Elastix SIP Trunk Configuration Guide

Elastix is a unified communications software that integrates the best tools available for Asterisk-based PBXs into a single, easy-to-use interface.

It also adds its own set of utilities and allows the creation of third party modules to make it the best software package available for open source telephony.

The goals of Elastix are reliability, modularity and ease-of-use. These characteristics added to the strong reporting capabilities make it the best choice for implementing an Asterisk-based PBX .

If you are not an advanced user of Asterisk, we highly recommend the use of one of GUI interfaces of Asterisk for SIP Configuraiton.

Elastix as an Astersik GUI is one of the fastest way to get started building custom telephony solutions with Asterisk.

Simply download the .iso file, burn it to a CD, drop it into the CD or DVD drive on the target computer and in less than 30 minutes you will have a full functional Asterisk system ready for your custom telephony application.

Elastix interface allow administrators to view, edit, and change most aspects of Asterisk via a web interface.

You can download free GUI version of Elastix from below link:

- Download Elastix

 

While our goal is to make all Bring Your Own Device installations as easy as possible, this option is intended for advanced users. VoiceTrunking does not provide technical support for Elastix SIP Configuration.

Below you can find Elastix SIP Trunk configuration guide for VoiceTrunking SIP Trunk service.

Outgoing Settings

Peer Details

username=5551231234 (your VoiceTrunking account assigned while signing up)

type=peer

secret=XXXXX (your VoiceTrunking password)

nat=auto

insecure=very

host=sip.VoiceTrunking.com

fromuser=5551231234
(your VoiceTrunking SIP account assigned while signing up)

fromdomain=sip.VoiceTrunking.com

dtmfmode=rfc2833

disallow=all

allow=g729

allow=ilbc

allow=ulaw

allow=alaw

Incoming Settings

USER Details

username=5551231234
(your VoiceTrunking SIP account assigned while signing up)

type=user

secret=XXXXXX (your VoiceTrunking password)

nat=auto

insecure=very

host=sip.VoiceTrunking.com

fromdomain=sip.VoiceTrunking.com

dtmfmode=rfc2833

disallow=all

context=from-trunk

allow=g729

allow=ulaw

allow=alaw

allow=ilbc

REGISTRATION STRING

5551231234:XXXXXXXX@sip.voicetrunking.com/55551231234

(for 5551231234 use your VoiceTrunking SIP account and for XXXXXXXX use your VoiceTrunking password)

NOTE: Asterisk does not support DNS SERVER lookups for inbound calls.

If you also have virtual phone number with your SIP Trunk service please add the following line to the "sip_general_custom.conf" file

srvlookup=no