Configuring Asterisk PBX with Lync 2010docshare02.docshare.tips/files/28275/282756988.pdf · 2017....

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Configuring Asterisk PBX with Lync Server 2010 in home lab 1 www.baaskarcharles.com 1 Integrating Asterisk FreePBX with Lync Server 2010 Author: Baaskar R

Transcript of Configuring Asterisk PBX with Lync 2010docshare02.docshare.tips/files/28275/282756988.pdf · 2017....

  • Configuring Asterisk PBX with Lync Server 2010 in home lab

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    Integrating Asterisk FreePBX with Lync Server 2010

    Author: Baaskar R

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    Integrating Asterisk FreePBX with Lync Server 2010 ............................................................................................ 1

    AsteriskNow package Source ....................................................................................................................... 3

    Installing AsteriskNow .................................................................................................................................. 3

    Configure AsteriskNow ................................................................................................................................. 3

    Add Extensions ......................................................................................................................................... 3

    Lync Stress Tool Incoming call Extension ................................................................................................. 4

    Add SIP Trunk ........................................................................................................................................... 5

    Add Inbound Routes ................................................................................................................................. 7

    Add Outbound Route ............................................................................................................................... 8

    Configure PSTN Gateway in Lync Server 2010 ............................................................................................. 8

    Create Dial Plan, Voice Policy and Trunk Configuration ........................................................................... 9

    Testing / Troubleshooting the Configuration ............................................................................................. 12

    AsteriskNow Configuration .................................................................................................................... 12

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    AsteriskNow package Source Download the latest AsteriskNow 2.0.2 Here

    Installing AsteriskNow Boot the VM with AsteriskNOW-2.0.2-x86_64-DVD.iso image

    In the Setup options, type 1 and press Enter (By Default, the installation would continue with Option

    1. Option 1 installs AsteriskNow with FreePBX GUI)

    Follow the install wizard, installation wiill complete without the any interruption

    Login to the server as root and password provided during the installation

    Configure AsteriskNow Type Setup to configure Network configuration

    Make sure you provide proper gateway and DNS server IP addresses

    Reboot the server using command shutdown –r now

    Now, Login to AsteriskNow web GUI using http://X.X.X.X ( IP address of AsteriskNow server)

    Default user name and password for Web GUI is admin

    Add Extensions Click Extension and select Generic SIP Device and Click Submit

    Type the User Extension and Display Name as shown below. I have extensions starting with 2 and

    XXX digits, example 2000.

    Under Device Options, enter the secret (password for the user extension)

    http://www.asterisk.org/downloads/asterisknowhttp://x.x.x.x/

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    Click Submit

    Note: I am not configuring Voice mail or DID number for now so have not configured remaining sections in

    the Extension tab.

    Lync Stress Tool Incoming call Extension Need to create extension “0000” with different target routing to handle all the Incoming calls genrated from

    Lync stress tool and via Mediation server to PBX system. Follow the steps below:

    Under FreePBX main Console, Click Applications -> Select Generic SIP Device -> Click Submit

    Add following information as in the below screenshot:

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    Add Extension as 0000 and Secret – as0000

    Under Optional Destionations -> No Answer, select Feature Code Admin and Directory#

    Type “1” in the CID Prefix; As shown below:

    Leave remaining options to Default.

    Add SIP Trunk Click Connectivity and Select Trunks

    In the Right side of the window, click Add Trunk - > SIP Trunk

    In General Settings, under Trunk Name, provide name for your SIP Trunk

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    Under Outgoing Settings,type values as in screenshot below:

    Click Submit Changes.

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    Brief description of Outgoing Settings (Peer details):

    host=10.10.1.16 ; IP address of the Lync Mediation Server or Front-End Server

    if Mediation server is collated.

    transport=tcp,udp ; Trasmission protocol add both TCP and UDP

    port=5068 ; Listening port of Mediation Server or Collated Front-End

    Server

    type=friend ; Friend variable indicates Host allows both incoming &

    outgoing calls and messages

    context=from-internal ; Default context generated by FreePBX GUI

    qualify=yes ; Yes allows Asterisk server to check if Peer (Lync Server) is

    still Online

    insecure=invite ; Do not require authentication of incoming INVITEs

    Add Inbound Routes Click Connectivity -> Inbound Route

    In Description Field: Provide name for the incoming Route

    In Set Destination, select Trunks and then select SIP Trunk created earlier.

