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Asterisk SIP Trunk Settings & VoIP 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:
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
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
REGISTRATION STRING
5551231234:XXXXXXXX@sip3.voipvoip.com/55551231234
(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.
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