SIP Trunking with Elastix

SIP Trunking with Elastix

Configuration Guide for Matrix SETU VTEP

SETU VTEP configuration guide for Elastix Page | 2

Contents Setup Diagram 3 SIP Trunk Configuration in Elastix for SETU VTEP 4 Outgoing Call configuration in Elastix 7

Incoming call configuration in Elastix 9 SIP Trunk Configuration in SETU VTEP for Elastix 11 E1 to SIP Routing Configuration in SETU VTEP 18 FAX Configuration in SETU VTEP 20 FAX Configuration in Elastix 21 Viewing SIP Trunk Status in SETU VTEP 21


Set Up Diagram:

(Note : PRI line from PBX is terminated to VTEP as per above setup diagram)

Host Server Environment Details

Description

Server OS CentOS release 5.7 (Kernel 2.6.18-274.e15 on an i686)

Software Type Elastix 2.4.0

Software version

Elastix (2.4.0) Elastix-2.4.0-Stable-i386-bin-04Feb2013

Test Setup Equipment:

Equipment Model Version

SETU VTEP N/A V1R9

Analog Telephone N/A N/A


SIP Trunk configuration in Elastix for SETU VTEP

Open the Web GUI of Elastix server in web browser as shown below (IP of server will be

same as IP of System on which Elastix is installed).

http://192.168.51.6/admin/config.php# (here, 192.168.51.6 IP is allocated for Elastix WEB

GUI access)

Enter username & password as entered while installing Elastix.

Default username: admin, Password: admin

On Successful login home Page opens

On the top of the screen, click on PBX Configuration menu. To add a Trunk select Add

SIP Trunk in Trunk menu


In Trunk Name field configure Type as VTEP

In CID Options field configure as Allow Any CID

Configure Maximum Channels as 8

In Dialed Number Manipulation field configure desired number pattern which has to be

routed via VTEP ( 1. Used in this example, all Outgoing numbers with prefix 1 has to be

routed via VTEP)

In Outgoing Settings, configure Trunk Name as vtep.

Configure PEER Details

o host=***provider ip address***

provider ip address is the IP address of VTEP. (i.e. 192.168.51.238)

o type=peer

Peer to Peer trunk between VTEP and Elastix

o dtmfmode=rfc2833

DTMF detection and Out-dial mode is set to rfc2833(OUTBAND)

Click on Submit changes.


Click on Apply Configuration Changes Here

New configuration setting will be saved and Elastix will be reloaded


Outgoing Call Configuration in Elastix:

Open Outbound Routes.

In Route Name field configure any desired Rule Name ( interop name used in this

Example)

In Dial Pattern field configure desired number pattern which has to be routed via VTEP

(_1. Used in this example, all Outgoing numbers with prefix 1 has to be routed via VTEP)

Prefix can be used before dialing the desired number. ( 9 used in this example)

Configure desired caller id in Caller ID field

In Trunk Sequence for Matched Routes configure the SIP trunk which we created.(In

this example, we have created VTEP)

Click on Submit Changes and Apply Configuration Changes Here as well.



Incoming Call Configuration in Elastix:

Open PBX Configuration. Click on Inbound Routes

Configure Description field as name of Incoming route. (In this example it is configured

as Incoming_VTEP )

Set Destination for Incoming call Router ( selected 7700 extension in Example)

Click on Submit

Click on Apply Configuration Changes Here.



SIP Trunk configuration in SETU VTEP for Elastix:

Open the browser, Internet Explorer/Mozilla Firefox, on the standalone computer/LAN

PC to which the Network Port of SETU VTEP is connected

Enter the current IP address of Network Port of your SETU VTEP in the address bar of the

browser.

If the IP Address has not been changed, you may enter the default IP Address of the

Network Port. ( 192.168.51.238 IP used in this example)

The login page opens.

In the Login Password box of the login page, enter the System Engineer password

(Default: 1234), and then click the Login button.


On successful login, the home page opens

Under Basic Settings on the left navigation bar, click the SIP Trunk link. The SIP Trunk

parameters page opens.

On the SIP Trunk page, choose any SIP Trunk number from 01 to 32 to register with

Asterisk, by clicking the SIP trunk number tab; for example, SIP Trunk number 1.

For the SIP Trunk number you select, make sure the SIP Trunk Enable check box is

selected.

Assign a Name to the SIP trunk as identification; for example, Elastix.


a) Registrar settings:

Click Registrar Settings to expand. In SIP Trunk Mode, select Peer to peer.

Click on Arrow Button beside peer to peer. A second window will pop-up


Click on Edit

Third window will open. Configure IP of Asterisk in Destination Address Field

Click Submit and Close to close 3rd window

Close 2nd windows by clicking on Close button

Click on Submit in main window to save


b) Selecting Preferred Codecs

Under Basic Settings on the left navigation bar, click the SIP Trunk link. The SIP Trunk


Click Vocoder Preference to expand

Set Vocoder preference. As 1st Preference and 2nd Preference, set either G.711 A-Law

or G.711 μ- Law.

From the Selected Codecs list box, move all other codecs—GSM FR, iLBC 30ms, iLBC 20

ms, G.729— one by one, to the Available Codecs box, by selecting the codec, and then

clicking the BACK ARROW.

Make sure only G.711 (A-law) and G.711 (μ- Law) remain in the Selected Codecs list

box. You may set one of these codecs as the First preference, by selecting it and using

the UP ARROW

Click Submit to save

v


c) Handling of Incoming Calls

Click Handling of Incoming Calls to expand.

In Route all Incoming calls( with CLI & without CLI ) option configure to the called party

number

In Select Destination for Routing Calls Configure Fixed

Click on submit


d) Handling of Outgoing Calls:

Click Handling of Outgoing Calls to expand.

Tick mark flag Connect Source Port When 183 ( Session Progress) is received on SIP

In SIP ID FROM Header of Invite message field configure Caller Id Received on Source

Port

Click on submit

Click


E1 to SIP Routing Configuration in SETU VTEP

Under Basic Settings on the left navigation bar, click the T1E1 Port. The T1E1


Click Handling of Outgoing Calls to expand.

Select Radio button Port Wise

Click arrow beside Port Wise. A new window will open

In Route all Incoming calls( with CLI & without CLI ) option configure to the called party

number

In Select Destination for Routing Calls Configure Fixed

Click on Submit and Close

C


Tick mark flag Connect Source Port when Progress Indicator is Received on T1E1 Port

Click on Submit in main window to save

Click Handling of

Click Handling of

Click Handling of


Fax Configuration in SETU VTEP:

Click Advanced Settings to expand.

Default Fax protocol is T.38. You can configure Pass-through or T.38 FAX by selecting

respective Radio Button.

v


Fax Configuration in Elastix:

Default FAX protocol in Elastix is Pass-through.

To configure T.38 in Elastix following command needs to be given from root,

1) Setup /etc/asterisk/udptl.conf manually

2) 2) add "t38pt_udptl=yes" to /etc/asterisk/sip_general_custom.conf

After allocating above commands Fax with T.38 protocol works if same is set at client

device also.

Viewing SIP Trunk Status in SETU VTEP

Once you have completed configuring the SIP Trunk, you may view its status.

To do this,

Click the Status link on the left navigation bar.

Under Status, click the SIP Trunk link.

The SIP Trunk Status page opens

Documents

SIP Trunking with Elastix