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
[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 trunk? or
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
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
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:
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 because of
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
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
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. Crash after
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 using?
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
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
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
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,
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).
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 be
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