[asterisk-dev] [Code Review] SIP: more code refactoring, more unit tests
David Vossel
dvossel at digium.com
Thu Feb 11 12:05:42 CST 2010
> On 2010-02-11 11:11:17, Nick Lewis wrote:
> > /trunk/channels/chan_sip.c, line 16505
> > <https://reviewboard.asterisk.org/r/499/diff/1/?file=8083#file8083line16505>
> >
> > if (get_name_and_number(...)) {
> > ast_log(...failed parsing to-header...)
> > }
> >
this is done within get_name_and_number
- David
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https://reviewboard.asterisk.org/r/499/#review1493
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On 2010-02-11 10:42:22, David Vossel wrote:
>
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/499/
> -----------------------------------------------------------
>
> (Updated 2010-02-11 10:42:22)
>
>
> Review request for Asterisk Developers.
>
>
> Summary
> -------
>
> ----- Code Refactoring Changes ----
> - read_to_parts() moved to reqresp_parser.c and has been renamed as get_name_and_number()
> - get_in_brackets() moved to reqresp_parser.c
> - find_closing_quotes() added to sip_utils.h
>
> ----- Logic Changes -----
> - get_name_and_number() now uses parse_uri() and get_calleridname() for parsing. Before this change only names within quotes were found, when names not within quotes are possible.
>
> ----- New Unit Tests -----
> sip_get_name_and_number_test
> sip_get_in_brackets_test
>
>
> Diffs
> -----
>
> /trunk/channels/chan_sip.c 246298
> /trunk/channels/sip/include/reqresp_parser.h 246298
> /trunk/channels/sip/include/sip_utils.h 246298
> /trunk/channels/sip/reqresp_parser.c 246298
>
> Diff: https://reviewboard.asterisk.org/r/499/diff
>
>
> Testing
> -------
>
> -tested sip redirect to verify get_name_and_number still works as expected.
> -ran unit tests, verified they all passed.
>
>
> Thanks,
>
> David
>
>
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