[asterisk-dev] [Code Review] 2227: Manage translation table between SIP and ISDN hangup causes

[Olle E Johansson]

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This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/2227/
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(Updated April 11, 2014, 4:57 p.m.)


Review request for Asterisk Developers.


Changes
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Adding alert of new version.


Bugs: ASTERISK-20759
    https://issues.asterisk.org/jira/browse/ASTERISK-20759


Repository: Asterisk


Description (updated)
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The SIP2CAUSE hangup code conversion tables has up to now been hard-coded in Asterisk. In some cases, like when building in-house ISDN/Q.SIG to SIP gateways, there's a need to manipulate this conversion. 

With this code, advanced users can add a "private" conversion. This is added in front of the built-in conversions.

Asterisk conversion tables does not change in this patch. Everything should work as before. To shrink the chan_sip.c file a small bit I decided to move this functionality into a new source code file.

Adding:
- new source code file sip2cause.c and include file sip2cause.h
- new configuration file sip2cause.conf

Reviewboard doesn't seem accept the new files, so they have to be found in the branch itself.

http://svn.digium.com/svn/asterisk/team/oej/earl-grey-sip2cause-configurable-trunk

The new files are:
* http://svnview.digium.com/svn/asterisk/team/oej/earl-grey-sip2cause-configurable-trunk/configs/sip2cause.conf.sample
* http://svnview.digium.com/svn/asterisk/team/oej/earl-grey-sip2cause-configurable-trunk/channels/sip/sip2cause.c
* http://svnview.digium.com/svn/asterisk/team/oej/earl-grey-sip2cause-configurable-trunk/channels/sip/include/sip2cause.h


2014-04: A new version will be coming soon with a new function - custom hangupcauses outside of the ISDN range (as discussed on asterisk-dev a while ago).


Diffs
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  /trunk/channels/sip/include/sip_utils.h 377205 
  /trunk/channels/chan_sip.c 377205 

Diff: https://reviewboard.asterisk.org/r/2227/diff/


Testing
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Tested all kinds of weird translations. This file should cause some errors (AST_CAUSE_SKREP doesn't exist, 903 is not a valid SIP reason code etc etc. 

[sip2cause]
604 => AST_CAUSE_SKREP
404 => UNALLOCATED
599 Bad => USER_BUSY
486 => NORMAL_CLEARING
603 => UNALLOCATED
        
[cause2sip]
SKREP => 503 Service Failure
UNALLOCATED => 903 Go to hell
UNALLOCATED => 499 I don't want to do that.
USER_BUSY => 503 I am not feeling well


Thanks,

Olle E Johansson

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