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.
I think the bug is in
On Tue, May 6, 2014 at 5:45 AM, Richard Kenner ken...@gnat.com wrote:
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
That is definitely a leak and the fix looks good.
Thanks.
That leak is most likely the one biting you.
It definitely is.
There is another leak in handle_cli_confbridge_kick() if the
participant to kick is not in the conference.
Confirmed. I missed that one in my code reading. I just
On Thu, May 1, 2014 at 11:08 AM, Administrator TOOTAI ad...@tootai.net wrote:
snip
As explained in one on my previous message, it's a bug, easily reproducible:
take a queues.conf (or sip.conf or iax.conf or voicemail.conf or ...) like
this (what is important is the #include):
snip
On Thu, May 1, 2014 at 11:08 AM, Administrator TOOTAI ad...@tootai.net
wrote:
snip
As explained in one on my previous message, it's a bug, easily
reproducible: take a queues.conf (or sip.conf or iax.conf or
voicemail.conf or ...) like this (what is important is the #include):
snip
Hi!
my asterisk-12.2.0 with pjsip-2.2.0 does not translate codecs any more.
I tried every combination. silent on both sides.
I compiled pjsip with no resample in pjsip.
./configure --prefix=/usr --enable-shared --disable-sound*--disable-resample*
--disable-video --disable-opencore-amr
is
PS.
if I configure both extension 7000 and 7001 to,
disallow=all
allow=alaw
or
disallow=all
allow=g722
everything is fine. as long as the allowed codec is equal in both
extensions.
Am 07.05.2014 07:00, schrieb Rainer Piper:
Hi!
my asterisk-12.2.0 with pjsip-2.2.0 does not translate
that's funny
I recompiled asterisk without bridge_native_rtp.so
to force asterisk to go to simple_bridge and not to native_bridge...
!!! AND THE CODEC TRANSLATION IN ASTERISK IS WORKING AGAIN !!! juhu
Am 07.05.2014 07:11, schrieb Rainer Piper:
PS.
if I configure both extension 7000 and
perhaps a silly question ...
if a channel switches from simple_bridge to native_bridge ... is the
channel switching to direct_media between the endpoints ?
if so, why doesn't turn direct_media = no and
disable_direct_media_on_nat = yes switching to native_bridge off ?
my pjsip.conf
