I have just installed and configured Asterisk 17 in a desktop PC running Ubuntu 18.4
My Asterisk and one of the clients using Zoiper Softphone are behind NAT. Another Client is an iPhone running on 4G network. I have configured my router to forward…
The output of the command asterisk -vvvvvrx 'core show channels verbose' shows total number of calls processed. That number is an indication of the calls processed since the service is running or last x hours or last x days?
hope everybody is safe during challenging time
i have a task to automate our demo installation process.
Our software is mainly a GUI that operates with asterisk via AGI and stores all data to MySQL / MariaDB storing all sip configuration , and CDR…
My FreePBX / Asterisk configuration was recently forced into allowing both anonymous inbound calls and SIP guests. So of course we're now getting blasted with spam/hack attempts.
They show up in the log as:
[2020-05-02 11:09:53] WARNING[30801]:…
I'm trying to get users to be able to record a message, hangup and have the call continue and dial Queues and playback the recording.
I've gotten most of the way there, but right now when I call Queue() in the h extension it hangs up immediately…
I run an Asterisk 16 installation and a WebPhone based on SIP.js. Unfortunately, I often don't hear the first few seconds when I call someone. But everything is fine with incoming calls.
The Asterisk is in a data center, the browser / client is…
I've downloaded source code of Asterisk from http://downloads.asterisk.org/pub/telephony/asterisk/
I'm getting error while compiling this from source code in Ubuntu 16.04.1.
Please suggest prerequisites for asterisk which needed for…
Our phones are down with the warning above when I try to run command show channels. Is a core restart was done, how long before phones back up? Or is there something else that needs to be done to get phones back up?
I have FreePBX setup and it has 2 NICs. I've been trying to configure it out so that:
eth0 will continue to be the one used for all the office phones and internal calls; and
eth1 is the new NIC that would be connected to the internet so that I…
I have a centos 6 server running an asterisk, freepbx, and apache 2.2.25. On this server, the apache server serves up a webpage called fop2. I've talked to the developer, and he says the files are all in line, and that the apache server isn't having…
I have a VoIP box setup with Asterisk and using chan_dongle to provide me with an inbound GSM trunk as well as a couple of DID SIP trunks with local numbers. I would like to be able to have the following call-flow:
Person ring the GSM mobile…
I have a queue with some members.
The strategy is leastrecent.
Example:
Members: 101, 102, 103, 104
A call goes to 102, and 102 answer the call, and then, the ext. is in use.
Another call go in... the call goes to 102 too, when other extensions are…
I am setting up a PBX that external users will connect to and I was wondering how important PBX placement was.
Are calls routed through the PBX itself, or do they go directly through the provider that provides the SIP address to the PBX, or is it…
I can't seem to get outbound calling for Asterisk working.
::Useful Info::
trixbox Version: 2.8.0.4Asterisk Version: 1.6
Pastebins: http://pastebin.com/6eH87Nri | http://pastebin.com/MU6F6dDC
The calls are supposed to transit a trunk provided by IP…
We have an aging digital phone system at my office. We would like to replace it with something modern, such as an IP based phone system. I have read a bit here and there, even browsed through some voip beginners books. We have a few quotes with…