[asterisk-dev] [Code Review] Fix directed group pickup feature code *8 with pickupsounds enabled , deadlock and segfault, affects 1.8.0 and trunk

rmudgett reviewboard at asterisk.org
Fri May 13 19:32:07 CDT 2011



> On 2011-05-13 18:10:22, Alec Davis wrote:
> > trunk/main/features.c, lines 5740-5742
> > <https://reviewboard.asterisk.org/r/1185/diff/16/?file=16493#file16493line5740>
> >
> >     the locks around this stream_and_wait need to go. They stopped the choppy audio, which happens with DAHDI and SIP, seems as though it's an unrelated issue, induced by Pickup.
> >     
> >     Using tt_monkeys to exagerate to problem, no other channels are able to dial out.

The locks are needed because we are injecting the pickupsound outside the thread running the bridge.  Without the locks on my setup, the calls got disconnected because chan_dahdi complained about incompatible audio frames.  We need to get the bridge to inject the pickupsound for us somehow.

A follow-up patch is needed.


- rmudgett


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On 2011-05-13 18:09:58, Alec Davis wrote:
> 
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/1185/
> -----------------------------------------------------------
> 
> (Updated 2011-05-13 18:09:58)
> 
> 
> Review request for Asterisk Developers and rmudgett.
> 
> 
> Summary
> -------
> 
> Since 1.8, the new pickupsound and pickupfailsound in features.conf cause many issues.
> 
> 1). chan_sip:handle_request_invite() shouldn't be playing out the fail/success audio, as it has 'netlock' locked.
> 2). dialplan applications for directed_pickups shouldn't beep.
> 3). feature code for directed pickup should beep on success/failure if configured.
> 
> Moved app_directed:pickup_do() to features:ast_do_pickup().
> 
> Functions below, all now use the new ast_do_pickup()
> app_directed_pickup.c:
>    pickup_by_channel()
>    pickup_by_exten()
>    pickup_by_mark()
>    pickup_by_part()
> features.c:
>    ast_pickup_call()
> 
> Created a sip_pickup() thread to handle the pickup and playout the audio, spawned from handle_request_invite.
> 
> 
> This addresses bug 18654.
>     https://issues.asterisk.org/view.php?id=18654
> 
> 
> Diffs
> -----
> 
>   trunk/apps/app_directed_pickup.c 317665 
>   trunk/channels/chan_sip.c 317665 
>   trunk/include/asterisk/features.h 317665 
>   trunk/main/features.c 317665 
> 
> Diff: https://reviewboard.asterisk.org/r/1185/diff
> 
> 
> Testing
> -------
> 
> pickup using *8 feature code, with pickup sounds enabled/disabled
> 
> exten => 71,1,Pickup()           ; any ringing extension in same pickupgroup 
> exten => 72,1,Pickup(85 at phones)  ; dahdi extension
> exten => 73,1,Pickup(823 at phones) ; sip extension
> 
> 
> Thanks,
> 
> Alec
> 
>

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