[asterisk-bugs] [Asterisk 0015940]: [patch] IAX does not allow CALLERID(num) contain non-numbers.

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=15940 
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Reported By:                dimas
Assigned To:                
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Project:                    Asterisk
Issue ID:                   15940
Category:                   Channels/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     confirmed
Asterisk Version:           SVN 
JIRA:                       SWP-225 
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2009-09-22 19:40 CDT
Last Modified:              2009-10-06 17:24 CDT
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Summary:                    [patch] IAX does not allow CALLERID(num) contain
non-numbers.
Description: 
( I already explained the issue here:
http://lists.digium.com/pipermail/asterisk-dev/2009-August/039257.html )

The thing is that IAX channel driver "shrinks" (ast_shrink_phone_number)
caller number. Which was probably the good thing before. However nowadays
when you have Skype support integrated, skype names passed via
CALLERID(num) get damaged.

I believe that only drivers which know they can not pass arbitrary number
(like DAHDI for example) should do shrinking of number before sending out.
While inside Asterisk (or relayed to another Asterisk with SIP/IAX) we
should presever original number.

Obviously I can create a patch for chan_iax which removes the shrinking
because this patch is trivial. However issue may be more global and affect
other drivers so I believe it is better to hear from Digium guys first -
maybe they have another opinion about how to solve that issue or do not
consider it an issue at all :)
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 (0111940) dimas (reporter) - 2009-10-06 17:24
 https://issues.asterisk.org/view.php?id=15940#c111940 
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Spent some time tracing these number shrinks into the past. It seems
Asterisk was doing shrinking of SIP numbers from the very beginning (SVN
revisions <1000).

Also I think it is a good idea if oej take a look because he was working
on https://issues.asterisk.org/view.php?id=6450 which is related. 

Issue History 
Date Modified    Username       Field                    Change               
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2009-10-06 17:24 dimas          Note Added: 0111940                          
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