that means you've been using matt's modified version.
you can get the latest stable version ( minus matt's new dtmf features, etc. ) from the sourceforge subversion repository:
> svn co https://svn.sourceforge.net/svnroot/iaxclient/trunk/app_conference
give it a whirl and let us know if it works
yeah, if you have the source code around, look for a file called 'VICIDIAL.txt' in the app_conference directory.j-On 7/12/06, Brian Capouch
<[EMAIL PROTECTED]> wrote:jeff oconnell wrote:
>> thanks brian, this is all really helpful feedback!>> just to be clear, which app_conference code were you us
"Brian Capouch" <[EMAIL PROTECTED]> wrote:
> Henry J. Cobb wrote:
>>
>> I tried several different combinations of app_conference and Asterisk
>> versions and then I had to get back to actually providing phone service
>> that didn't crash.
>>
>
> I hate to me-too, but my experience was identical. C
thanks brian, this is all really helpful feedback!just to be clear, which app_conference code were you using?the svn trunk version from sourceforge? or the VD_app_conference matt's been working on?
j-On 7/12/06, Brian Capouch <[EMAIL PROTECTED]> wrote:
I hate to me-too, but my experience was identi
Henry J. Cobb wrote:
I tried several different combinations of app_conference and Asterisk
versions and then I had to get back to actually providing phone service
that didn't crash.
I hate to me-too, but my experience was identical. Crash after crash,
and I tried everything that was suggest
"Matt Florell" <[EMAIL PROTECTED]> wrote:
> My backtraces never actually mention play_sound, but the crashes only
> happen right after app_conference attempts to play out DTMF tines with
> the playing function.
This is because Malloc isn't crashing when the mistake is made.
It crashes later becau
Hello,
My backtraces never actually mention play_sound, but the crashes only
happen right after app_conference attempts to play out DTMF tines with
the playing function.
Here's the backtrace for two of the crashes that we had with app_conference:
http://205.201.151.24/files/app_conference-crash-
interesting. i didn't realize the problem seems to specifically be the sound playback via the manager interface.a couple weeks ago, my asterisk on my dev box crashed, i did some preliminary investigation, but since we hadn't had any problems in production or qa, i chalked it up to me messing up my
It really depends on the application. app_conference does wonderfully
for long conferences without a lot of entry/exit and no playing of
audio files.
The issues with the double-free crashes that we've had all seem to be
caused by playing of audio files(like the entry/exit sounds or the
DTMF broad
[EMAIL PROTECTED] wrote:
> On 7/12/06, Henry J. Cobb <[EMAIL PROTECTED]> wrote:
>
>> When I've tried it, app_conference always crashed within the hour.
>
> that's strange. we've use app_conference for months and months on end
> without incident.
>
> are you building app_conference from the main svn
On 7/12/06, Henry J. Cobb <[EMAIL PROTECTED]> wrote:
When I've tried it, app_conference always crashed within the hour.
that's strange. we've use app_conference for months and months on end without incident.are you building app_conference from the main svn trunk? or are you using matt's VD_app_conf
"jeff oconnell" <[EMAIL PROTECTED]> wrote:
> but while i'm in the code, i'll also take a look and see if i can
> figure out what your memory issues are...
When I've tried it, app_conference always crashed within the hour.
I think that the entire Asterisk server, including app_conference, needs
to
Sounds good, let me know if you want the gdb bt full output from the
core dumps that I have.
The DTMF broadcasting was a workaround to be able to use a non-Zap
channel in a conference(non-Zap channels in a meetme cannot always
send DTMF and it's strange design made it very difficult to alter).
Th
matt,
i was looking at your dtmf changes today. they look pretty interesting.
right now i'm working on a scheme for cleaning up the clicking we hear
when dtmf tones are not fully filtered by front-end asterisk servers.
meetme seems to do this by calling:
ast_channel_setoption( chan, AST_OPTION_
I have written such a modification into app_conference. It allows the
option of rebroadcasting DTMF tones and/or RFC frames to participants
if enabled.
There are also a few other modifications in the version that I am
using. I released it a month ago and have used it on a few servers
since. It wo
henry,
did you have any luck setting this up?
i'm actually working right now to _suppress_ dtmf clicks in app_conference,
and would be happy to look at the dtmf pass-through, if you're still in need.
j-
On 5/29/06, Henry J. Cobb <[EMAIL PROTECTED]> wrote:
We need to conference together a call
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