After upgrade, all my extensions are "unavailable, &quo

I was trying to debug a FWD outgoing route, and I clicked on
"Check for module updates". Freepbx said updates were available but that I would need to upgrade to 2.3.0, so I followed the instructions and
upgraded.
Now all of my extensions are “unavailable, please leave your message after the tone.” I can dial out OK, and outside callers reach my IVR
and listen to my greeting and list of extensions, but when they select an extension they are immediately sent to “unavailable” message.

How can I fix this? Or if not fixable, how can I go back to 2.2.2?

THANKS

what were you previously running? (sounds like you were running off of svn alpha code):

anyhow, for 2.3 - go pull the tarball and to an install_amp --force-version 2.2.0. Then go to module admin and accept all the updates that are now available and you should have your system back up.

Yes, I had been running an SVN version.
Originally I had installed 2.2.1-with-modules.tgz
then I needed the VMX module, so I updated to SVN per instructions.
It worked OK.
Then I was trying to setup FWD trunk, and couldn’t make outgoing FWD calls, so I clicked on “Module Update” and was told to
upgrade to 2.3.0beta1 which I did. Then my extensions broke.

So you’re saying I need to
#cd /usr/src/freepbx-2.3.0beta1
#./install_amp --force-version 2.2.0

Does it matter which --force-version I specify?
Presumably, the last SVN version I used was >2.2.1

xxxxxxxxxxxxxxxxxxxx

Now it’s totally broken.
If I dial “7777” I now get “The number you have dialed is not in service” :evil:

Dialing an extension still gives me “unavailable, please leave a msg”

If I dial in on a PSTN from outside the office I get:

-- Starting simple switch on 'Zap/5-1'
-- Executing NoOp("Zap/5-1", "No DID or CID Match") in new stack
-- Executing Answer("Zap/5-1", "") in new stack
-- Executing Wait("Zap/5-1", "2") in new stack
-- Executing Playback("Zap/5-1", "ss-noservice") in new stack

What’s wrong???
Why ss-noservice?

I haven’t any blacklist and I don’t allow anonymous SIP.

OK, I read the example for the new Day/Night module.
I setup a default inbound route to point to a Day/Night definition
which in turn points to IVR-Open

I don’t recall whethor I had explicitly setup a default inbound route
before, or whethor the new 2.3beta upgrade deleted the old inbound route setting when creating the new Day/Night module.

I am using trixbox 2.0 and it seems after upgrading to freePBX2.2.2, my default inbound route is broken as well. (I can’t remember what version of FreePBX it was running before).

What I found is that there seems to be a bug in the 2.2.2 code, where there is no “_X.” route when you add a default route.

in the extensions_additional.conf file in 2.2.2, the context [ext-did] has

exten => s,1,Set(__FROM_DID=s)
exten => s,n,Gosub(app-blacklist-check,s,1)
exten => s,n,Goto(ext-local,1001,1)

While in the older load (i had a backup of the config before the upgrade).
the [ext-did] has

exten => s,1,Set(FROM_DID=s)
exten => s,n,Gosub(app-blacklist-check,s,1)
exten => s,n,Goto(ext-local,1001,1)
exten => _X.,1,Noop(Catch-All DID Match - Found ${EXTEN} - You probably want a DID for this.)
exten => _X.,n,Goto(ext-did,s,1)

There is a _X. to catch all in the older code.

To fix the problem I manually added a _X. route inbound route entry via the web GUI.

Sorry, my bad.

It looks like I had replaced the new extensions.conf with the old extensions.conf.

The new extensions.conf has a [ext-catchall] context, so there is no need to add the new “_X.” route