I'm using Asterisk 11 and have a problem with when making call transfer on
remote Asterisk.
This dial plan below works when I make a call directly to remote asterisk
dialing FXO on remote asterisk.
exten => 4,1,Dial(${FD_L1},25,trw)
exten => 4,n,GotoIf($["${DIALSTATUS}"="BUSY"]?vmail:line2)
On Tue, May 9, 2017, at 11:06 AM, marek cervenka wrote:
> i can upgrade asterisk to DONT_OPTIMIZE version at night
>
> before that, do you see something strange?
>
> is it known issue?
The only issue that looks like it could be related is ASTERISK-26969[1].
Once you have an unoptimized
i can upgrade asterisk to DONT_OPTIMIZE version at night
before that, do you see something strange?
is it known issue?
[Thread debugging using libthread_db enabled]
Using host libthread_db library "/lib64/libthread_db.so.1".
Core was generated by `/usr/sbin/asterisk -f -C
when run from console without systemd i found its segfaulting
turned core dump on because it was off
Dne 09/05/2017 v 13:52 marek cervenka napsal(a):
hi,
i have strange problem with asterisk 13.15.0+pjsip bundled/centos
7/systemd start script
we are using chan_pjsip only for webrtc
Greetings,
I think this is a better solution:
I've created a simular solution for our main incoming line. Extentions
can add/remove themselfs from
the distrubuting extention.
I used the DATABASE functions of Asterisk to accomplish this following.
my example:
first create a few
hi,
i have strange problem with asterisk 13.15.0+pjsip bundled/centos
7/systemd start script
we are using chan_pjsip only for webrtc endpoints . switched from sipml5
to jssip with upgrade to 13.15.0(from 13.9.0) few days ago
today i have problems with stopping/crashing asterisk
I would use a Queue with RingAll strategy. Then, I would Pause/Unpause
Agents.
A "Paused" agent would not receive calls from the Queue, but can still
receive direct calls.
You can set an extension to Pause the member and another to Unpause it,
using the applications PauseQueueMember and
