Get help with installing, upgrading and running a PBX such as Asterisk.
By tombal
#1099
I am trying to configure an SIP trunk from voicehost.co.uk to work with Asterisk.

Although I set a non-specific catch-all inbound route, it comes up with the error:

Call from 'ST20042T001' (X.X.X.X:5060) to extension '+448XXXXXXX' rejected because extension not found in context 'inbound'
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By WelshPaul
#1101
To setup a Voicehost trunk in FreePBX...

Navigate to Connectivity > Trunks > make the following changes:

General Settings
Trunk Name: Voicehost

Outgoing Settings
Trunk Name: Voicehost
PEER Details:
Code: Select all
secret=PUT YOUR PASSWORD HERE
type=peer
insecure=port,invite
defaultuser=VOICEHOST USERID
fromuser=VOICEHOST USERID
context=from-trunk
canreinvite=no
fromdomain=st.sipconvergence.co.uk
host=st.sipconvergence.co.uk
nat=yes
disallow=all
allow=alaw&ulaw&gsm&g722
dtmfmode=rfc2833
Register String: VOICEHOSTUSERID:VOICEHOSTPASSWORD@st.sipconvergence.co.uk/YOUR DID NUMBER

Now navigate to Connectivity > Outbound Routes and configure it as shown below:
outbound.png
Outbound Route
outbound.png (58 KiB) Viewed 1593 times
Finally just create an inbound route by navigating to Connectivity > Inbound Routes as shown below:
inbound.png
Inbound Route
inbound.png (40.29 KiB) Viewed 1593 times
In the above I have my trunk ring a group, however just select the extension you want it to ring in on in your case. You can use the same values as shown above to setup your trunk in asterisk sip.conf.

The error in your post indicates you are receiving an inbound call, however instead of it going to extension 200 or whatever extension you want it t go to you have set it to go to extension +448XXXXXXX which does not exist. It's a phone number!

OK thanks, what VPN are you using ??

A good write up on SIP ALG: https://www.voicehost…

Thanks very much. Really appreciate it! :-D

Attached below is my latest OBIHAI UK configuratio…