That's my plan, had hoped you could help, but thanks for your reactivity
and your answers !
Best regards
Le ven. 1 oct. 2021 à 15:59, Joshua C. Colp a écrit :
> On Fri, Oct 1, 2021 at 10:54 AM Killian Matter
> wrote:
>
>> It an app with 2 dialplan function, one to setup
count isn't at 0 ?
Le ven. 1 oct. 2021 à 15:48, Joshua C. Colp a écrit :
> On Fri, Oct 1, 2021 at 10:42 AM Killian Matter
> wrote:
>
>> I don't manage it so it's through the dialplan ... Even so it's useless
>> to try to decrease it through an API call ?
I don't manage it so it's through the dialplan ... Even so it's useless to
try to decrease it through an API call ?
Le ven. 1 oct. 2021 à 15:37, Joshua C. Colp a écrit :
> On Fri, Oct 1, 2021 at 10:35 AM Killian Matter
> wrote:
>
>> Hello !
>> I wanted t
Hello !
I wanted to ask, is there a way to put the use count of my custom module to
0 so that it's possible to unload without forcing ?
Becaus once I've made a call, impossible to unload cause use count isn't 0.
Thanks,
K.m
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That's it, it should be fine with this thanks !
I feel dumb not finding it sooner
Le ven. 26 mars 2021 à 12:11, Joshua C. Colp a écrit :
> On Fri, Mar 26, 2021 at 8:09 AM Killian Matter
> wrote:
>
>> Hello, I'm digging my head to find a way so that in a call as the
Hello, I'm digging my head to find a way so that in a call as the
user/caller speak, he hear himself back, always having his echo of what he
say. I thought I record the call then playback the record but I need the
echo in live as he speaks.
Thanks !
k.m
--
Hello,
I was wondering if it was possible to call an URL in VOIP with asterisk, is
it?
Thanks !
k.m
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Either way i try to catch the event when the bridge is done or like you
said get the bridge and iterate the channels on it
Killian MATTER
Le mer. 3 févr. 2021 à 11:43, Joshua C. Colp a écrit :
> On Wed, Feb 3, 2021 at 6:40 AM Killian Matter
> wrote:
>
>> Hello !
>&g
Hello !
forgive me, i'm struggling to get the outbound channel or the channels of a
call.
I already have the inbound channel, is it possible to go from the inbound
channel and get either way the other channel or the whole lot of info about
the two channels of the call ?
Killian M
Sorry, I had School last, back to work this week, so i processed the core
file. Looked into it, but there so much info, should something about the
seg fault catch my eyes in those files ?
Killian MATTER
Le dim. 24 janv. 2021 à 09:27, Dennis Buteyn a
écrit :
> On 1/24/21 10:19 AM, LSV wr
Well hello, when the call is up, working, then asterisk put up a seg fault
error. Just Segmentation Fault.
What info would you want/do you need to help ?
Killian MATTER
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e on hang up it doesn't have much time to detach audiohook and the
dialplan goes on so some data are freed before the detach of audiohook and
datastore do their job and so seg fault.
Killian MATTER
Le lun. 18 janv. 2021 à 12:08, Joshua C. Colp a écrit :
> On Mon, Jan 18, 2021 at 7:00 AM
it look like a sync problem but is it even possible ? (e didn't ever see a
sync prob in C language)
Killian MATTER
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don't try to get access to something i shouldn't so i don't know at this
point.
Killian MATTER
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Okay, thanks, so I'll have to get the other channel and apply an audiohook
to it
Killian Matter
Le jeu. 14 janv. 2021 à 14:42, Joshua C. Colp a écrit :
> On Thu, Jan 14, 2021 at 9:40 AM Killian Matter
> wrote:
>
>> Is it possible to have a 2 way Audiohook, that ta
27;ll take it.
Thanks,
Killian Matter
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Okay, for now it's works ! Thanks for the help !
Killian Matter
Le jeu. 14 janv. 2021 à 11:55, Joshua C. Colp a écrit :
> On Thu, Jan 14, 2021 at 6:53 AM Killian Matter
> wrote:
>
>> Is it the same for datastore ?
>>
>
> Yes, datastores are also automatically d
Is it the same for datastore ?
Le jeu. 14 janv. 2021 à 11:51, Joshua C. Colp a écrit :
> On Thu, Jan 14, 2021 at 6:48 AM Killian Matter
> wrote:
>
>> i'm trying to detach the audiohook after an h extension that's the
>> origin of the problem, the flow of RTP
i'm trying to detach the audiohook after an h extension that's the
origin of the problem, the flow of RTP just stop before. Have to find
something else than the h extension, might try events.
Le jeu. 14 janv. 2021 à 11:43, Joshua C. Colp a écrit :
> On Thu, Jan 14, 2021 at 6:
by media flow you meant to allow media stream no ?
I forgot to say i use SIP.
Le jeu. 14 janv. 2021 à 11:33, Joshua C. Colp a écrit :
> On Thu, Jan 14, 2021 at 6:31 AM Killian Matter
> wrote:
>
>> No, no media flowing
>>
>
> Audiohooks predate timers, and require
No, no media flowing
Le jeu. 14 janv. 2021 à 11:15, Joshua C. Colp a écrit :
> On Thu, Jan 14, 2021 at 6:10 AM Killian Matter
> wrote:
>
>> It seems that nothing else change it's status, i ran the loop for more
>> than 15 minutes and still, no change of status.
>
It seems that nothing else change it's status, i ran the loop for more than
15 minutes and still, no change of status.
Killian Matter
Le jeu. 14 janv. 2021 à 10:02, Dennis Buteyn a
écrit :
> That loop is equivalent to:
>
> audiohook->status = AST_AUDIOHOOK_STATUS_SHUTDOWN;
&
Hello ,
I'm developing a module on asterisk, while debugging i've come across a
problem I don't quite understand.
I'm using a noise filter, at the end of the call I stop my filter , so
clean up everything, detach the audiohook and there is the problem. It's
stuck in the while loop in *ast_audiohoo
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