[asterisk-dev] [Code Review]: Restore SIP DTMF overlap dialing method.

[Tilghman Lesher]

> On Nov. 14, 2011, 9 a.m., mjordan wrote:
> > /branches/1.8/channels/chan_sip.c, line 15185
> > <https://reviewboard.asterisk.org/r/1582/diff/2/?file=21719#file21719line15185>
> >
> >     This should probably be strncmp

Why?  If the pickup extension is *8, and the phone user dialed *81, I don't think that should be treated as equivalent.


- Tilghman


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On Nov. 11, 2011, 6:31 p.m., rmudgett wrote:
> 
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> 
> (Updated Nov. 11, 2011, 6:31 p.m.)
> 
> 
> Review request for Asterisk Developers, mjordan and Pavel Troller.
> 
> 
> Summary
> -------
> 
> See ASTERISK-18702 it has a very good description of the issue.
> 
> Basically, the recent fix for ASTERISK-17288 to get RFC3578 SIP overlap support working correctly removed a long standing ability to do overlap dialing using DTMF in the early media phase of a call.
> 
> The large change block for PAGE2 flags is just to make room for the now two bit field SIP_PAGE2_ALLOWOVERLAP.
> 
> Also:
> 
> * Renames SIP_GET_DEST_PICKUP_EXTEN_FOUND to SIP_GET_DEST_EXTEN_MATCHMORE since that is what it really means.
> 
> * Fixed get_destination() inconsistency with the pickup extension matching.
> 
> * Fixed initialization of PAGE3 of global_flags in reload_config().
> 
> 
> This addresses bugs ASTERISK-17288 and ASTERISK-18702.
>     https://issues.asterisk.org/jira/browse/ASTERISK-17288
>     https://issues.asterisk.org/jira/browse/ASTERISK-18702
> 
> 
> Diffs
> -----
> 
>   /branches/1.8/channels/sip/include/sip.h 344897 
>   /branches/1.8/configs/sip.conf.sample 344897 
>   /branches/1.8/channels/chan_sip.c 344897 
>   /branches/1.8/CHANGES 344897 
> 
> Diff: https://reviewboard.asterisk.org/r/1582/diff
> 
> 
> Testing
> -------
> 
> It compiles. :)
> 
> 
> Thanks,
> 
> rmudgett
> 
>

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