Hi,
we have quite a few Asterisk machines running and try to keep them on a
current version of the Asterisk 11 branch. But since we upgraded to 11.14.0
a couple weeks ago, we have to restart the Asterisk process every week
because the load gets too high and our monitoring complains.
Those
Hi,
On Thu, Feb 5, 2015 at 4:56 PM, Scott Griepentrog sgriepent...@digium.com
wrote:
Can you tell me if the memory usage by Asterisk is also increasing with
load over time?
Yes, the memory usage does rise a bit.
USER PID %CPU %MEMVSZ RSS TTY STAT START TIME COMMAND
11.14
Hi,
I'm registering an Asterisk against my TLS capable service, using
res_pjsip. My config looks like this:
[devtrunk_reg]
type=registration
outbound_auth=devtrunk_auth
server_uri=sip:example.org\;transport=tls
client_uri=sip:1234...@example.org\;transport=tls
Hi,
I have an Asterisk 13.8.2, which is supposed to be only a client to an
encrypted SIP service. All local phones are connected via UDP.
Since I can't use PJSIP (see my mailing list post from yesterday), I
tried configuring chan_sip to work that way. My settings are:
[general]
context=public
Hi,
while implementing an application based on ARI, I wanted to hangup
calls in different states with different hangup reasons. After not
finding anything in the documentation I found the possible reasons in
the
Hi,
On Thu, Dec 6, 2018 at 2:44 PM Joshua C. Colp wrote:
> Nope. No specific reason. The ones there are what most people would use, and
> what those not as familiar with telephony would understand.
Thanks for the quick response. I have submitted a patch for adding
more hangup reasons.
BTW: Is
Hi,
we have an Asterisk server basically passing on calls using the Dial
application. In the pjsip endpoint settings, the dtmf_mode is set to audio.
This works with most calls. However, there is a scenario where DTMF tones
don't get forwarded the way I would expect them to get forwarded.
A:
