[asterisk-dev] [Code Review] rfc 4474 patch
Russell Bryant
russell at digium.com
Tue Mar 30 14:25:43 CDT 2010
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This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/596/
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(Updated 2010-03-30 14:25:43.506337)
Review request for Asterisk Developers, Russell Bryant and Olle E Johansson.
Changes
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Trivial update, getting this to show up on -dev
Summary (updated)
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This is a patch to trunk which implements the subset of RFC 4474 suitable for
asterisk calling number authentication. It can sign SIP INVITEs and verify the
signatures of incoming INVITES. The current implementation implements much
of the RFC. The result of the signing and validation is placed in a channel
variable ( IDENTITY_RESULT )
for dialplan interpretation. This allows the dialplan to authenticate the
CallerID and prevent SPIT.
The private keys for signing are taken from
a local directory, a cache, or an http server. The private key is used
to sign a digest string. The certificate is retrieved from the location
specified in the Identity-Info header, or from the cache, if present.
SSL is not currently supported.
This addresses bug 0016187.
https://issues.asterisk.org/view.php?id=0016187
Diffs
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/trunk/channels/chan_sip.c 255321
/trunk/channels/sip/include/sip.h 255321
/trunk/configs/extensions.conf.sample 255321
/trunk/configs/sip.conf.sample 255321
Diff: https://reviewboard.asterisk.org/r/596/diff
Testing
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This patch has been tested between multiple asterisk instances and
with in-charges reference system which runs openser.
Thanks,
Ed
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