[asterisk-dev] [Code Review] Fix Dial m and r options generating warnings for voice frames.

[Mark Michelson]

-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/1805/#review5770
-----------------------------------------------------------



/branches/1.8/apps/app_dial.c
<https://reviewboard.asterisk.org/r/1805/#comment10560>

    This entire case is impossible to enter. The else that encloses this switch statement only happens if the frame read is NOT an AST_FRAME_CONTROL. You need to move the special "caller_entertained" exceptions for VIDUPDATE, SRCUPDATE, and SRCCHANGE up into the if block above this else. Currently VIDUPDATE and SRCUPDATE are passed along without checking any flags and SRCCHANGE is not handled at all.


- Mark


On March 7, 2012, 4:10 p.m., rmudgett wrote:
> 
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/1805/
> -----------------------------------------------------------
> 
> (Updated March 7, 2012, 4:10 p.m.)
> 
> 
> Review request for Asterisk Developers and Mark Michelson.
> 
> 
> Summary
> -------
> 
> Incoming calls that do not use ulaw will generate a bunch of warning messages and possibly fail when a DAHDI channel is then called if the Dial m or r options are used.
> 
> You get:
> [Mar 6 20:39:05] WARNING[2994]: chan_dahdi.c:9061 dahdi_write: Cannot handle frames in g722 format
> [Mar 6 20:39:05] WARNING[2994]: app_dial.c:1410 wait_for_answer: Unable to forward voice or dtmf
> 
> A bandaid patch was applied to silence the above warnings for each frame by setting up a translation path for future frames.  The bandaid patch also emits this warning:
> Codec mismatch on channel %s setting write format to %s from %s native formats %s
> 
> This problem also occurs if the dial is forked to multiple destinations: Dial(DAHDI/1&DAHDI/2)
> 
> 
> This addresses bugs ASTERISK-16901 and ASTERISK-17541.
>     https://issues.asterisk.org/jira/browse/ASTERISK-16901
>     https://issues.asterisk.org/jira/browse/ASTERISK-17541
> 
> 
> Diffs
> -----
> 
>   /branches/1.8/apps/app_dial.c 358576 
>   /branches/1.8/main/channel.c 358576 
> 
> Diff: https://reviewboard.asterisk.org/r/1805/diff
> 
> 
> Testing
> -------
> 
> Without patch, the codecs are not setup when Dial options m or r are used and the SIP codec is not ulaw.
> 
> With the patch, the voice frames are not passed and the diagnostic messages are not generated.
> 
> 
> Thanks,
> 
> rmudgett
> 
>

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20120308/fcb715e8/attachment-0001.htm>