulation IP-> Tel Calls]]
+<br style="clear:both;" />
+:*Manipulation Table for route the inncoming call
+::*Submit
+<br style="clear:both;" />
-In the example below all four FXO lines belong to Hunt Group ID 1. When dialing 0 on an intercom station the first available line will be granted. Directory number 0 must be programmed in the '''AlphaCom directory table''' with feature 83/<node>. See paragraph 2.2.3.
+= AudioCodes MP-114/118 Configuration =
+''All text and pictures refer to version 5.80''
+<br/>
+<br/>
+For configuration of the MP114/118 please see [[Configuration guide for AudioCodes MP114/118, v5.4 and higher]].
-[[Image:Endpoint Phone Number.jpg|600px]]
+The following highlighted points deviate from the standard MP114/118 configuration and are special for the SIP2SIP transcoding with AlphaCom using AudioCodes Mediant 600:
-If there are unused lines, leave the fields for that line blank.
+== SIP Parameters ==
-==== FXO Line Select ====
+:In the Proxy & Registration page ('''Configuration''' tab -> '''Protocol Configuration''' menu -> '''Protocol Definition''' submenu -> '''Proxy & Registration''' page item) set the ‘Use Default Proxy’ field to ‘Yes’.
-'''NOT TESTED'''
+:*Proxy Name ''must'' be blank
+:*Gateway Name ''must'' be blank
+:*Click the Proxy Set Table button. In the '''‘Proxy Address’''' field enter the IP address of the Mediant 600.
+:*Set 'Transport Type' to 'UDP'.
-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 Satelitte lines. Line selection is achieved by assigning each port a unique '''Phone Number''' in the '''Endpoint Phone Number Table.''' These directory numbers must be programmed in the '''AlphaCom directory table''' with feature 83/<node>. See paragraph 2.2.3.
+[[Image:SIP Parameters.jpg|600px]]
-<br>
-Replace “0” with four lines, “41” to “44”.
+:*Press Submit to save changes
-[[Image:FXO Line Select.jpg|600px]]
+== Audio Codec ==
+:Select the voice coder in 'Coders Table' page ('''Configuration''' tab -> '''Protocol Configuration''' menu -> '''Protocol Definition''' submenu -> Coders page item).
+:*From the drop-down list select '''G.729'''
-In this table the four FXO lines are selected by dialing 41 – 44
+[[Image:MP114-Coders.JPG|600px]]
-If there are unused lines, leave all fields for that line blank.
+:*Press Submit to save changes
-Group hunt is not used in this call mode, and the '''IP to Hunt Routing Table''' must be empty.
-[[Image:FXO Line Select2.jpg|600px]]
+= Cisco Call Manager Configuration =
+''All text and pictures refer to version 6.1.2''
+<br/>
+<br/>
+For configuration of the Cisco Call Manager please see [[Cisco Call Manager 6 configuration guide]].
-===Telephone Network to Alpha Com===
+The following highlighted points deviate from the standard CUCM configuration and are special for the SIP2SIP transcoding with AlphaCom using AudioCodes Mediant 600:
-'''NOT TESTED'''
-You can choose between three different ways of handling an incoming call from the telephone line:
+== Configure a Region ==
-* Selective Dialing
+[[Image:SIP2SIP-Cisco-Region.PNG|left|500px|thumb|Region Configuration]]
-* Automatic Dialing
+<br style="clear:both;" />
-* Delayed Automatic Dialing
+:System -> Region -> Add New
+::*Name: Zenitel
+::*Press Save
+::*Under '''Modify Relationships to other Regions''' select Zenitel.
