Paris Stamatopoulos
2012-Jan-19 09:10 UTC
[asterisk-users] Asterisk rewrites "From" header when CALLERID(num-pres)=prohib_passed_screen is set
Hello all, We are testing Asterisk 1.8.8.1. In the following scenario, peer 54321 dials 12345: INVITE sip:12345 at 10.1.1.88 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:54321 at 10.1.1.86>;tag=5ebe58983f6c0c84o3 To: <sip:12345 at 10.1.1.86> Call-ID: adc7c928-b6f6d534 at 10.1.1.86 CSeq: 102 INVITE Max-Forwards: 69 Contact: "54321" <sip:54321 at 10.1.1.86:5060> 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:asterisk at 10.1.1.88>) instead of keeping it intact. The "Remote-Party-ID" on the other hand, is correct. INVITE sip:12345 at 10.1.1.87:5061 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.88:5060;branch=z9hG4bK25e11cb8 Max-Forwards: 70 From: "asterisk" <sip:asterisk at 10.1.1.88>;tag=as6d2aa852 To: <sip:12345 at 10.1.1.87:5061> Contact: <sip:asterisk at 10.1.1.88:5060> Call-ID: 54deebbf2bb308740e6b9ca817e693a9 at 10.1.1.88:5060 CSeq: 103 INVITE User-Agent: Asterisk Authorization: Digest username="asterisk", realm="10.1.1.87", algorithm=MD5, uri="sip:12345 at 10.1.1.87:5061", 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:54321 at 10.1.1.88>;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
Stefan Schmidt
2012-Jan-19 10:12 UTC
[asterisk-users] Asterisk rewrites "From" header when CALLERID(num-pres)=prohib_passed_screen is set
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:12345 at 10.1.1.88 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:54321 at 10.1.1.86>;tag=5ebe58983f6c0c84o3 > To: <sip:12345 at 10.1.1.86> > Call-ID: adc7c928-b6f6d534 at 10.1.1.86 > CSeq: 102 INVITE > Max-Forwards: 69 > Contact: "54321" <sip:54321 at 10.1.1.86:5060> > 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:asterisk at 10.1.1.88>) instead of keeping it intact. The "Remote-Party-ID" on the other hand, is correct. > > INVITE sip:12345 at 10.1.1.87:5061 SIP/2.0 > Via: SIP/2.0/UDP 10.1.1.88:5060;branch=z9hG4bK25e11cb8 > Max-Forwards: 70 > From: "asterisk" <sip:asterisk at 10.1.1.88>;tag=as6d2aa852 > To: <sip:12345 at 10.1.1.87:5061> > Contact: <sip:asterisk at 10.1.1.88:5060> > Call-ID: 54deebbf2bb308740e6b9ca817e693a9 at 10.1.1.88:5060 > CSeq: 103 INVITE > User-Agent: Asterisk > Authorization: Digest username="asterisk", realm="10.1.1.87", algorithm=MD5, uri="sip:12345 at 10.1.1.87:5061", 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:54321 at 10.1.1.88>;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 > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users-- F?r weitere Fragen stehen wir gerne unter voip at sil.at oder 059944 - 2440 zur Verf?gung. Mit freundlichen Gr?ssen -- Stefan Schmidt Teamleiter VOIP // voip at sil.at // Tel 059944-2440// ------------------------------------------------- SILVER SERVER GmbH // Lorenz-Mandl-Gasse 33/1 // A-1160 Wien // Fax 059944-9000 // www.sil.at // -------------------------------------------------