-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/4256/#review14078
-----------------------------------------------------------



/asterisk/trunk/tests/bridge/atxfer_fail_blonde/test-config.yaml
<https://reviewboard.asterisk.org/r/4256/#comment24592>

    This isn't a chan_sip only bug right?  If not, I'd recommend using 
chan_pjsip (looking to the future)
    
    [this probably doesn't really matter much in the current grand scheme of 
things or for this issue, but thought I'd mention just in case]


Is this missing a sip.conf?

- Kevin Harwell


On Dec. 31, 2014, 12:01 p.m., Scott Griepentrog wrote:
> 
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/4256/
> -----------------------------------------------------------
> 
> (Updated Dec. 31, 2014, 12:01 p.m.)
> 
> 
> Review request for Asterisk Developers.
> 
> 
> Bugs: ASTERISK-24513
>     https://issues.asterisk.org/jira/browse/ASTERISK-24513
> 
> 
> Repository: testsuite
> 
> 
> Description
> -------
> 
> This test starts an attended transfer, converts to blonde mode by hanging up 
> the transferer, and then fails the transfer by hanging up the transferee.  
> Then after allowing the recall attempt to complete, checks to insure that 
> there are not leaked channels.
> 
> Improvements to channel_test_condition: count the actual channels listed in 
> "core show channels" output to check for leaks.  Also added unittest.
> 
> 
> Diffs
> -----
> 
>   /asterisk/trunk/tests/bridge/tests.yaml 6149 
>   /asterisk/trunk/tests/bridge/atxfer_fail_blonde/test-config.yaml 
> PRE-CREATION 
>   
> /asterisk/trunk/tests/bridge/atxfer_fail_blonde/configs/ast1/extensions.conf 
> PRE-CREATION 
>   /asterisk/trunk/lib/python/asterisk/channel_test_condition.py 6149 
> 
> Diff: https://reviewboard.asterisk.org/r/4256/diff/
> 
> 
> Testing
> -------
> 
> Currently fails while ASTERISK-24513 is not yet patched.
> 
> 
> Thanks,
> 
> Scott Griepentrog
> 
>

-- 
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-dev

Reply via email to