> Really, I think we're pretty positive there's a ref leak (since
> otherwise, the CBAnn channel would be long gone). If you can get a
> ref debug log and the standard Asterisk DEBUG log showing the
> problem, that would help a lot in finding out what is going on.
That can't be done in the 12.2.0
Thanks, it almost what I need.
But I can't find a way to pass channel variables to Originate cmd in
dialplan.
Is it possible at all?
On Wed, Apr 30, 2014 at 3:13 PM, Richard Mudgett wrote:
>
>
>
> On Wed, Apr 30, 2014 at 4:33 PM, Igor Dvorzhak wrote:
>
>> Hi all,
>>
>> I need a command to orig
We've built an alert system at our company so that if our monitoring
software notices anything very bad happening, and we don't react to a text
message after a few minutes, then it will begin to call our telephones
directly. This seems to help a lot with staff who are asleep, or who might
not be ne
On Wed, Apr 30, 2014 at 4:33 PM, Igor Dvorzhak wrote:
> Hi all,
>
> I need a command to originate a new channel from dialplan. I should be
> able to continue execution of the current context after this command.
>
> How to do this?
>
Look at this application:
*CLI> core show application Originate
Hi all,
I need a command to originate a new channel from dialplan. I should be able
to continue execution of the current context after this command.
How to do this?
Best,
Igor
--
_
-- Bandwidth and Colocation Provided by http:/
On Tue, Apr 29, 2014 at 1:31 AM, binary dreamer
wrote:
> i would like to read information from a file (txt)
There are a few applications and functions that may help you out.
In Asterisk 10 or before try the ReadFile application. Otherwise in 11
or beyond I believe you want the FILE function.
ht
On 30-04-14 12:50, [Digital^Dude] ® wrote:
make gives this:
IIRC Digium's policy is that there's no support on this list for
patented technologies like AMR which are possibly not officially
licensed. Obviously to prevent any legal liability.
HTH,
Patrick
--
Thank you for your feedback.
The call that I attached was actually from me. I get a message on my ATT
phone that says "the wireless customer you...message 1 WA01ML". It is
pretty hit or miss.
On Wed, Apr 30, 2014 at 1:45 AM, Thorsten Göllner wrote:
> Hi,
>
> it seems, that the caller hangs
On Wed, Apr 30, 2014 at 8:20 AM, Richard Kenner wrote:
> > If the reference count on the bridge is off, you should see the
> conference
> > bridge 'hanging around' after the last participant has left.
>
> And how would I be sure this is the case? I did "core set debug 1" and
> didn't see the deb
Le 30/04/2014 15:19, Matthew Jordan a écrit :
On Wed, Apr 30, 2014 at 8:13 AM, Administrator TOOTAI
mailto:ad...@tootai.net>> wrote:
Please, people from Digium, Matt again closed the new bug
ASTERISK-23689 I opened (clone from 23683) telling that it's not a
bug. Did he carefully r
> If the reference count on the bridge is off, you should see the conference
> bridge 'hanging around' after the last participant has left.
And how would I be sure this is the case? I did "core set debug 1" and
didn't see the debug line about destroying the conference, but it doesn't
show up in
On Wed, Apr 30, 2014 at 8:13 AM, Administrator TOOTAI wrote:
> Please, people from Digium, Matt again closed the new bug ASTERISK-23689 I
> opened (clone from 23683) telling that it's not a bug. Did he carefully
> read the comments on the new bug? If not, please forward him this email,
> *it's* a
did you try rebooting after installing 11.9?
-Original Message-
From: Administrator TOOTAI
Sender: asterisk-users-bounces@lists.digium.comDate: Wed, 30 Apr 2014 15:13:59
To:
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Reload problems
Please, people from Digium, Matt again closed the new bug ASTERISK-23689
I opened (clone from 23683) telling that it's not a bug. Did he
carefully read the comments on the new bug? If not, please forward him
this email, *it's* a bug or you have to explain me why it is not!
Le 30/04/2014 13:00,
On Tue, Apr 29, 2014 at 6:03 PM, Richard Kenner wrote:
> > If the channel still hangs around after the conference is destroyed
> > then there is a problem.