+::*Select Audio Codec to '''G.729'''
+::*Press Save
+::*Press Reset/Restart
+<br style="clear:both;" />
+== Configure a SIP Trunk ==
+[[Image:CiscoSipTrunk 1.PNG|left|thumb|500px|Trunk Configuration]]
+<br style="clear:both;" />
+[[Image:CiscoSipTrunk 2.PNG|left|thumb|500px|Trunk Configuration, continued]]
+<br style="clear:both;" />
+[[Image:SIP2SIP-Cisco-SIPTrunk.PNG|left|thumb|500px|Trunk Configuration, continued]]
+<br style="clear:both;" />
+:Device -> Trunk -> Add New
+::*Trunk Type: SIP Trunk
+::*Device Protocol: SIP
+::*Press Next
+::*Device Name: AlphaCom
+::*Description: AlphaCom
+::*Device Pool: Zenitel
+::*Media Resource Group List: AlphaCom_MRGL
+::*Enable: 'Media Termination Point Required'
+::*Calling Search Space: DefaultUser
+::*Destination Adress: '''The IP adress of the Mediant 600'''
+::*SIP Trunk Security Profile: AlphaCom
+::*SIP Profile: Standard SIP Profile
+::*Press Save
+::*Press Reset/Restart
-==== Selective Dialing ====
+<br style="clear:both;" />
-'''NOT TESTED'''
-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’ page item must be left blank (Default setting).
+= Mobile VoIP GSM Gateway MV-370 =
-('''Configuration''' tab > '''Protocol Configuration''' menu > '''Endpoint Settings''' submenu > '''Automatic Dialing''' page item).
+''All text and pictures refer to version 6.693.t''
+<br/>
+<br/>
+For configuration of the MV-370 please see [[Configuration_guide_for_Mobile_VoIP|Configuration guide for Mobile VoIP]].
-[[Image:Selective Dialing.jpg|600px]]
+The following highlighted points deviate from the standard MV-370 configuration and are special for the SIP2SIP transcoding with AlphaCom using AudioCodes Mediant 600:
-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.
+== SIP settings ==
-<br>
+[[Image:MV-370-SIP.PNG|left|thumb|500px|SIP Settings]]
+<br style="clear:both;" />
-In the 'DTMF & Dialing' page ('''Configuration''' tab > '''Protocol Configuration''' menu > '''Protocol Definition''' submenu > '''DTMF & Dialing''' page item) select ‘Advanced Parameter List’ in order to view all parameters, and 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 for Overlap Dialing’ specifies the waiting time for more digits before setting up the call.
+In the menu '''SIP Settings -> Service Domain''', enter information for "Realm 1":
+:*'''Active''' = ON
-[[Image:Selective Dialing2.jpg|600px]]
+:*'''User Name''' = Any text, used for Caller ID. This text will be shown in the display on incoming calls from the GSM network, together with the telephone number
+:*'''Proxy Server''' = IP address of the Mediant 600
-==== Automatic Dialing (Call to Switchboard) ====
-'''NOT TESTED'''
-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’ page item. Set ‘Auto Dial Status’ to ‘Enable’. ('''Configuration''' tab > '''Protocol Configuration''' menu > '''Endpoint Settings''' submenu > '''Automatic Dialing''' page item).
-[[Image:Automatic Dialing (Call to Switchboard).jpg|600px]]
-In the example above, incoming calls on line 1 are routed to station 103, calls on line 2 are routed to station 101, and calls on line 3 and 4 are routed to RingingGroup 6701.
-==== Delayed Automatic Dialing ====
-'''NOT TESTED'''
-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:Delayed Automatic Dialing.jpg|600px]]
-The ‘'''Hotline Dial Tone Duration'''’ can be changed from the 'DTMF & Dialing' page item ('''Configuration''' tab > '''Protocol''' '''Configuration''' menu > '''Protocol Definition''' submenu > '''DTMF & Dialing''' page item). Select the ‘Advanced Parameter List’. The default value is 16 seconds.
-==== Caller ID ====
+'''Status''' will show ''Not Registered''.
-'''NOT TESTED'''
-Use the ‘Caller Display Information’ page to send display information to the intercom station that receives the call. ('''Configuration''' tab > '''Protocol Configuration''' menu > '''Endpoint Settings''' submenu > '''Caller Display Information''' page item).
+Enable DTMF signalling by SIP INFO method:
+:*'''SIP Settings > DTMF Setting''': Enable ''Send DTMF SIP Info''
-[[Image:Caller ID.jpg|600px]]
+== Codec settings ==
+[[Image:MV-370-Codec.PNG|left|thumb|500px|Codec Settings]]
-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. Display the Navigation Tree in '''Full''' View. Caller ID from FXO line must be enabled in '''Configuration''' tab > '''Protocol Configuration''' menu > '''SIP Advanced Parameters''' submenu > '''Supplementary Services''' page item.
