Server Fault Stack Exchange
Server Fault Stack Exchange

Questions tagged [asterisk]

Asterisk is software that enables a server to act as an IP PBX system, VoIP gateway, conference server, and more.

see

http://www.asterisk.org/

and

http://en.wikipedia.org/wiki/Asterisk_(PBX)

586 questions
0
votes
0 answers

Status puplication on asterisk

Is it posible to make asterisk publish user agent statuses like online, dnd, etc..? Then i change status on Linphone client i get folowing warning: [Oct 22 12:30:47] WARNING[46]: res_pjsip_pubsub.c:3353 pubsub_on_rx_publish_request: No registered…
eri
  • 274
  • 2
  • 4
  • 15
0
votes
2 answers

Forwarding SIP headers with asterisk (PJSIP)

I'm trying to forward a specific incoming header to the other leg of the call, but can't figure out how to pass the value of the header in the incoming leg to the pre-dial handler [addheaders] exten => addheader,1,Verbose("Setting header") exten =>…
Pownyan
  • 101
  • 2
0
votes
1 answer

command that cannot work in background (asterisk)

am trying to run script in background which contain command such like "asterisk -rvd >> xyz.log", when I run it directly it work well but when run it with any background way (service, cron, &, nohub,,) it stop directly with this message: stopped so…
Mhd
  • 1
0
votes
0 answers

asterisk not recognizing answer from sip trunk

I have an Asterisk server (15.5, FreePBX) with three SIP trunks from different providers configured, two of them are working fine while the third for every call keep sendind the invite despite the correct answer from the trunk. The trunk was working…
Spuria
  • 11
  • 3
0
votes
0 answers

Asterisk automatic hang up when I try to call

I just set up a server with Asterisk set up but when I try to make a VoIP call the line automatically drops... I use CommPeak as a SIP provider and Bria for the interface Here's my…
dolor3sh4ze
  • 101
  • 3
0
votes
1 answer

Asterisk/FreePBX queues no longer working

Incoming calls go straight through to failover regardless what other settings are. I have deleted the queue and rebuilt it, nothing seams to fix it... Asterisk 16 Freepbx 15 Debian 10 There is a warning message "Unable to join queue" even though it…
Ben
  • 13
  • 4
0
votes
1 answer

adding modules to installed asterisk - alsa.so not found

I have set load= chan_alsa.so and i got this error ERROR[77064] loader.c: Error loading module 'alsa.so': /usr/lib/asterisk/modules/alsa.so: cannot open shared object file: No such file or directory is there a missing module or alsa is global ?
mordechai
  • 101
  • 2
0
votes
0 answers

Asterisk - Position Announcement to First Caller

When Asterisk announces the position of a caller in the queue, it plays queue-thereare, followed by the caller's position, followed by queue-callswaiting. We have custom messages for each of these that work perfectly around the caller's position…
Nick Coons
  • 345
  • 4
  • 16
0
votes
0 answers

Audio issue in public network with Asterisk 17

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…
Econ_newbie
  • 101
0
votes
1 answer

asterisk -vvvvvrx 'core show channels verbose'

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?
Rahatur
  • 145
  • 7
0
votes
1 answer

how to run multiple instances of asterisk on one server

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…
d3m0n
  • 87
  • 1
  • 3
0
votes
1 answer

FreePBX / Asterisk: use inbound routes to block spammers/hackers

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]:…
JimB
  • 177
  • 1
  • 1
  • 10
0
votes
1 answer

Asterisk: Using Queue() in the h extension

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…
Mattisdada
  • 57
  • 2
  • 15
0
votes
1 answer

Asterisk WebRTC outgoing call delay

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…
Hativ
  • 111
  • 6
-1
votes
3 answers

Asterisk compilation issue

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…
harshu9713
  • 25
  • 2
  • 6
1 2 3
…
39 40