Setting up Inbound trunks from Gradwell

Asterisk Asterisk Support
1

Hi,

First post here - and if anyone could lend a little assistance, I’d appreciate it - I have a new Asterisk system (PIAF and using FreePBX to admin), and am trying to connect this up to SIP trunks. Outbound is working fine, with the firewall forwarding ports 5060 and 10k-20k back to Asterisk with a static external IP.

I’m trying to get the Inbound trunk connected, and Gradwell’s support keep forwarding me to their SwitchVox support article:
gradwell.com/support/kb/article.php?id=148

I’m trying to translate that into a User Context and User Details in the Trunk configuration in FreePBX, but no matter what I try, Inbound calls don’t arrive. (In fact, when logged into the Asterisk CLI -vvvvvr I don’t see any entries when I dial in.

My Current config:

PEER Details:
host=sip.trunk.gradwell.net
type=peer

USER Context: One of my DID numbers
USER Details:
type=user
qualify=no
host=sip.trunk.gradwell.net
fromdomain=sip.trunk.gradwell.net
dtmfmode=rfc2833
disallow=all
allow=g729&alaw&ulaw

Register string: blank

Anyone have any ideas on where to look to see why this is going wrong - or has anyone tried connecting with Gradwell before? I have a separate trunk set up for Skype Connect, and both inbound and outbound work fine for that (which uses Register strings and user/pass authentication), and Gradwell are assuring me the trunk is fine, having done traces, saying it’s the Asterisk config that’s incorrect.

2

You normally need to register with SIP providers, so you want a non-blank register string.

3

Hi

Gradwell Sip inbound dont accept registrations, You point the trunk at your fixed IP address or domain name.

You will need to make sure nat is setup correctly and the default inbound context correct

the actual knowledgebase article is gradwell.com/support/kb/article.php?id=105

4

That knowledgebase says you need to enable Allow Anonymous Inbound Calls…yikes! :open_mouth:

5

Hah, yeah…that’s not a good idea.