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

Olle E Johansson reviewboard at asterisk.org
Wed Dec 5 05:34:53 CST 2012


-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/2227/
-----------------------------------------------------------

Review request for Asterisk Developers.


Summary
-------

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


This addresses bug ASTERISK-20759.
    https://issues.asterisk.org/jira/browse/ASTERISK-20759


Diffs
-----

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

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


Testing
-------

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

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20121205/cb8fbafa/attachment.htm>


More information about the asterisk-dev mailing list