-* Set ‘'''Enable Caller ID'''’ to ‘Enable’ and choose the ‘Caller ID Type’ as used by the PSTN supplier. Check with the local telephone company to find the ‘Caller ID Type’ used.
-== Audio Codec ==
-[[image:Configuration guide for AudioCodes MP114 118 - Audio Codec.jpg|thumb|right|250px|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.
 <br style="clear:both;" />
+In the menu '''SIP Settings -> Codec Settings''', set codec priority:
+:*'''Codec Priority 1:''' = G.729
-== About Saving Changes ==
+Leave the rest as is.
-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.
-[[image:Mediant 1000 - Trunk Settings.jpg|thumb|left|180px|Mediant 1000 - Trunk Settings]]
-<br>
-<br>
-<br>
-<br>
-<br>
-<br>
-<br>
-<br>
-<br>
-<br>
-<br>
-=== 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:
-[[image:Mediant 1000 - Trunk Group Table.jpg|thumb|left|250px|Mediant 1000 - Trunk Group Table]]
-<br>
-<br>
-<br>
-<br>
-<br>
-<br>
-<br>
-=== Trunk Group Settings ===
-In '''Protocol Management > Trunk Group Settings''' you assign the rules for channel allocation:
-[[image:Mediant 1000 - Trunk Group Settings.jpg|thumb|left|250px|Mediant 1000 - Trunk Group settings]]
-<br>
-<br>
-<br>
-<br>
-<br>
-<br>
-<br>
-<br>
-<br>
-=== 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.
-[[Image:Mediant 1000 - IP to Trunk Group Routing Table.jpg|thumb|left|250px|IP to Trunk Group Routing Table]]
-<br>
-<br>
-<br>
-<br>
-<br>
-<br>
-<br>
-== 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 =
-''' NOT TESTED '''
-== 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'').
-: From AlphaPro 10.05: Check the flag '''Private Ringing from SIP''' in Exchange & System > System > Calls and Options.
-: From 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'''.
-[[image:Configuration guide for AudioCodes MP114 118 - SIP Gateway configuration.jpg|thumb|left|250px|]]
-<br>
-<br>
-<br>
-<br>
-<br>
-<br>
-<br>
-=== AlphaCom configuration ===
-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:
-[[image:Configuration guide for AudioCodes MP114 118 - AlphaCom configuration.jpg|thumb|left|180px|Door Opening when SIP calls the door]]
-<u>'''''Calling from the telephone to the door'''''</u><br>
-The Standard door opening event is used.
-<br><br><br><br><br><br><br><br><br>
-[[image:Configuration guide for AudioCodes MP114 118 - AlphaCom configuration (2).jpg|thumb|left|180px|Door Opening when door calls out on SIP gateway]]
-<u>'''''Calling from the door to the telephone'''''</u><br>
-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.
-<br><br><br><br><br><br><br><br><br>
-== 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 to overcome this problem.
-In AlphaPro, go to Exchange & System > System > Voip. Set the flag '''Optimized voice duplex control when conversation with SIP trunk/stations'''. This flag is avaliable from AMC 10.05 and AlphaPro 10.26.
-When the 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)'''
-== Feature Guide ==
-* The telephone line is accessed either by dialing a:
-** Prefix number: Dial Prefix + Phone number. Use M-key as a “Send” button, or wait for the number to be sent automatically.
-** Global number: Dial the phone number without using prefix
-* 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> M'''. Example: I 0 P 40002500 M
-=== 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 ===
-* An intercom call can be forwarded to a telephone
-** From keypad: '''71 + 0 + <phone number> + M'''
-** From preprogrammed DAK: '''I 71 I 0 P 40002500'''
-=== Search List ===
-* A telephone number can be included in the Search List of a station.
-** Format: '''I <prefix> P <phone number>'''. Example: I 0 P 40002500
-[[Category:SIP]]
+[[Category:AlphaCom - SIP Integration]]
 [[Category:3rd party integration]]