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

[Peter P GMX]

Hello,

I finally found the "sip_cid_type=none" setting.

Best regards
Peter


Peter P GMX schrieb:
> 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.
>
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