Configuring AsteriskNOW

Setup Walkthrough: 1. Start internet browser 2. Go to servers IP address using port 8088, e.g. http://192.168.1.254:8088 3. Enter login info, default username is admin, default password is password 4. Change password when prompted. 5. Configure using the GUI. 1. Copy the link displayed on the Server to a Client PC Browser

2. Enter Username admin and Password password

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Creating Dial Plans

1. Click Dial Plans

2. Click New Dial Plan

Next:

3. Enter Dial Plan Name

4. Select features to be enabled

5. Click Save

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Next:

6. Click Default Checkbox of DialPlan1

7. Click Apply Changes

For every change in configuration, the administrator must click Apply Changes for the change(s) to take effect. Alternatively, he or she can use the command service asterisk restart from the servers command line (or the web GUI CLI). This command restarts Asterisk and consequently applies any change(s) made.

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Creating Users
2. Click Create New User 1. Click Users

Next:
Basic Options/Parameters that Need to be Set: CallerID the extensions/users caller ID name Dial Plan the dial plan that the extension will use SIP use sip as signaling protocol Codec Preference audio and video codecs to be used (by priority) SIP/IAX Password SIP/IAX password that would be configured on end devices for SIP/IAX authentication In Directory to allow the user to be mapped in the Directory service/extension (If Voicemail is enabled) VoiceMail Access PIN code users code to access voicemail after dialing the voicemail extension Email Address the email address to which voicemails would be sent to as an attachment(if configured to do so)

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User Extensions Screen After Creating Two Users:

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Voicemail
2. Enter extension no. Ex. 5000

1. Click Voicemail 3. Configure Parameters

4. Click Save

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E-mail Settings:

5. Click Email Setting for Voicemails

6. Configure Parameters

7. Click Save

***Remember to always apply changes for every change***

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http://www.jonathanmanning.com/2011/07/15/how-to-configure-asterisk-to-send-voicemail-email-via-gmail-smtp-guide/

On the Server:
1. Uninstall sendmail and postfix if they are installed. To do this, enter the command, yum remove postfix and yum remove sendmail. 2. Install ssmtp on the server. To do this, enter the command yum y install ssmtp. 3. To be on the safe side, backup the original ssmtp.conf file. To do enter the following commands in order: cd /etc/ssmtp , mv ssmtp.conf ssmtp.conf.org , touch ssmtp.conf 4. Create a new ssmtp.conf file or simply edit the old one (as long as you keep a copy of it in case things go wrong) and modify some of the necessary parameters. To edit the config file you could use nano or vi. The commands are nano /etc/ssmtp/ssmtp.conf or vi /etc/ssmtp/ssmtp.conf The following text in blue are the parameters that need to be modified in the ssmtp.conf file: mailhub=smtp.gmail.com:587 hostname=voicemail@your-domain-name.com root=voicemail@your-domain-name.com AuthUser=voicemail@your-domain-name.com AuthPass=Enter-Password UseSTARTTLS=yes UseTLS=yes FromLineOverride=yes

For example, username@gmail.com **since we are configuring the server to use gmail **the gmail account should be active and valid, if not first create a gmail account Password for your gmail account

5. Add an entry for the root user to the revaliases file. To do this, you need to edit the revaliases file. The commands are nano /etc/ssmtp/revaliases or vi /etc/ssmtp/revaliases Simply add the line: root:voicemail@your-domain-name.com:smtp.gmail.com:587

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Conferencing
2. Click New Conference Bridge

1. Click Conferencing

3. Enter Information

4. Click Update
***Remember to always apply changes for every change***

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Follow Me (Call Forwarding)

2. Click Edit 1. Click Follow Me

3. Click Add FollowMe Number

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4. Select existing number

5. Click Add

6. Click Save

***Remember to always apply changes for every change***

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Call Transfer

2. Check the option(s) that need to be enabled

1. Click Call Features

3. Assign (or leave to default settings) a key to transfer a call

***Remember to always apply changes for every change***

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Directory

1. Click Directory

2. Enter Directory number 3. Parameters to be enabled 4. Click Save

***Remember to always apply changes for every change***

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Advanced Options
2. Click Advanced Options

1. Click Options

3. Click Show Advanced Options

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THESE ADDITIONAL OPTIONS CAN NOWBE ACCESSED

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SIP Settings

2. Click TOS 3. Configure Parameter

4. Click Save

1. Click SIP Setting

***Remember to always apply changes for every change***

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2. Click Misc 3. Configure Parameter

4. Click Save

1. Click SIP Setting

***Remember to always apply changes for every change***

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2. Click Codecs 4. Click Save

3. Enable Codecs to be used by checking them. **enable H.263, H.263p, and H.264 for Video Calls

1. Click SIP Settings

***Remember to always apply changes for every change***

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Configuring X-Lite
1. Click Softphone 2. Click Account Settings

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Enter any Acount name Check Call Uncheck IM/Presence

Users Extension Number (configured on Asterisknow GUI) Servers IP Address SIP/ IAX Password (configured on AsteriskNOW GUI) Enter any Display name

Check Register with domain and receive calls

Click OK

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Enter Voicemail Extension configured on the server

Click OK

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Connected to AsteriskNow Server

Connected to AsteriskNow Server

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Offline Unable to connect to AsteriskNow Server

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