trixbox / Asterisk setup to use Bandwidth.com SIP Trunks
If you are trying to get your SIP trunks with Bandwidth.com established, you will most likely want to use something very similar to the following config. This can be accessed via the Asterisk > FreePBX > Setup > Trunks menu. You will want to delete the default IAX trunk and add a new SIP trunk. Delete any italicized comments before copy-pasting this config.
Outbound dial prefix = +1
Trunk name= bandwidth (or whatever you want – must be unique in your system)
[Outgoing Settings]
allow=g729 (if you have not licensed g729 from Digium, you want to use “allow=ulaw” instead)
dtmfmode=RFC2833
fromdomain=4.79.212.236
fromuser=+1xxxxxxxxxx (your 10 digit phone number here)
hidecallerid=yes
host=4.79.212.236
nat=yes
port=5060
type=peer
username=+1xxxxxxxxxx (your 10 digit phone number again)
[Incoming Settings]
User Context = from-pstn
canreinvite=yes
context=from-pstn
dtmfmode=RFC2833 (important if you want your IVR prompts to work)
fromdomain=4.79.212.236
host=4.79.212.236
insecure=very
nat=yes
port=5060
type=peer
username=+1xxxxxxxxxx (your 10 digit phone number again)
Filed under: Technology
Thank you Dennis. I have just deployed my trixbox PBX and am now waiting for Bandwidth.com to provision my truncks and DIDs. They had to purchase additional numbers for m rate center so there is a slight delay but I am excited. Your post was most helpful in providing the information that trixbox, asterisk, and Bandwidth.com weren’t providing. I’ll let you know how it goes once my numbers are provisioned and everything’s up and going.
Hi Chris,
You’re welcome. Let me know if you have any other issues and I will try to help where I can. I have left the business where Trixbox was installed, so I do not have a reference available, but I might be able to remember off the top of my head. Best of luck with your deployment!
-Dennis
Thank you Dennis. Your settings worked great as a reality check.
Thanks again!!
WJ