Thanks Matt, I think I found what is going wrong , but Before that I want to confirm and Can you please verify what I am doing below is correct -
During Record using agi handle_recordfile and When there is a disconnect while in ast_waitstream , I can see the ast_closestream is initiated only channel.c during call cleanup to close the stream opened for beep . can I call the ast_closestream in agi handle_recordfile itself if I get return response as -1 to close the filestream opened for beep, I hope this shouldn't affect the flow thanks, bala On Wed, May 21, 2014 at 10:55 PM, Matthew Jordan <[email protected]> wrote: > On Wed, May 21, 2014 at 7:33 PM, bala murugan <[email protected]> > wrote: > > > > > > > > On Wed, May 21, 2014 at 8:23 PM, bala murugan <[email protected]> > wrote: > >> > >> HI , > >> > >> has any one noticed in res_agi.c handle_recordfile , if there is a > >> call disconnect while beep is played back to caller the beep filestream > is > >> not getting closed and it leads to FileDescriptor Leak. This is still > there > >> in asterisk 12 . I have a Fix and Let me know I can submit the same . > >> > >> thanks, > > > > Nope, that one would be new as far as I'm aware of. I'd be interested > in seeing that patch, as the play back of the beep sound file is > handled by ast_streamfile, which is used rather extensively. > > If you'd like to submit a patch up stream to the project, please open > an issue on the issue tracker [1], sign a license contributor > agreement [2], and attach the patch in unified diff format to the > issue. You can find more information on submitting a patch to Asterisk > on the Asterisk wiki [3]. > > [1] https://issues.asterisk.org > > [2] https://wiki.asterisk.org/wiki/display/AST/Digium+License+Agreement > > [3] https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process > > Thanks! > > Matt > > -- > Matthew Jordan > Digium, Inc. | Engineering Manager > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > Check us out at: http://digium.com & http://asterisk.org > > -- > _____________________________________________________________________ > -- 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 >
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