QUESTION inbound call > trunk > extension

istvan pirger

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Hi. Brand new here. Using FreePBX with Asterisk.
I have a VOIP provider giving me each DID (phone number) in a way that it becomes a trunk.

For example
phone number: 06 1 zzz zzzz
user: 061zzzzzzz
password: **
proxy: sip3.voice.eu
port: 5060

This turns into a PEER Details:
type=peer
username=061zzzzzzz
secret=**
host=sip3.voice.eu
qualify=yes
nat=yes
fromuser=061zzzzzzz
fromdomain=sip3.voice.eu
insecure=port,invite

and so on.. so I have 7 trunks for 7 numbers (DIDs).

I can't seem to direct inbound calls to extensions, based on witch DID/trunk is called.
I have inbound and outbound routes.

Any suggestions?
 
Last edited:
Welcome to the forums!
First, please sanitize the configuration you posted. It's a very bad idea to post credentials (working or not) for anything on such a public forum.
Second, It's unfortunate if they require you to have a separate trunk for each DID. Be that as it may, you should be able to use inbound routes normally unless your provider is providing the DID in a non-standard way. Do you see the DID number mentioned in the Asterisk logs at all during inbound calls?
 
This is what i see when a call comes in:

Executing [h@ext-local:1] Macro("SIP/xx x xxx xxxx-000006db", "hangupcall,") in new stack
Executing [s@macro-hangupcall:1] GotoIf("SIP/xx x xxx xxxx-000006db", "1?theend") in new stack
Goto (macro-hangupcall,s,3)
Executing [s@macro-hangupcall:3] ExecIf("SIP/xx x xxx xxxx-000006db", "0?Set(CDR(recordingfile)=)") in new stack
Executing [s@macro-hangupcall:4] Hangup("SIP/xx x xxx xxxx-000006db", "") in new stack
 
By that, check your inbound routes and at the bottom of the page it says where it sends the call. it is in 2 drop-down menus -- First may read <Terminate Call> and second may read <Hang Up>. To send to say... Extension 2544 change the first one to read <Extensions> and the list will change to all your extensions and then choose <2544> (in my example).

By the output you provided, you may have your inbound route set to just hang up on everyone or have them misconfigured as to have the default route take over which by default IIRC does hang up on everyone.

On inbound routes, check the right side for your routes and make sure they look like: IncomingDID/Any CID (e.g. 2479982233/Any CID) also make sure your provider does not send the 1 with it -- meaning not 12459982233 if they do adjust your inbound routes to match.

See if that helps. Good Luck
 
If that is a UK Manchester DID I think it has a digit missing?

Anyway edit the 1st message to sanitise the identifying DID
AND change your 'secret=dj********'
Or someone will spend your £-p
 
I just discovered the edit button, so i sanitized them, thanks.

When ever I add a DID number to the incoming calls and call that number then it says the number is not available... that's the problem having each DID connected to a trunk, because now the IPS servicing the DIDs and CIDs (in and out), all I get is a bunch of registered trunks, and I can't direct calls coming in on different trunks... makes sense?
 

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