>
> Am I missing something obvious: I'm looking in the confbridge_exec
> function. I see a "conference = NULL" line, but no attempt to free
Hello,
I'm trying to fetch outbound SIP PROGRESS Reason cause code in the
dialplan,
Asterisk 1.8.26.1 sip show settings:
Q.850 Reason header:Yes
Store SIP_CAUSE:Yes
However, i'm not getting any value in the dialplan variables, any
successful users of this feature?
--
__
Le 30/04/2014 13:04, Derek Andrew a écrit :
Does a reload (not a sip reload) reload everything or does it also
require the sip.conf file to be modified?
reload as well as module reload chan_sip.so does nothing. Only way i had
till I found why was to restart asterisk :-(
On Wed, Apr 30, 20
Is your script really so simple?
Enable agi debugging (agi set debug on) and take look at it when this
happens.
-Thorsten-
Am 30.04.2014 11:47, schrieb Hoggins!:
Hello all,
I have a strange problem with a very simple AGI script, using the GET
DATA command.
When using this command, Asterisk
Does a reload (not a sip reload) reload everything or does it also require
the sip.conf file to be modified?
On Wed, Apr 30, 2014 at 5:00 AM, Administrator TOOTAI wrote:
> Le 30/04/2014 12:39, Administrator TOOTAI a écrit :
>
>> Le 30/04/2014 12:15, Administrator TOOTAI a écrit :
>>
>>> Hi,
>>>
On Wednesday 30 Apr 2014, [Digital^Dude] ® wrote:
> make gives this:
>
> codec_amr.c: In function 'amrtolin_sample':
> codec_amr.c:227: error: 'AST_FORMAT_AMRNB' undeclared (first use in this
> function)
> codec_amr.c:227: error: (Each undeclared identifier is reported only once
> codec_amr.c:227:
Le 30/04/2014 12:39, Administrator TOOTAI a écrit :
Le 30/04/2014 12:15, Administrator TOOTAI a écrit :
Hi,
after upgrade from 11.8.1 to 11.9.0 on our test server, and from
1.8.26.1 to 1.8.27 on production one, some CLI commands like "sip
reload" or "iax2 reload" does nothing.
We opened bug
Hi,
some more information could be usefull.
On Wed, Apr 30, 2014 at 12:15:03PM +0200, Administrator TOOTAI wrote:
> after upgrade from 11.8.1 to 11.9.0 on our test server, and from
> 1.8.26.1 to 1.8.27 on production one, some CLI commands like "sip
> reload" or "iax2 reload" does nothing.
Is Ast
make gives this:
codec_amr.c: In function 'amrtolin_sample':
codec_amr.c:227: error: 'AST_FORMAT_AMRNB' undeclared (first use in this
function)
codec_amr.c:227: error: (Each undeclared identifier is reported only once
codec_amr.c:227: error: for each function it appears in.)
codec_amr.c: In functi
Le 30/04/2014 12:15, Administrator TOOTAI a écrit :
Hi,
after upgrade from 11.8.1 to 11.9.0 on our test server, and from
1.8.26.1 to 1.8.27 on production one, some CLI commands like "sip
reload" or "iax2 reload" does nothing.
We opened bug 23683 but it was immediately closed by Matt Jordan,
Hi,
after upgrade from 11.8.1 to 11.9.0 on our test server, and from
1.8.26.1 to 1.8.27 on production one, some CLI commands like "sip
reload" or "iax2 reload" does nothing.
We opened bug 23683 but it was immediately closed by Matt Jordan,
telling that he can't reproduce it. But we can.
Ex
Hello all,
I have a strange problem with a very simple AGI script, using the GET
DATA command.
When using this command, Asterisk often returns 0 as a result after a
"GET DATA beep 5000" command, without even waiting for input from the
calling party.
It is quite random : sometimes Asterisk behaves
Hi,
it seems, that the caller hangs up immediatly after calling. Try to
reproduce it by yourself. Dial the number (to reach your asterisk
server) and hangup after ~ 0.5 sec (or whatever).
Best regards,
-Thorsten-
Am 30.04.2014 01:11, schrieb Bryce Lowe:
Hello,
I am trying to diagnose an i
27 matches
Mail list logo
