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

David Vossel dvossel at digium.com
Mon Oct 19 09:58:10 CDT 2009


-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/408/
-----------------------------------------------------------

Review request for Asterisk Developers.


Summary
-------

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


Diffs
-----

  /trunk/channels/chan_iax2.c 224490 
  /trunk/channels/chan_sip.c 224490 
  /trunk/configs/iax.conf.sample 224490 
  /trunk/configs/sip.conf.sample 224490 

Diff: https://reviewboard.asterisk.org/r/408/diff


Testing
-------

tested 'shrinkcallerid' option in both chan_sip and chan_iax


Thanks,

David




More information about the asterisk-dev mailing list