[asterisk-dev] [Code Review] SIP/IAX2 'shrinkcallerid' option

Russell Bryant russell at digium.com
Mon Oct 19 18:35:06 CDT 2009


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Ship it!


This looks good to me.

Also, I would like to leave it as you have it to maintain existing behavior unless this option is used.  This behavior is ancient, and I'd rather not introduce any surprises to existing users.

- Russell


On 2009-10-19 09:58:09, David Vossel wrote:
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> (Updated 2009-10-19 09:58:09)
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> 
> Review request for Asterisk Developers.
> 
> 
> Summary
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> The shrinking of caller id removes '(', ' ', ')', non-trailing '.', and '-' from the string.  This means values such as 555.5555 and test-test result in 555555 and testtest.  There are instances, such as Skype integration, where a specific value is passed via caller id that must be preserved unmodified.  This patch makes the shrinking of caller id optional in chan_sip and chan_iax in order to support such cases.  By default this option is on to preserve previous expected behavior.
> 
> I don't know the history or purpose of why we shrink caller id values to begin with, perhaps this behavior can be deprecated in the future.
> 
> 
> This addresses bug 15940.
>     https://issues.asterisk.org/view.php?id=15940
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> 
> Diffs
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>   /trunk/channels/chan_iax2.c 224490 
>   /trunk/channels/chan_sip.c 224490 
>   /trunk/configs/iax.conf.sample 224490 
>   /trunk/configs/sip.conf.sample 224490 
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> Diff: https://reviewboard.asterisk.org/r/408/diff
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> 
> Testing
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> tested 'shrinkcallerid' option in both chan_sip and chan_iax
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> 
> Thanks,
> 
> David
> 
>




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