[asterisk-dev] [Code Review] Add option to prevent SIP diversion headers from being sent

[Matt Jordan]

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Update CHANGES as well.


/trunk/configs/sip.conf.sample
<https://reviewboard.asterisk.org/r/1769/#comment10261>

    Don't put the Asterisk version number in here



/trunk/configs/sip.conf.sample
<https://reviewboard.asterisk.org/r/1769/#comment10262>

    Don't put an issue number in here


- Matt


On Feb. 23, 2012, 10:05 a.m., jrose wrote:
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> (Updated Feb. 23, 2012, 10:05 a.m.)
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> 
> Review request for Asterisk Developers and Mark Michelson.
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> Summary
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> A while back, when 1.6.3 was a thing, A largish patch was introduced that enabled SIP diversion headers to be sent, presumably for call forwarding.  This adds an option to keep diversion headers introduced with that patch from being added to SIP dialogs because of interoperability issues that were introduced with that support.
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> This addresses bug ASTERISK-16862.
>     https://issues.asterisk.org/jira/browse/ASTERISK-16862
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> 
> Diffs
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>   /trunk/channels/chan_sip.c 355731 
>   /trunk/channels/sip/include/sip.h 355731 
>   /trunk/configs/sip.conf.sample 355731 
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> Diff: https://reviewboard.asterisk.org/r/1769/diff
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> Testing
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> I actually couldn't find a useful test scenario.  Call forwarding is a pretty roughly defined feature that tends to be implemented in the dialplan, and I couldn't find a method for call forwarding that actually included the Diversion header.  Still, the approach is really simple.  I don't think it'll be a problem.
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> Thanks,
> 
> jrose
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>

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