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    Click Submit

    Add Outbound Route In Route name, Enter a name

    In Set Patterns that will use this Route, enter 7XXX (I planned to assign 7XXX series for Lync

    extensions hence, this pattern will allow all the extensions dailed with starting 7 and followed by 3

    digits will get routed to SIP Trunk)

    In Trunk Sequence for Matches Routes. Select SIP Trunk created earlier

    Click Submit Changes

    This ends the configuration of AsteriskNow.

    Configure PSTN Gateway in Lync Server 2010 Open the Lync Topology Builder and download the deployed topology

    In the section PSTN Gateway, right click New PSTN Gateway

    Enter AsteriskNow IP address

    Type Listening Port as 5060

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    Under SIP Transport Protocol, select TCP and click OK

    Right Click PSTN Gateway newly added in the Topology, publish the topology.

    Create Dial Plan, Voice Policy and Trunk Configuration First, lets create dial plan and assign the PSTN Gateway , voice policy to dial plan.

    Open the Lync Control Panel, click on Voice Routing

    Under Dial Plan, double click, Global

    Now under Normalization Rules, click New and enter values similar to below screenshot:

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    I have imcoming calls from Asterisk PBX to Lync with 7XXX extensions, creating normalization rules to check

    starting digit as 7 and add “+” infront of 7XXX extension to make the dialed number as e.164 format.

    Click Ok twice. In the main Dial Plan main window, select Global and click Commit and select Commit

    All

    Now Click on Voice Policy

    Click on Voice Policy and downlob click Global

    In Associated PSTN Usages, click New

    Enter the name for Voice Policy

    In Assoicated Routes, click New

    Under Name section, Enter Name

    Click Add and enter 7 (my Lync extentions starts with 7)

    Under Associated gateways, click Add and select PSTN listed

    CLick Ok 3 times to come back to main window

    Select GLobal and select Commit - > Commit All

    Click on Trunk Configuration

    Double click Global

    Under Assoicated Transliation Rules, Click new

    Under Name field, Enter Name

    Add remaining as per screenshot.

    This rule is created for allowing outgoing call from Lync, Normalization is required for outgoing calls. For

    examples, Lync client will dial out 3000 as +3000. Normalizatin rule is created to remove "+" and send 3000

    to PBX system

    CLick OK twice

    Select Global and Commit All.

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    Configuration of Lync Policies to handle Lync Stress Tool Outgoing Calls

    Note: You need to remove the above rule “Call from Lync to PSTN” and add below rule to make Lync Server

    work for Stress tool. To check the End to End Connectivity, don’t create the rule below yet.

    To handle the outgoing call (Number +14250000000) from Lync Stress tool and route it to PBX system, create

    addition Rule in the Trunk Configuration as below:

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    This ends the Lync configuration.

    Configure 3CX Softphone & add Extension

    Download and install 3CX latest version from Here

    Open the 3CX soft phone and Right on the Display Screen

    Select Accounts and Click New

    Add the information as shown in the screenshot below:

    Note: You need to create Extension 0000 in the Asterisk PBX system before configuring 3CX softphone.

    Testing / Troubleshooting the Configuration

    AsteriskNow Configuration 1. To test the connectivity between AsteriskNow and Lync Mediation Server

    [root@pbx ~]# telnet 5068

    If result shows “Not Connected” then Lync Mediation Server is not listening on port 5068.

    Check your Lync topology on what port is configured for Lync Mediation Listening Port (Don’t get

    confused with PSTN Gateway listening Port)

    Go to Lync mediation server and open Command Prompt, type Netstat –a This gives all listening

    ports and respective IP address of Mediation Server.

    2. Check AsteriskNow service running

    [root@pbx ~]# service asterisk status

    asterisk (pid 4652) is running...

    http://www.3cx.com/VOIP/voip-phone.html

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    OR

    [root@pbx ~]# /etc/init.d/asterisk status asterisk (pid 4652) is running...

    3. How to enable debugging in AsteriskNow

    [root@pbx ~]# asterisk –r

    [root@pbx ~]# sip set debug on

    If you want to enable debug on specific peer then follow below command:

    [root@pbx ~]# sip set debug peer

    To List Peer names added in your PBX:

    [root@pbx ~]# sip show peers

    4. Check SIP peers and SIP Trunk status and registration in AsteriskNow

    [root@pbx ~]# sip show peers

    [root@pbx ~]# sip show registry