[asterisk-dev] [Code Review] Fix SIP connected line updates.

[Mark Michelson]

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/branches/1.8/channels/chan_sip.c
<https://reviewboard.asterisk.org/r/1199/#comment7074>

    Can you explain this if statement? It reads a bit weird. This reads to me like you're going to update the connected number if either there is a cid_num set for p or if the calling presentation is not allowed. It seems like it should only be updating the connected number if there is a cid_num set for p AND the calling presentation IS allowed.
    
    I'm probably just misinterpreting it, but a comment in the code would help to make it less confusing.


- Mark


On 2011-04-26 17:49:41, rmudgett wrote:
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> This is an automatically generated e-mail. To reply, visit:
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> (Updated 2011-04-26 17:49:41)
> 
> 
> Review request for Asterisk Developers and Mark Michelson.
> 
> 
> Summary
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> This patch fixes a couple connected line update problems:
> 
> 1) The connected line needs to be updated when the initial INVITE is sent
> if there is a peer callerid configured.  Previously, the connected line
> information did not get reported until the call was connected so SIP could
> not report connected line information in ringing or progress messages.
> 
> 2) The connected line should not be updated on initial connect if there is
> no connected line information.  Previously, all it did was wipe out any
> default preset CONNECTEDLINE information set by the dialplan with empty
> strings.
> 
> 
> This addresses bug 18367.
>     https://issues.asterisk.org/view.php?id=18367
> 
> 
> Diffs
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>   /branches/1.8/channels/chan_sip.c 315663 
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> Diff: https://reviewboard.asterisk.org/r/1199/diff
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> 
> Testing
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> Both issues now work as intended.
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> 
> Thanks,
> 
> rmudgett
> 
>

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