I am using AMI from Python. I guess the same can be achieved from the Asterisk CLI. I want to be able to get details about every extension in the PBX. More specifically, for every extension I want to know:
If the extension is in a call, what is the…
On my asterisk server, when i do a sip reload, i get the message "Using SIP CoS mark 4", followed by a registration time out.
I need to make that statement say "Using SIP CoS mark 5". How do I change the SIP CoS mark 4 to SIP CoS mark 5?
Can anyone explain where to use GSM-FXO gateway. I understand where to use GSM-FXS - for example to connect analogue telephone to gsm gateway. But do we use GSM-FXO to connect to two telephone stations?
as the nrpe user when
Command: /scripts/nagisk.pl -c version
Outputs: Asterisk 1.8.11-cert10
Using nrpe
Command: /usr/lib/nagios/plugins/check_nrpe -H [HOSTNAME] -c check_asterisk_version
Outputs: Error getting version
I have this line in my…
So I have a dedicated machine running Ubuntu 12.04 server with FreePBX 2.9. Earlier I was attempting to familiarize myself with SQL via sqlite3, but I received this dreaded error:
SQLite header and source version mismatch
So I saw a possible fix…
I have an Asterisk server doing automated calls and I'm noticing an unexplained high load in it. The server is only running Asterisk. Database and other support applications run in a different machine.
What can be causing this high load?
If the load…
i'm new on the world of PBX and network infrastructure.
I worked in a model and wanted to know how feasible it is to implement it in a production environment.
The requirement is to create a solution in the cloud to offer SME customers with our…
I need to be able to collect core dumps, but since Asterisk's CWD is / (so claims procfs), it'll never be able to write them. I've confirmed my suspicions by allowing world write to / and SIGABRT the process, lo and behold, I had a core.
I can…
We have recently set up a new Aterisk pbx server based on our previous pbx (also Asterisk) which was installed by an external company.
We've kept the configuration of the server the same, all we did was add db support to Asterisk, which wasn't…
This issue has been bugging me for the past several days having spent over 40 hours investigating this issue intensively.
Effectively we run asterisk 1.4.42 which I understand is old, however is the last real stable asterisk version which works…
I'm trying this library using the test service: http://tryit.jssip.net/
When I register, using WS, not WSS, peer is registered succesfully
When I make a call i got
"rejecting secure audio stream without encription details"
Do you know a possible…
I'm rather new to Asterisk, and I need my server to support WebRTC. As far as I know, Asterisk version in Asterisk Now is compiled without SRTP support, which is necessary for WebRTC.
So, I try to compile Asterisk 11.5.0 with SRTP on my Ubuntu…
I am running Ubuntu 13.04 with the corresponding versions of monit (5.5-6) and asterisk (1.8.13.1). I have monit set up to watch my asterisk log file for disconnection to my SIP provider and restart asterisk for a new connection. Here is the…
I am trying to install the dahdi module on my server but it keeps failing when i run the make all command. See below:
root@server:~/dahdi-linux-complete-2.6.2+2.6.2# make all
make -C linux all
make[1]: Entering directory…