asterisk users - Dec 2008 - async agi question

Hi,

I am developing asterisk support for our application using the Async AGI
and Asterisk-Java.
One thing I haven't been able to implement is how to stop playing a
sound. Something similar to StopIO for Dialogic GlobalCall or
DivaStopSending for Eicon.
Is there any way to achieve this today which I have missed? Or could
someone give me hints on how I could implement this in the res_agi.c The
command "asyncagi break" does stop ongoing playing but also breaks the
async agi control. I only want the first.

Thanks in advance,
/Henrik

Hello Henrik,

I have not used Asterisk from a user perspective lately, but, when I
added the async agi functionality, I used to control this using a
"manager redirect" action to the same priority where the channel calls
async agi, that will work like a break that re-enters the async agi
loop . This, of course, requires you to save the state of the channel
somehow in your program to "remember" that the next time that channel
calls async agi the sound was already played and such.

http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+Redirect

Let me know if that does not work for you and we can probably write
something in res_agi.c

Moy

On Fri, Dec 5, 2008 at 3:01 AM, Henrik Westerberg
<henrik.westerberg at ain.se> wrote:
> Hi,
>
> I am developing asterisk support for our application using the Async AGI
> and Asterisk-Java.
> One thing I haven't been able to implement is how to stop playing a
> sound. Something similar to StopIO for Dialogic GlobalCall or
> DivaStopSending for Eicon.
> Is there any way to achieve this today which I have missed? Or could
> someone give me hints on how I could implement this in the res_agi.c The
> command "asyncagi break" does stop ongoing playing but also
breaks the
> async agi control. I only want the first.
>
> Thanks in advance,
> /Henrik
>
>
>
>
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-- 
"I do not agree with what you have to say, but I'll defend to the
death your right to say it." Voltaire

Hi Moy,

thank you for your answer and this new testasync2.diff. It matches better my
mind about the subject.

I tried it out and it worked fine. The execution went right this time, as
you can see at bottom this mail.

 A last question: would you apply this fix to
asterisk-async-AGI-rev92324.patch for asterisk 1.6? do you think it would be
useful too? I think so.

Anyway, thanks a lot for your help. Tell me if you need to test anymore and,
as I said before, don't hesitate contact me for whatever help can need.

Best regards.
Jose


[Apr 17 09:40:39] DEBUG[1363]: manager.c:2108 process_message: Manager
received command 'AGI'
[Apr 17 09:40:39]     -- Playing 'es/demo-congrats' (escape_digits=1)
(sample_offset 0)
[Apr 17 09:40:39] DEBUG[1402]: rtp.c:2753 ast_rtp_write: Ooh, format changed
from unknown to alaw
[Apr 17 09:40:39] DEBUG[1402]: rtp.c:2770 ast_rtp_write: Created smoother:
format: 8 ms: 20 len: 160
[Apr 17 09:40:39] DEBUG[1402]: channel.c:1793 ast_settimeout: Scheduling
timer at 160 sample intervals
[Apr 17 09:40:45] DEBUG[1363]: manager.c:2108 process_message: Manager
received command 'Redirect'
[Apr 17 09:40:45] DEBUG[1363]: channel.c:1378 ast_softhangup_nolock:
Soft-Hanging up channel 'SIP/502-08288f98'
[Apr 17 09:40:45] DEBUG[1402]: channel.c:1793 ast_settimeout: Scheduling
timer at 0 sample intervals
[Apr 17 09:40:45] DEBUG[1402]: res_agi.c:439 launch_asyncagi:
ast_check_hangup returned true after handling command on chan
SIP/502-08288f98 (softhangup = 0x2)
[Apr 17 09:40:45] DEBUG[1402]: res_agi.c:497 launch_asyncagi:
launch_asyncagi returned (0x4) for chan SIP/502-08288f98
[Apr 17 09:40:45] DEBUG[1402]: pbx.c:2427 __ast_pbx_run: Spawn extension
(sip_sercom,801,0) exited non-zero on 'SIP/502-08288f98'
[Apr 17 09:40:45]   == Spawn extension (sip_sercom, 801, 0) exited non-zero
on 'SIP/502-08288f98'
[Apr 17 09:40:45] DEBUG[1402]: pbx.c:1831 pbx_extension_helper: Launching
'NoOp'
[Apr 17 09:40:45]     -- Executing [801 at sip_sercom:1]
NoOp("SIP/502-08288f98", "entrada numeracion del 8 801") in
new stack
[Apr 17 09:40:45] DEBUG[1402]: pbx.c:1831 pbx_extension_helper: Launching
'AGI'
[Apr 17 09:40:45]     -- Executing [801 at sip_sercom:2]
AGI("SIP/502-08288f98", "agi:async") in new stack



2009/4/16 Moises Silva moises.silva at gmail.com
> This can be tricky, I did not spend much time looking at different
> return codes. My reasoning was that returnstatus |> agi_handle_command()
should preserve the command return status and
> return it to the caller (just as I saw was done for regular AGI), but
> also preserve the AGI status. However, taking a second look at the
> code and definitions of AST_PBX_KEEPALIVE, I think my fix is wrong,
> probably I just broke the "break asyncagi" AGI command and it
seems
> the AST_PBX_KEEPALIVE code is not even used by the caller, just
> ignored. Can you help me to test something else?
>
> http://moythreads.com/testasync2.diff
>
> Let me know if it works,
>
> Moy
>
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