[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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