[Freeswitch-users] how to get rid of second P-Asserted-Identity?

[Peter P GMX]

When receiving a call from PSTN and forwarding it to another PSTN number
and setting P-Asserted-Identity header, I found that 2
P-Asserted-Identity headers are present in the INVITE message. As the
target provider only checks the first one, the call is denied.

The first P-Asserted-Identity header is from the the incoming call. The
second is set in our dialplan.
Is there achance to drop the first header part?

Best regards
Peter

Here's the dialplan:
    <extension name="Forward to Net">
      <condition field="destination_number" expression="^\S*$">
        <action application="set"
data="effective_caller_id_number=02x1204xxxxx"/>
        <action application="set"
data="effective_caller_id_name=02x1204xxxxx"/>
        <action
application="export"><![CDATA[sip_h_P-Asserted-Identity=<sip:02x1204xxxxx at my.domain.de>]]></action>
        <action
application="export"><![CDATA[sip_h_P-Preferred-Identity=<sip:${caller_id_number}@my.domain.de>]]></action>
        <action application="bridge"
data="sofia/external/0162xxxxxxxxxxxxxx at provider.domain"/>
      </condition>
    </extension>

Here is the corresponding INVITE.
INVITE sip:0162xxxxxxxxxxxxxx at sip1.my.domain.de SIP/2.0.
Via: SIP/2.0/UDP 82.xxx.xx.1x3:5080;rport;branch=z9hG4bKjD7UvXB8XF5jD.
Max-Forwards: 29.
From: "02x139xxxxx" <sip:02x139xxxxx at 82.xxx.xx.1x3>;tag=U4v0ypBB1X78F.
To: <sip:0162xxxxxxxxxxxxxx at sip1.my.domain.de>.
Call-ID: 1cdc72b9-f7ea-122d-e685-001ec9b9dd2d.
CSeq: 132448209 INVITE.
Contact: <sip:mod_sofia at 82.xxx.xx.1x3:5080>.
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15434M.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY.
Supported: timer, precondition, path, replaces.
Allow-Events: talk, refer.
Privacy: none.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 320.
X-FS-Support: update_display.
P-Asserted-Identity: "02x139xxxxx" <sip:02x139xxxxx at 82.xxx.xx.1x3>.
(from incoing call)
P-Asserted-Identity:
<sip:02x1204xxxxx at my.domain.de>.                       (set in dialplan)
P-Preferred-Identity: <sip:02x139xxxxx at my.domain.de>.
.
v=0.
o=FreeSWITCH 1277106526 1277106527 IN IP4 82.xxx.xx.1x3.
s=FreeSWITCH.
c=IN IP4 82.xxx.xx.1x3.
t=0 0.
m=audio 26564 RTP/AVP 8 0 98 3 101 13.
a=rtpmap:8 PCMA/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:98 SPEEX/8000.
a=rtpmap:3 GSM/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=rtpmap:13 CN/8000.
a=ptime:20.