[asterisk-dev] [Code Review]: Extend SIP REFER message originated by Transfer with SIPAddHeader-added headers

[Marquis]

> On May 3, 2012, 1:27 p.m., Terry Wilson wrote:
> > /trunk/channels/chan_sip.c, line 12905
> > <https://reviewboard.asterisk.org/r/1159/diff/2/?file=16051#file16051line12905>
> >
> >     Since this is really just a global option, why add it to the refer instead of just always checking global_refer_addheaders?

Looking at this more carefully, it's obvious the submitter was simply doing someting similar to what was already done with INVITEs.  However, I think the same criticism applies there as well.  So I think it would be better to just check the global and get rid of the addition to the refer struct.


- Marquis


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On April 3, 2011, 9:36 p.m., kkm wrote:
> 
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> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/1159/
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> 
> (Updated April 3, 2011, 9:36 p.m.)
> 
> 
> Review request for Asterisk Developers, Russell Bryant and Olle E Johansson.
> 
> 
> Summary
> -------
> 
> There is currently no way to augment a REFER message from Transfer with extra headers. The attached patch implements the feature.
> 
> The feature is enabled by default, with a new sip.conf setting to disable it. The rationale for it to be on by default is that it is in fact very controllable from the dialplan: It takes one application call to SIPRemoveHeader with no argument to remove all previously accumulated additional SIP headers. Since Transfer normally terminates the channel, there is no need in practice to keep any SIP headers beyond it in the channel, so that removing these does not impose any dialplan programming complexity.
> 
> A hunk near chan_sip.c line 11697 also fixes an issue with Refer-To header gaining an extra set of <> around the address only when retransmitted due to an authentication request.
> 
> 
> This addresses bug 19059.
>     https://issues.asterisk.org/jira/browse/19059
> 
> 
> Diffs
> -----
> 
>   /trunk/CHANGES 312554 
>   /trunk/channels/chan_sip.c 312554 
>   /trunk/channels/sip/include/sip.h 312554 
>   /trunk/configs/sip.conf.sample 312554 
> 
> Diff: https://reviewboard.asterisk.org/r/1159/diff
> 
> 
> Testing
> -------
> 
> Confirmed working per spec with sip set debug on packet dump.
> 
> Deployed and used on a production server under 1.8.3 and working for 2 weeks already.
> 
> 
> Thanks,
> 
> kkm
> 
>

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