Asterisk SIP Trunk Settings & VoIP Service Configuration Setup

Digium makes Asterisk available to the open source community under the GNU General Public License (GPL) and uses business-class Asterisk to power a broad family of products for small, medium and large businesses.

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

There are several GUI interfaces for Asterisk that simplify installation of Asterisk. These interfaces allow administrators to view, edit, and change most aspects of Asterisk via a web interface.

You can download free GUI versions of Asterisk from one the below links below:

- Download PBX in a Flash

- Download Elastix

- Download AsteriskNow

VoIPVoIP SIP trunking service enables customers to make calls from 1.9 cents/minute with no volume commitments, no monthly fees, no channel restrictions, with optional availability of US phone number with area code of your choice (or port you own US phone number for free), 800 toll free numbers or Virtual Phone Numbers from any 40+ countries of your choice.

Click here to learn more about VoIPVoIP Sip Trunking service and prices.

Below you can find Asterisk GUI SIP trunk settings and configuration guide for voip setup with VoIPVoIP phone service.

NOTE: While our goal is to make all Use Your Own Device installations as easy as possible, this option is intended for advanced users. VoIPVoIP does not provide technical support for Asterisk.

Outgoing Settings

Peer Details

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

type=peer

qualify=yes

secret=XXXXX (your VoIP VoIP password)

nat=auto

insecure=very

host=sip3.voipvoip.com

fromuser=5551231234 (your VoIP VoIP account assigned while signing up)

fromdomain=sip3.voipvoip.com

dtmfmode=rfc2833

disallow=all

allow=g729

allow=ilbc

allow=ulaw

allow=alaw

Incoming Settings

USER Details

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

type=user

secret=XXXXXX (your VoIP VoIP password)

nat=auto

insecure=very

host=sip3.voipvoip.com

fromdomain=sip3.voipvoip.com

dtmfmode=rfc2833

disallow=all

context=from-trunk

allow=g729

allow=ulaw

allow=alaw

allow=ilbc

DNS SERVER

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

REGISTRATION STRING

5551231234:[email protected]/5551231234

(for 5551231234 use your VoIP VoIP account and for XXXXXXXX use your password)

Problems? Please check our installation troubleshooter.

NOTE: VoIPVoIP does not provide technical support for asterisk.

Asterisk Security Issues: Please note that VoIPVoIP is not responsible for preventing unwanted physical or remote access to your Asterisk IP PBX. If your Asterisk IP PBX is compromised then you will be responsible for any damage caused. Click here to read our Security Recommendations for Asterisk.