Vitelity inbound calls not connecting

[The Deacon]

Over the past 3 months, I've had several complaints from friends / clients who call the house / business that when they call, that their call just "hangs" and doesn't complete.

I've never experienced this until recently. I can make calls, but am unable to receive any calls.

I pinged the Vitelity folks and their recommendation is that I change my Inbound and Outbound configuration.

I'm running IncrediblePBX (Asterisk 1.4.21.2) on a Revo 1660.

Has anyone experienced this or had to do this? Or was my configuration wrong from the get-go?

-Rick

They are recommending that I change my inbound from this:

[vitel-inbound]
disallow=all
allow=ulaw
canreinvite=no
context=from-trunk
host=inbound7.vitelity.net
insecure=port,invite
username=<username>
secret=<secret_pw>
t38pt_rtp=no
t38pt_tcp=no
t38pt_udptl=no
type=friend

to this:

[vitel-inbound]
type=friend
dtmfmode=auto
host=inbound7.vitelity.net
context=inbound
username=<username>
secret=<secret>
allow=all
insecure=very
canreinvite=no

 

[jeffmac]

You did not say you were running PIAF - but I will at least assume you are running FreePBX....

I think FreePBX depends on the "from-trunk" context to do Inbound Routes. Vitelity cannot know what your context names are, so I'd call that recommendation less than helpful.

Other than that, they've suggested an old version of "insecure=" - the "very" setting is out of date (although still accepted) but your setting of "port,invite" is the current preferred setting to get the same function.

Also, they've opened you to all codecs rather than the "ulaw" you have coded, and changed your dtmf setting to auto (which you had defaulted).

I'd guess they gave you a "canned response" without any analysis of the problem.

Here's what I've used ever since getting a Vitelity account: (I have no idea where I got it originally)...

username=myid
type=friend
secret=mypassword
nat=yes
insecure=invite,port
host=inbound6.vitelity.net
disallow=all
context=from-trunk
canreinvite=no
allow=ulaw&gsm

Jeff

 

[ppmax]

I think you need the nat=yes item in your settings...

 

[kmcdaniel]

yes you need nat=yes....Also you don't need insecure=port,invite
As that is only if you have Asterisk 1.6.

I too had this problem. Also make sure you are not setting caller id on the trunk. Set it on the extension side.