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I’ve gone rather mad with VoIP configurations since I decided to move to using my own PBX for my personal and business telephony. My choice of installation was Elastix, mostly because it looks nice, but it works really well. These details will work with pretty much any other Asterisk PBX such as Trixbox though. Right, so let’s setup the SIP trunk. The details you need to set are as follows:
Maximum Channels: 2
PEER Details:
allow=alaw&ulaw&G.726&gsm&g729&g723&g777
canreinvite=no
context=from-trunk
disallow=all
dtmfmode=rfc2833
fromdomain=sip.voipcheap.com
fromuser=USERNAME
host=sip.voipcheap.com
insecure=very
nat=YES
port=5060
qualify=yes
secret=PASSWORD
srvlookup=yes
type=peer
username=USERNAME
User Context: leave blank
User Details: clear so it’s blank
Register String: USERNAME:PASSWORD@sip.voipcheap.com/USERNAME
So long as you have replaced all of the USERNAME’s and PASSWORDS with your VoipCheap ones, you should be laughing. Remember that to dial out, you will need to setup and outbound route!
November 24th, 2012 at 12:31 pm
How many channels can you use on this type of setup ?
January 7th, 2013 at 9:33 pm
You would need to ask your VoIP provider regarding the number of channels as it may vary, but the most we ever use outbound is 1-2. We quite often have 5-10 people in a conference, and a £10/month VPS handles it fine.
October 24th, 2014 at 2:10 pm
Is this post still up to date? i.e will it still work? What VPS are you running it on?
October 25th, 2014 at 10:13 am
Hi Nathan, I don’t use VoIP Cheap any more as we had a number of call quality issues with them. In addition, we occasionally got messages on our PBX which were actually other people’s “in-progress” calls! Saying that, if you must use them, these instructions should still work.
I’ve used a few VPS providers. Now I just use PrimeXeon.