Dear list,
using astlinux i see this :

When one extension make a call to another extension the remote party
display calling numer and called number .
In any other asterisk env i have that called party display only calling
number .

Here the sip message :

INVITE sip:1...@192.168.210.67:5060;rinstance=2b7d14b2e0f5cbf6 SIP/2.0
Via: SIP/2.0/UDP 192.168.210.234:5060;branch=z9hG4bK32cc5a92;rport
*From: "113" <sip:1...@192.168.210.234>;tag=as35abae33*
To: <sip:1...@192.168.210.67:5060;rinstance=2b7d14b2e0f5cbf6>
*Contact: <sip:1...@192.168.210.234>*
Call-ID: 01a9b41b44bc67a2588ee338435fd...@192.168.210.234
CSeq: 102 INVITE
User-Agent: Faro-PBX
Max-Forwards: 70
Date: Tue, 12 May 2009 13:13:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 399

Anyone know where this coming from ?

Obviusly my dialplan does not override any callerid.
I'm only using the sdtexten macro.
Regards

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