On 2018-10-12 12:22, Dmitry Melekhov wrote:
>> AMI:
>> - The Command action now sends the output from the CLI command as a
>> series
>> of Output headers for each line instead of as a block of text with
>> the
>> --END COMMAND-- delimiter to match the output from other actions.
>>
>>
Hi,
I have a problem with TLS certificate change and, especially, automated
renewal. How to make Asterisk reload the certificate with minimal
service disruption?
Asterisk (PJSIP) doesn't seem to detect certificate changes on its own.
It won't normally reload transport settings either,
On 2016-12-12 02:21, David Cunningham wrote:
Is there any equivalent of the CONNECTEDLINE function which can be
called from an application using the AMI?
You can use dialplan functions from AMI using GetVar, so this should work:
Action: GetVar
Variable: CONNECTEDLINE(num)
Jacek
--
On 2016-09-09 11:34, Olivier wrote:
Adding an /etc/sysctl.d/foobar.conf file with the bellow content allowed
me to at last produce core dump files (in /var/tmp directory), even if
asterisk is run by asterisk user (and by root).
I choosed this /var/tmp directory to make sure core dumps are not
On 2016-08-10 11:53, Joshua Colp wrote:
Jacek Konieczny wrote:
On 2016-08-09 10:06, Faheem Muhammad wrote:
trip time and Call Setup time of SIP Requests.
In case of GSM Network with high delay you need to set the T1 timer a
higher value like 1000ms (500 ms default). Similarly you can reduce
change T1 for trunks, as they
might not be fast enough to respond and I cannot set it for phones only.
It seems I need to bring back the chan_sip behaviour – 'do not bother
with INVITE to Unreachable devices'.
Jacek
On Tue, Aug 9, 2016 at 12:03 PM, Jacek Konieczny <jaj...@jajcus.net
<mail
On 2016-08-09 10:06, Faheem Muhammad wrote:
Jacek,
This might be a bug or configuration issue, but you need to understand
the SIP Session Timers. With Session Timers you can control the round
trip time and Call Setup time of SIP Requests.
I don't think you really mean SIP Session Timers
Hi,
We have been migrating our PBX system from Asterisk 1.8 and chan_sip to
Asterisk 13 and chan_pjsip. Things are mostly, ok, but now I have
stumbled on a behaviour difference I don't like.
With chan_pjsip when a phone went unexpectedly offline (Ethernet cable
disconnected) Asterisk would
