Oliver,
Not per peer. What you can do is if you have a few spare IP's is set up
OpenSiPS and bounce the calls that way. The call will be Carrier ->
OpenSipS -> Asterisk. Asterisk will see the IP of OpenSipS and you can
decide there what to do with the call.
On Wed, Sep 6, 2017 at 5:39 AM,
Marcelo,
I had to drop back to 11.25.1 yesterday. Had other things all day
yesterday. I have not had a chance to try anything yet with the suggestion
from Josh.
Jerry
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Hello,
I'm quite sure this question has already be asked previously but before
diving into it with a lab setup, I would like to re-ask here the thereafter
question.
I've got a bunch of very old Asterisk boxes (lastest Asterisk version is
1.6.1.X), all belonging to the same network, I would like
hi,
i know about architecture limits of app_queue
https://issues.asterisk.org/jira/browse/ASTERISK-25806
what CPUs are you actually using for asterisk + app_queue ? (my actual
scenario 90simult calls, 50agents, call recording to SSD (mixmonitor),
no transcoding, CDR/CEL via odbc to
Hello Jerry.
Does the Joshua's tips helped you to solve your issues or are you still
facing audios problems?
I am asking you because I need to update some servers but I can't have this
kind of problems.
Thanks.
Regards,
On 5 Sep 2017 2:02 pm, "Joshua Colp" wrote:
> On Tue,