Asterisk SIP Trunk Setting Example
Using freePBX/Trixbox you are able to do most of Asterisk's configuration without editing the individual configuration files such as sip.conf or extensions.conf. You can setup most of the features in web interface such as sip trunk, call routing, voicemail and other calling features. Below we will focus on the SIP trunk setup and parameters that will work with TieUs SIP Trunk Services.
SIP Trunk Configuration:
2. Select Trunks in left side navigation, and Select “Add SIP Trunk” in the middle of page
3. Once you are in “Add SIP Trunk” detail Page, scroll to the “Outgoing Settings” section
4. Give it a Trunk Name, ex, TIEUS_SIP
5. Under PEER Details, copy and paste the following sample, if your asterisk is version 1.4, 1.6 or 1.8, and replace the user id and password to your own id/password.
context = from-pstn
6. Scroll down to registration, enter your registration string in the following format
7. Click on Submit Changes to save the page
8. Click on the red bar on top of the page that has Apply Configuration Change to take effect.
Outbound Route Configuration:
1. under PBX Setting, Click on the Outbound Routes to configure your Asterisk box to send traffic to TieUs
2. Under Add Route Page, Enter a route name in Route Name field, ex, TO_TIEUS
3. Scroll down to Trunk Sequence and select the SIP/TIEUS_SIP trunk from the drop down list (if you have setup the trunk as TIEUS_SIP in previous page)
4. Click on Submit Changes, and click on the Apply Configuration Change to take effect.
1. under PBX setting, click on Extensions and select Generic SIP device on drop down list.
2. Enter the extension number under User Extension field, for example, 1000
3. Enter desired Display name. (ex, John, or Extension 1000)
4. Enter the desired password in the Secret field.
5. Click on Submit Changes, and click on the Apply Configuration Change to take effect.
Inbound Route Configuration (receive incoming call)
1. under PBX setting, click Inbound Routes
2. Most of customer will have a TieUs SIP account that bound with a DID number, we will create one route for this purpose.
3. Under Add Incoming Route section, enter your assigned user id in the DID Number field, for the previous example company, we will put 60428812741344 as the DID Number. Leave the Caller ID Number field blank.
4. Now go to the Set Destination section, and select Extension Option, from the drop down list, select your extensions. (in this example, select extension 1000)
5. After Submit Changes, you should see anew Inbound Route Entry Names
6. click on the Apply Configuration Change to take effect
Making Test Call and Trouble Shooting
2. Make incoming call by calling the DID from other phone to see if destination extension will ring.
3. If you have trouble makes call or receives call. Try to increase verbose log level in Asterisk Console. “asterisk –vvvvvr” This way, you will able to see some warning log about why the call cannot go out and come in.
4. Call our technical support for further assistance.
|Contact Our Technical Support For More Details! .......|