Issues with Inbound route not working

FreePBX Installation / Upgrade
#1

Hello,

I just recently got my FreePBX server all up and running including my polycom phones. Were using a 40 channel sip trunk to place and receive outside calls. The outbound routes worked fine, but I’m having problems getting my in bound route to work. When I called the number I got a “This number is not in service message”. I decided to check the Asterisk log to see if it was asterisk saying that our our trunk. So i connect to the asterisk console and placed the call again and saw this:

  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Executing [[email protected]:1] NoOp("SIP/XXX.XXX.XXX.XXX-00000027", "Received incoming SIP connection from unknown peer to XXXXXXXXXX") in new stack
    -- Executing [[email protected]:2] Set("SIP/XXX.XXX.XXX.XXX-00000027", "DID=XXXXXXXXXX") in new stack
    -- Executing [[email protected]:3] Goto("SIP/XXX.XXX.XXX.XXX-00000027", "s,1") in new stack
    -- Goto (from-sip-external,s,1)
    -- Executing [[email protected]:1] GotoIf("SIP/XXX.XXX.XXX.XXX-00000027", "0?checklang:noanonymous") in new stack
    -- Goto (from-sip-external,s,5)
    -- Executing [[email protected]:5] Set("SIP/XXX.XXX.XXX.XXX-00000027", "TIMEOUT(absolute)=15") in new stack
Channel will hangup at 2012-08-28 08:28:27.127 EDT.
    -- Executing [[email protected]:6] Answer("SIP/XXX.XXX.XXX.XXX-00000027", "") in new stack
    -- Executing [[email protected]:7] Wait("SIP/XXX.XXX.XXX.XXX-00000027", "2") in new stack
Really destroying SIP dialog '[email protected]' Method: OPTIONS
    -- Executing [[email protected]:8] Playback("SIP/XXX.XXX.XXX.XXX-00000027", "ss-noservice") in new stack
    -- <SIP/XXX.XXX.XXX.XXX-00000027> Playing 'ss-noservice.ulaw' (language 'en')
    -- Executing [[email protected]:9] PlayTones("SIP/XXX.XXX.XXX.XXX-00000027", "congestion") in new stack
    -- Executing [[email protected]:10] Congestion("SIP/XXX.XXX.XXX.XXX-00000027", "5") in new stack
  == Spawn extension (from-sip-external, s, 10) exited non-zero on 'SIP/XXX.XXX.XXX.XXX-00000027'
    -- Executing [[email protected]:1] Hangup("SIP/XXX.XXX.XXX.XXX-00000027", "") in new stack
  == Spawn extension (from-sip-external, h, 1) exited non-zero on 'SIP/XXX.XXX.XXX.XXX-00000027'

So I know the call made it into the server, but my inbound route for that DID is not working.

The settings on the inbound route are as follows:

Description: Main
DID: XXXXXXXXXX (matches the DID the trunk is reporting)
CallerID Number: (blank)
CID Priority Route: unchecked
Alert Info: (blank)
CID name prefix: (blank)
Music On Hold: default
Signal RINGING: checked
Pause Before Answer: (blank)
Privacy Manager: no
Detect Faxes: yes
Fax Detection type: sip
Fax Detection Time: 4
Call Recording: allow
CID Lookup Source: none
Language: (blank)
Destination: IVR Main

Whats the next steps I should be doing to figure out why it’s not wanting to route the inbound calls?

#2

Your trunk settings are wrong, I doubt it’s a registration issue:

Your trunk is not matching:

Received incoming SIP connection from unknown peer

The unknown peer part indicates this.

As Big Dicko sez you can also allow anonymous SIP.

#3

You willl need to either register the inbound trunk with the provider or allow anonymous inbound calls if using ip based routing.

#4

The trunk settings were given to me by the provider. They do IP based filtering. So i did have to set allow anonymous.

Though I have run into another issue some what related to this. The inbound route points to a IVR I set up. I’ve enabled Direct Dial to extensions. This works, but I have also added in a custom extension 3999 defined in the extensions_custom.conf file. From any extension it can be dialed and functions fine, but when trying to dial it from the IVR it reports back as invalid. After some digging I believe the “from-internal-custom” is not being included anywhere in “from-did-direct-ivr” definitions.

I’ve tired just adding in a include to the from-did-direct-ivr for the custom extension, but when i do that calling the IVR does this whole mess again playing “Your call can not be completed as dialed” I take the include out and reload asterisk then everything functions correctly.

#5

Put you custom extension in

ext-local-custom

and/or

from-did-direct-ivr-custom

the IVR should find it in there. Or easier define your custom extension in the gui.

#6

Adding it to ext-local-custom did not help. I have tired both a include to it and just simply moving all the programming for it under the ext-local-custom.

It just gives me a “invalid” response in the ivr.

---- Just saw your edit

I do have the custom extension defined in the gui. Under Admin > Custom Extensions unless there is another place i should be doing it. Though this not a normal “phone” extension it requires a authentication and then input from the caller. After the input it fires off and run a external program and based on that programs out put it does something else, and sort of repeats this a couple times. Then Hangs up when done.

#7