Am 14.08.2014 17:22, schrieb Mitch Claborn:
Is it possible (and advisable) to copy menuselect options from
Asterisk 11 to Asterisk 12? If so, is menuselect.makeopts the only
file to copy?
I am not sure - but I would'nt do that. Make a hardcopy from your
console and transcribe the settings
I compile everything and then disable the unwanted modules in
modules.conf like:
modules.conf:
;
; Asterisk configuration file
;
; Module Loader configuration file
;
[modules]
autoload=yes
preload = res_odbc.so
preload = res_config_odbc.so
;noload = res_odbc.so
;noload = res_config_odbc.so
On Thu, Aug 14, 2014 at 6:42 PM, CDR vene...@gmail.com wrote:
snip
The machine has 30 asterisk process, most of them dormant.
There is no way with 164 active calls we may have 10484 handles allocated.
I have no idea how to debug this. I suggest that an experienced
engineer from Digium logs
Hello,
After having thought this through a bit I have some thoughts I'd like to
share.
In this case where the rtp profile is RTP/AVP Asterisk accepts and handles
the call normally. If a webrtc client calls a sip client, or even another
webrtc client, rtpengine is needed to step in (in my setup
On Fri, Aug 15, 2014 at 10:41 AM, Olli Heiskanen
ohjelmistoarkkite...@gmail.com wrote:
Hello,
After having thought this through a bit I have some thoughts I'd like to
share.
In this case where the rtp profile is RTP/AVP Asterisk accepts and handles
the call normally. If a webrtc client
Thanks Paul, I appreciate your thoughts.
I understand your way, it's logical in your environment. I prefer to use
LTS versions of Asterisk so I'm guessing what I want to do is not quite
possible with Asterisk 11.
I'd prefer my setup to work like this in different cases.
webrtc (rtp/savpf) --
On Fri, Aug 15, 2014 at 12:17 PM, Olli Heiskanen
ohjelmistoarkkite...@gmail.com wrote:
Thanks Paul, I appreciate your thoughts.
I understand your way, it's logical in your environment. I prefer to use LTS
versions of Asterisk so I'm guessing what I want to do is not quite possible
with
