[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@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@10.1.1.86;tag=5ebe58983f6c0c84o3
To: sip:12345@10.1.1.86
Call-ID: adc7c928-b6f6d534@10.1.1.86
CSeq: 102 INVITE
Max-Forwards: 69
Contact: 54321 sip:54321@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@10.1.1.88) instead of keeping it intact. The Remote-Party-ID
on the other hand, is correct.
INVITE sip:12345@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@10.1.1.88;tag=as6d2aa852
To: sip:12345@10.1.1.87:5061
Contact: sip:asterisk@10.1.1.88:5060
Call-ID: 54deebbf2bb308740e6b9ca817e693a9@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@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@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 H263/9
a=rtpmap:98 h263-1998/9
a=rtpmap:99 H264/9
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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Re: [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@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@10.1.1.86;tag=5ebe58983f6c0c84o3
To: sip:12345@10.1.1.86
Call-ID: adc7c928-b6f6d534@10.1.1.86
CSeq: 102 INVITE
Max-Forwards: 69
Contact: 54321 sip:54321@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@10.1.1.88) instead of keeping it intact. The Remote-Party-ID
on the other hand, is correct.
INVITE sip:12345@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@10.1.1.88;tag=as6d2aa852
To: sip:12345@10.1.1.87:5061
Contact: sip:asterisk@10.1.1.88:5060
Call-ID: 54deebbf2bb308740e6b9ca817e693a9@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@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@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 H263/9
a=rtpmap:98 h263-1998/9
a=rtpmap:99 H264/9
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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Re: [asterisk-users] Asterisk rewrites From header when CALLERID(num-pres)=prohib_passed_screen is set
When the date was Thu Jan 19 2012 12:12:04, Stefan Schmidt wrote: 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? This behaviour is new in 1.8, since in 1.6 it work differently (not forcing the From header to the default asterisk caller ID). Does the RFC say that a UAS must change the From header to secure privacy, even though there is a special header for privacy (or set privacy=full in Remote-Party-ID)? In case asterisk is not functioning as a PBX but as intermediate proxy, how do we secure its interoperability with other systems? So the issue here is that we are unable to change/keep the From header when CALLERID(num-pres)=prohib_passed_screen is set. 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 http://tools.ietf.org/html/draft-ietf-sip-privacy-00: 6.2 UAS Behavior A UAS supporting this extension and receiving an INVITE from its trusted proxy looks for a Remote-Party-ID header field to identify the originator of the request. If the Remote-Party-ID contains an rpi-screen parameter with a value of no, the UAS SHOULD NOT trust the validity of the information provided. Otherwise, the UAS SHOULD use the information provided to identify the caller rather than any information provided in the From header field. So even if asterisk doesn't support the Privacy header, it supports the Remote-Party-ID header where the privacy=full parameter is present. 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. Kind 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
Re: [asterisk-users] Asterisk rewrites From header when CALLERID(num-pres)=prohib_passed_screen is set
On 01/19/2012 05:56 AM, effie mouzeli wrote: When the date was Thu Jan 19 2012 12:12:04, Stefan Schmidt wrote: 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? This behaviour is new in 1.8, since in 1.6 it work differently (not forcing the From header to the default asterisk caller ID). Does the RFC say that a UAS must change the From header to secure privacy, even though there is a special header for privacy (or set privacy=full in Remote-Party-ID)? In case asterisk is not functioning as a PBX but as intermediate proxy, how do we secure its interoperability with other systems? So the issue here is that we are unable to change/keep the From header when CALLERID(num-pres)=prohib_passed_screen is set. Asterisk cannot act as a proxy, it is a B2BUA. If you want to make its behavior *appear* to be a proxy, there are a number of things you can do, but it will never just 'pass along' headers from an incoming INVITE to an outgoing INVITE. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- 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
Re: [asterisk-users] Asterisk rewrites From header when CALLERID(num-pres)=prohib_passed_screen is set
When the date was Thu Jan 19 2012 15:23:04, Kevin P. Fleming wrote: On 01/19/2012 05:56 AM, effie mouzeli wrote: When the date was Thu Jan 19 2012 12:12:04, Stefan Schmidt wrote: 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? This behaviour is new in 1.8, since in 1.6 it work differently (not forcing the From header to the default asterisk caller ID). Does the RFC say that a UAS must change the From header to secure privacy, even though there is a special header for privacy (or set privacy=full in Remote-Party-ID)? In case asterisk is not functioning as a PBX but as intermediate proxy, how do we secure its interoperability with other systems? So the issue here is that we are unable to change/keep the From header when CALLERID(num-pres)=prohib_passed_screen is set. Asterisk cannot act as a proxy, it is a B2BUA. If you want to make its behavior *appear* to be a proxy, there are a number of things you can do, but it will never just 'pass along' headers from an incoming INVITE to an outgoing INVITE. This is understandable. On the other hand, I cannot understand why asterisk sets the From header to asterisk, even if we explicitly set the CALLERID(name) and CALLERID(num). -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
