Sunday, February 24, 2013

Creating Free VOIP EXCHANGE part 2

 After the install login from web with
UserName: wwwadmin
Password:password
Then

 Settings>Asterisk SIP settings>NAT settings. Click "Auto Configure" It will fill out the IP addresses. If you reboot, localnet address may change. You may use "10.0.0.0/8" instead. Whenever you make a change, scroll down and click submit. Then "Apply Config" red button will appear at the top. Click it occasionally to reload newly submitted configurations.

2.2. Settings>Asterisk SIP settings>Audio codecs. Select the codecs and reorder. Personally, I use G722, G729 and ulaw.

2.3. Applications>Extensions. Add new SIP extensions. User extension and the secret are the username and the password you will use in your sip client to register with PIAF. Select nat=yes.

2.4. Other>Google Voice. Fill out your GV information. Asterisk must be restarted to take it into effect. In Web GUI, Admin>Asterisk CLI, execute "core restart gracefully" Once restarted, you can start making outbound GV calls from a registered phone. 

2.5. Connectivity>Inbound routes. Add your GV number as DID number. Scroll down and set destination as your extension you created in #2.3. You should now be able to answer incoming calls, if GV forwards to gchat.

Again for Installing codecs (from command line)

install-codecs

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In the Next part i will explain about the peering of two sip trunk and Iax2 trunk