How do i make CID work for incoming routes?

Hi

I have freepbx version 2.10.1.1. Sorry to be super noob. But how do i get CID for incoming routes working?

I have two trunks registered & calls work inbound & outbound to ring groups I have set up. Thats all good.

Lets say I have trunks

77777777 & 8888888

How do I make it so that when a caller, say my mobile/cell calls in from 0422999999 , it automatically goes to a particular extension?

In incoming routes there is just the standard ANY did/any did incoming route.

If I add another route & in Caller ID number put 0422999999 & tick the box CID priority route ( or even if I untick it), go to extension 201. Nothing happens. The pbx totally ignores it any just processes the normal Any DID/CID incoming route.

What am I doing wrong?

Leap frog, I see no Chevrons next to incoming route entries. I seem to remember those little arrows in elastix or trixbox, but I dont see them in free pbx. Is there some other way to order in the version I have?

Skyking. I can access the CLI ok, not sure what I am missing there. But where is the asterisk full log? /etc/asterisk/logger.conf? I must have the wrong one, because there is nothing in that one.

Maybe if I show you my set up, there might be something obvious to you guys.
There are the following which I have filled out:

  1. Description : Divert my mobile to ext 201
  2. DID number: 77777777 ( because that is the trunk that will be called?)
  3. Caller Id number: 0422999999 (because that is my mobile number. The number I want CID to identify?)
  4. CID Priority route: ticked (Not sure if I should tick or untick this, but either way, nothing has happened.)
  5. Set destination: Extension 201

Does this look right or wrong?

You need to move the CID route above the any CID route. Always order routes from most specific to least specific.

Ahhh ok, I thought so! But I cant see where you sort them. Where do you sort the order of the incoming routes?

Drag and drop the little arrow looking chevron things in the list on the right.

Skyking, you are giving OP advice for outbound routes and s/he is asking about inbound routing. There is no ordering of inbound routes.

D2011, it looks like the method you have used in the first post is the correct procedure. Make sure you have defined the CID (and DID if used) EXACTLY how the trunk is providing that info. You can learn lots from watching the CLI and checking the Asterisk full log.

The DID and CID must match what is in the CDR exactly.

Ignore my comment on the arrows, as what pointed out I was mistaken.

This is the output from the CLI. Maybe you can see something that I can’t. This is the first bit that is informative.

-- Executing [77777777@from-trunk:1] NoOp("SIP/77777777-000000df", "Catc                                                                                                                                                              h-All DID Match - Found 88888888 - You probably want a DID for this.") in new                                                                                                                                                               stack
-- Executing [88888888@from-trunk:2] Goto("SIP/77777777-000000df", "ext-                                                                                                                                                              did,s,1") in new stack
-- Goto (ext-did,s,1)
-- Executing [s@ext-did:1] ExecIf("SIP/77777777-000000df", "1?Set(__FROM_D                                                                                                                                                              ID=s)") in new stack
-- Executing [s@ext-did:2] Gosub("SIP/77777777-000000df", "app-blacklist-c                                                                                                                                                              heck,s,1()") in new stack
-- Executing [s@app-blacklist-check:1] GotoIf("SIP/77777777-000000df", "0?                                                                                                                                                              blacklisted") in new stack
-- Executing [s@app-blacklist-check:2] Set("SIP/77777777-000000df", "CALLE                                                                                                                                                              D_BLACKLIST=1") in new stack
-- Executing [s@app-blacklist-check:3] Return("SIP/77777777-000000df", "")                                                                                                                                                               in new stack
-- Executing [s@ext-did:3] Set("SIP/77777777-000000df", "CDR(did)=s") in n                                                                                                                                                              ew stack
-- Executing [s@ext-did:4] ExecIf("SIP/77777777-000000df", "0 ?Set(CALLERI                                                                                                                                                              D(name)=anonymous)") in new stack
-- Executing [s@ext-did:5] Set("SIP/77777777-000000df", "__CALLINGPRES_SV=                                                                                                                                                              allowed_not_screened") in new stack
-- Executing [s@ext-did:6] Set("SIP/77777777-000000df", "CALLERPRES()=allo                                                                                                                                                              wed_not_screened") in new stack
-- Executing [s@ext-did:7] Goto("SIP/77777777-000000df", "ext-group,600,1"                                                                                                                                                              ) in new stack
-- Goto (ext-group,600,1)

This is the 2nd bit:

dialparties.agi: Starting New Dialparties.agi
dialparties.agi: Caller ID name is ‘Anonymous’ number is 'anonymous’
dialparties.agi: Methodology of ring is ‘ringall’
– dialparties.agi: Added extension 101 to extension map
– dialparties.agi: Added extension 102 to extension map
– dialparties.agi: Added extension 103 to extension map
– dialparties.agi: Added extension 104 to extension map
– dialparties.agi: Added extension 105 to extension map
– dialparties.agi: Added extension 106 to extension map
– dialparties.agi: Added extension 0422999999 to extension map
– dialparties.agi: Extension 101 cf is disabled
– dialparties.agi: Extension 102 cf is disabled
– dialparties.agi: Extension 103 cf is disabled
– dialparties.agi: Extension 104 cf is disabled
– dialparties.agi: Extension 105 cf is disabled
– dialparties.agi: Extension 106 cf is disabled
– dialparties.agi: Extension 0422999999 cf is disabled
– dialparties.agi: Extension 101 do not disturb is disabled
– dialparties.agi: Extension 102 do not disturb is disabled
– dialparties.agi: Extension 103 do not disturb is disabled
– dialparties.agi: Extension 104 do not disturb is disabled
– dialparties.agi: Extension 105 do not disturb is disabled
– dialparties.agi: Extension 106 do not disturb is disabled
– dialparties.agi: Extension 0422999999 do not disturb is disabled
– dialparties.agi: DbDel CALLTRACE/101 - Caller ID is not defined
– dialparties.agi: DbDel CALLTRACE/102 - Caller ID is not defined
– dialparties.agi: DbDel CALLTRACE/103 - Caller ID is not defined
– dialparties.agi: DbDel CALLTRACE/104 - Caller ID is not defined
– dialparties.agi: DbDel CALLTRACE/105 - Caller ID is not defined
dialparties.agi: EXTENSION_STATE: 4 (UNKNOWN)
dialparties.agi: Extension 106 has ExtensionState: 4
– dialparties.agi: Checking CW and CFB status for extension 106
– dialparties.agi: DbDel CALLTRACE/106 - Caller ID is not defined
dialparties.agi: EXTENSION_STATE: 4 (UNKNOWN)
dialparties.agi: Extension 0422724995 has ExtensionState: 4
– dialparties.agi: Checking CW and CFB status for extension 0422999999
– dialparties.agi: DbDel CALLTRACE/0422999999 - Caller ID is not defined
– dialparties.agi: Filtered ARG3: 101-102-103-104-105-106-0422999999
– <SIP/0731234138-000000e1>AGI Script dialparties.agi completed, returning 0
– Executing [s@macro-dial:7] Dial(“SIP/77777777-000000e1”, “SIP/101&SIP/102&SIP/103&SIP/104&SIP/105,20,TtrM(auto-blkvm)”) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Called SIP/101
– SIP/101-000000e2 is ringing

Why does it say I am anomymous, yet it can see my mobile number? What am I missing or configurted wrong?

Can anyone see what is wrong with my config?