Hello,
IMHO asterisk acts exactly as it should. How else do you think it should
it prevent sending out the callerid name or num when you set it to prohib?
Asterisk doesnt support the privacy header for outgoing calls so
changing the name and number is the only way to do this. Maybe you could
do this in your dialplan with SipAddHeader("Privacy: full") instead of
setting the prohib flag.
in the Remote-Party-ID header is a special privacy option which asterisk
sets when using this header so you will see the original values there
but privacy is also set to full.
best regards
steve
Am 19.01.12 10:10, schrieb Paris Stamatopoulos:
> Hello all,
>
> We are testing Asterisk 1.8.8.1. In the following scenario, peer 54321 dials
> 12345:
>
> INVITE sip:[email protected] SIP/2.0
> Record-Route: <sip:10.1.1.86;lr=on;ftag=5ebe58983f6c0c84o3>
> Via: SIP/2.0/UDP 10.1.1.86;branch=z9hG4bKa6c1.4be79d43.0
> Via: SIP/2.0/UDP
> 192.168.4.80:5063;rport=5063;received=10.1.1.86;branch=z9hG4bK-ca013550
> From: "54321" <sip:[email protected]>;tag=5ebe58983f6c0c84o3
> To: <sip:[email protected]>
> Call-ID: [email protected]
> CSeq: 102 INVITE
> Max-Forwards: 69
> Contact: "54321" <sip:[email protected]:5060>
> Expires: 240
> User-Agent: Linksys/SPA942-6.1.5(a)
> Content-Length: 399
> Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
> Supported: replaces
> Content-Type: application/sdp
>
> v=0
> o=- 24058301 24058301 IN IP4 192.168.4.80
> s=-
> c=IN IP4 192.168.4.80
> t=0 0
> m=audio 16450 RTP/AVP 18 0 2 4 8 96 97 98 101
> a=rtpmap:18 G729a/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:2 G726-32/8000
> a=rtpmap:4 G723/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:96 G726-40/8000
> a=rtpmap:97 G726-24/8000
> a=rtpmap:98 G726-16/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:30
> a=sendrecv
>
> Peer 54321's caller ID will be blocked, since it enters the following context:
>
> [outgoing]
> exten => _X.,1,Log(NOTICE, Test)
> exten => _X.,n,Set(CALLERID(num-pres)=prohib_passed_screen)
> exten => _X.,n,Set(CALLERID(name-pres)=prohib_passed_screen)
> exten => _X.,n,Dial(SIP/${EXTEN}@Peer)
> exten => _X.,n,Hangup()
>
> When asterisk dials peer 12345, it rewrites the "From" header ("asterisk"
> <sip:[email protected]>) instead of keeping it intact. The "Remote-Party-ID"
> on the other hand, is correct.
>
> INVITE sip:[email protected]:5061 SIP/2.0
> Via: SIP/2.0/UDP 10.1.1.88:5060;branch=z9hG4bK25e11cb8
> Max-Forwards: 70
> From: "asterisk" <sip:[email protected]>;tag=as6d2aa852
> To: <sip:[email protected]:5061>
> Contact: <sip:[email protected]:5060>
> Call-ID: [email protected]:5060
> CSeq: 103 INVITE
> User-Agent: Asterisk
> Authorization: Digest username="asterisk", realm="10.1.1.87", algorithm=MD5,
> uri="sip:[email protected]:5061",
> nonce="6dfc149a8a6801201ba2b28860d6df704f17daeb",
> response="69d5626fcc5a24980bf641eb1f013813"
> Date: Thu, 19 Jan 2012 08:57:15 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
> PUBLISH
> Supported: replaces
> Remote-Party-ID: "54321"
> <sip:[email protected]>;party=calling;privacy=full;screen=yes
> Content-Type: application/sdp
> Content-Length: 525
>
> v=0
> o=root 2107042325 2107042326 IN IP4 10.1.1.88
> s=m1
> c=IN IP4 10.1.1.88
> b=CT:384
> t=0 0
> m=audio 12136 RTP/AVP 18 3 8 0 9 111 101
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:3 GSM/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:9 G722/8000
> a=rtpmap:111 G726-32/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
> m=video 16724 RTP/AVP 34 98 99
> a=rtpmap:34 H263/90000
> a=rtpmap:98 h263-1998/90000
> a=rtpmap:99 H264/90000
> a=sendrecv
>
>
> Please note that in sip.conf we have set:
>
> trustrpid = yes
> sendrpid = yes
>
> Any input will be appreciated!
>
> Regards,
>
> -effie
>
> --
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Mit freundlichen Grüssen
--
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Teamleiter VOIP // [email protected] // Tel 059944-2440//
-------------------------------------------------
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