Incoming call do not enter Custom Context in extensions_custom.con

Hi, I’ve a Freepbx 13 with Asterisk 13.

I’ve a custom context to add null to incoming call number declared in extensions_custom.conf :

 [from-trunk-add-null]
exten => _X.,1,NOOP("Entered [add-null]")
exten => _X.,n,Set(CALLERID(number)=0${CALLERID(number)})
exten => _X.,n,Set(CALLERID(name)=${CALLERID(number)})
exten => _X.,n,Goto(from-trunk,${EXTEN},1)
include => from-trunk

In my trunk, settings, in incoming section I’ve added the incoming context :

context=from-trunk-add-null

But with an incoming call from this trunk, it doesn’t enter the custom context. It just enters the from-trunk context.

Why not just use the setcid module?

Because I’ve a trunk with 80 DID … So I don’t want to use setcid module 80 times… :slight_smile:

My guess is that inbound calls are not recognized by Asterisk as coming from that trunk, possibly being treated as anonymous? Call traces with sip debug should show this.

You are right. How can I change this ?

<------------->
[2017-07-13 15:07:59] VERBOSE[3088] chan_sip.c: Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
[2017-07-13 15:08:01] VERBOSE[3088] chan_sip.c:
<--- SIP read from UDP:XX.XX.XX.XX:5060 --->
OPTIONS sip:YY.YY.YY.YY SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX:5060;branch=z9hG4bK1d3df7c6
Max-Forwards: 70
From: "Unknown" <sip:[email protected]>;tag=as28dfc2fc
To: <sip:YY.YY.YY.YY>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-AsteriskNOW-12.0.25(11.10.2)
Date: Thu, 13 Jul 2017 13:07:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------->
[2017-07-13 15:08:01] VERBOSE[3088] chan_sip.c: --- (13 headers 0 lines) ---
[2017-07-13 15:08:01] VERBOSE[3088] chan_sip.c: Sending to XX.XX.XX.XX:5060 (no NAT)
[2017-07-13 15:08:01] VERBOSE[3088] chan_sip.c: Looking for s in from-trunk (domain YY.YY.YY.YY)
[2017-07-13 15:08:01] VERBOSE[3088] chan_sip.c:
<--- Transmitting (no NAT) to XX.XX.XX.XX:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP XX.XX.XX.XX:5060;branch=z9hG4bK1d3df7c6;received=XX.XX.XX.XX
From: "Unknown" <sip:[email protected]>;tag=as28dfc2fc
To: <sip:YY.YY.YY.YY>;tag=as698b1cd9
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Server: FPBX-13.0.192.9(13.15.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:YY.YY.YY.YY:5060>
Accept: application/sdp
Content-Length: 0