View source for SIP2SIP Transcoding using Mediant 600

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]]


= 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 ==
[[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;" />

: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
<br>
:Connect your PC directly to the device, using an Ethernet Crossover cable
<br>
: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)
<br>
:To access the embedded web server, start your internet browser (e.g. Internet Explorer) and in the address field enter 10.1.10.10
<br>
:You will be prompted for a username and password (default): 
:*Username: Admin
:*Password: Admin
<br>
:The Home Page page should now be displayed
<br>
'''''<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;" />

RETURN TO THIS
[[Image:M1K Ext.Comm. IPConfig.png|left|thumb|500px|IP Settings]]
<br style="clear:both;" />

: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<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:
::* 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.
[[Image:M600-InsertLicense.JPG|left|thumb|500px|Software Upgrade Key]]
<br style="clear:both;" />

: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
::*Click "Add Key"
::*Burn
::*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;" />

:In the "Applications Enabling" page ('''Configuration''' tab -> '''Protocol Configuration''' menu -> '''Applications Enabling''' page):<br>
::*From the "Enable IP2IP Application" drop-down list, select "Enable"
::*Submit & Burn
::*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;" />

:In the "IP Media Settings" page ('''Configuration''' tab -> '''Media Settings''' menu -> '''IP Media Settings''' page):<br>
::*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):<br>
[[Image:M600-SetProxy1.PNG|left|thumb|500px|Proxy Set ID1]]
<br style="clear:both;" />

:*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
::*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

<br style="clear:both;" />

== 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):<br>

[[Image:M600-SetIPGroup1.PNG|left|thumb|500px|IP Group 1]]
<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. 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;" />
:*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

<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:M600-CoderGroup1.PNG|left|thumb|500px|Coder Group 1]]
<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|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 MP-114)
::*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|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 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
<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|left|thumb|500px|Profile ID 1]]
<br style="clear:both;" />
:*Profile ID#1 for AlpahCom XE7
::*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;" />

== 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):<br>

[[Image:M600-RoutingIPTrunkGroup.PNG|left|thumb|500px|Inbound IP Routing]]
<br style="clear:both;" />

:*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

<br style="clear:both;" />

== 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):<br>

[[Image:M600-RoutingTelToIP.PNG|left|thumb|500px|Outbound IP Routing]]
<br style="clear:both;" />

:*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

<br style="clear:both;" />

= 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):
[[Image:ProxySet1.JPG|left|thumb|500px|Proxy Set ID1]]
<br style="clear:both;" />
:*Proxy Set ID#1 for AlpahCom XE7
::*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;" />

== 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):

[[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

<br style="clear:both;" />
----
[[Image:IPGrouptable-SIPserverAccount.jpg|left|thumb|500px|IP Group 2]]
<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

<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):

[[Image:CoderGroupAlphaCom.jpg|left|thumb|500px|Coder Group 1]]
<br style="clear:both;" />
:*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
<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: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;" />

== 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):
[[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.

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):
[[Image:IP-to-Tel-Routing.jpg|left|thumb|500px|Inbound IP Routing]]
<br style="clear:both;" />

:*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

<br style="clear:both;" />

== 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):
[[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.

:*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

<br style="clear:both;" />

== 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):
[[Image:Manipulation-table-Outgoing-call.jpg|left|thumb|500px|Manipulation Tel->IP Calls]]
<br style="clear:both;" />
:*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):<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;" />

= 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]].

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'.

[[Image:SIP Parameters.jpg|600px]]

:*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'''

[[Image:MP114-Coders.JPG|600px]]
:*Press Submit to save changes

= 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 [[Configuration guide for Cisco Call Manager 6]].

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 ==
[[Image:SIP2SIP-Cisco-Region.PNG|left|500px|thumb|Region Configuration]]
<br style="clear:both;" />
: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

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= Mobile VoIP GSM Gateway MV-370 =
''All text and pictures refer to version 6.693.t''
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For configuration of the MV-370 please see [[Configuration_guide_for_Mobile_VoIP|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 ==
[[Image:MV-370-SIP.PNG|left|thumb|500px|SIP Settings]]
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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 ==
[[Image:MV-370-Codec.PNG|left|thumb|500px|Codec Settings]]
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In the menu '''SIP Settings -> Codec Settings''', set codec priority:
:*'''Codec Priority 1:''' = G.729

Leave the rest as is.


[[Category:SIP]]
[[Category:3rd party integration]]