asterisk/channels/sip
Alec L Davis 0ccc1f5274 Merged revisions 353321 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r353321 | alecdavis | 2012-01-31 11:16:22 +1300 (Tue, 31 Jan 2012) | 25 lines
  
  Merged revisions 353320 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
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    r353320 | alecdavis | 2012-01-31 10:57:49 +1300 (Tue, 31 Jan 2012) | 18 lines
    
    RFC3261 Section 8.1.1.5. The sequence number value MUST be expressible as a 32-bit unsigned integer
    
    * fix: use %u instead of %d when dealing with CSeq numbers - to remove possibility of -ve numbers.
    
    * fix: change all uses of seqno and friends (ocseq icseq) from 'int' or 'unsigned int' to uint32_t.
    
    Summary of CSeq numbers.
    An initial CSeq number must be less than 2^31
    A CSeq number can increase in value up to 2^32-1
    An incrementing CSeq number must not wrap around to 0.
    
    Tested with Asterisk 1.8.8.2 with Grandstream phones.
     
    alecdavis (license 585)
    Tested by: alecdavis
     
    Review: https://reviewboard.asterisk.org/r/1699/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-30 22:28:37 +00:00
..
include Merged revisions 353321 via svnmerge from 2012-01-30 22:28:37 +00:00
config_parser.c Set port to a default sane value if a bogus one is provided when parsing hostnames. 2012-01-13 20:32:19 +00:00
dialplan_functions.c Merged revisions 310088 via svnmerge from 2011-03-08 20:34:05 +00:00
reqresp_parser.c Misc minor fixes in reqresp_parser.c and chan_sip.c. 2012-01-19 23:31:17 +00:00
sdp_crypto.c Merged revisions 336936 via svnmerge from 2011-09-20 16:56:11 +00:00
security_events.c Merged revisions 337595,337597 via svnmerge from 2011-09-22 16:35:20 +00:00
srtp.c Add SRTP support for Asterisk 2010-06-08 05:29:08 +00:00