Questions tagged [asterisk]

Asterisk is a communications server software and an open source framework for building communications applications.

Filter by
Sorted by
Tagged with
0
votes
1answer
15 views

Asterisk: how to get current sip.conf settings?

I'm not sure if sip general parameter (such as rtptimeout) is set. There is no mention of it in /etc/asterisk/sip.conf. But may be it is set somewhere else (in include files, for example). How do I ...
0
votes
1answer
25 views

Asterisk: how to end frozen calls

Sometimes my voip provider shuts down service without ending calls which are in progress (these calls are incoming to my server and they are coming to queue). So calls are never gracefully finished, I ...
0
votes
1answer
35 views

Asterisk fails to accept incoming calls

I have configured a local Asterisk server. There are local users and a single trunk for external calls. Users can successfully make local and outgoing calls. However when external calls come to the ...
0
votes
1answer
39 views

Outbound audio issues with Asterisk

I am running Asterisk 16.9.0 on a CentOS server at a client's location. That server connects via a SIP trunk provided by the telco. I configured Zoiper on a couple of PC's there and connecting using ...
0
votes
1answer
56 views

Incoming calls are rejected because there is no 's' extension

I've have configured VOIP provider and voip device. Sip config as follows sip.conf [operator] canreinvite=no context=from-operator host=operator_ip secret=operator_secret type=peer qualify=yes port=...
0
votes
0answers
28 views

Asterisk PBX hook-like scripting

I am very confused about scripting on the asterisk PBX by reading the manual. I'm trying to execute a python script at the moment my PBX is receiving a call, and pass parameters to the script taken ...
0
votes
1answer
204 views

Execute ssh via plink on Windows Host with sudo commands

I've Windows and I need access to Asterik (Linux server), via SSH, to execute one command. Now I open Putty, enter IP Server, Port, user and password SSH, execute the sudo command, enter password ...
0
votes
1answer
69 views

Asterisk: ODBC options not available in menuselect

I have done yum install unixODBC and yum install freetds, but when I try make menuselect while installing Asterisk, none of the ODBC options are available. func_odbc and res_odbc show up with an XXX ...
0
votes
1answer
26 views

How to establish connection between android phones using zoiper running Asterisk?

I have this scenario: 2 smartphones and 1 laptop, all in the same network. Both smartphones have Zoiper in order to comunicate with an asterisk server. The laptop is running Mac OS (Host) and inside ...
1
vote
1answer
173 views

Connect a music player in SIP voice call

I need to establish a normal Voip conversation between 2 persons and one of them, using a softphone in Windows, insert audio from a music player which can be controlled by the own user. The voip ...
0
votes
1answer
32 views

How to set the MixMonitor extension that match the RTP codec used?

I have a dialplan where I am using the application MixMonitor. This application take a file path, and the extension is used to define the format of the file (wav, alaw, ulaw, g729, ...). For the ...
0
votes
1answer
369 views

How to use Jitsi as a softphone SIP client?

Wikipedia lists Jitsi features as including: IPv6 support for SIP and XMPP But where and how are SIP credentials configured? So that I can troubleshoot an Alice and Bob SIP call: sip diagram Above is ...
1
vote
0answers
99 views

Asterisk outbound_registration real time issue

I configured an asterisk to run with realtime database. Every thing is working as expected, except the outbound_registration part, which was not running unless using pjsip.conf static file! From ...
0
votes
0answers
40 views

Is it possible to use signal level as signalisation for AEX2400 board in Asterisk?

I want to be able to use signal level on an analog line to be able to generate valid call via a AEX2400 with FXO modules. To understand better my question, lets say I have hooked up a smartphone to ...
-1
votes
1answer
106 views

How to receive phone calls using a data only card without using third party online service?

I’m needing a 5G nsa card for my laptop no larger than 42mm×30mm. The only one I found is the MV31‑W from Gemalto (the manufacturer). But there are maybe other options. After a call to the ...
0
votes
3answers
61 views

Router won't forward ranges of ports to Linux server. Is there a software solution?

I need to forward a large range of ports to a Linux machine running Asterisk. The machine is behind a household router. The router can be successful configured to forward single ports to Asterisk, e.g....
0
votes
1answer
112 views

Where Is Asterisk FreePBX Video Call Recording File

I would like to know where can I find (if it does exist) the video file for a call with video. I already have all the things running properly: Call recording enabled for extensions as Force Call ...
1
vote
0answers
24 views

Setting up a computer as a phone station via my router's telephone ports [closed]

First off, I should note that this is my first time working with telephone hard-/software beyond simply using a phone, plugging it in or setting up a router. In addition, I'm aware I could just use ...
0
votes
1answer
160 views

How to make asterisk call file as user account or extension

I am trying to figure out why the callerid works when I call with twinkle through asterisk but not when making a call file. The callerid with the call file is always unknown. Twinkle account is setup ...
0
votes
1answer
67 views

Asterisk 13 & Pocketsphinx Segmentation Fault

I've built a voice recognition system for german digits using asterisk 13, pocketsphinx and a plugin and server (based on astsphinx) to connect asterisk with pocketsphinx. When testing the system with ...
0
votes
1answer
140 views

Asterisk configuration, need to use a different internip for each trunk

I'm currently configuring my asterisk server, that has two trunks, one for the incoming calls, one for the outgoing calls. For each trunk, I would need to define a specific internip. So far, the IP ...
0
votes
1answer
346 views

Asterisk: check original Caller ID on transfer

I am using Asterisk 16.5.0 with usecallmanager patch (which allow me to use Cisco phones features). We have a number of ConfBridge rooms, which are supposed to be called by employees only. Some ...
0
votes
1answer
47 views

How to omit a bad PSTN line in Asterisk

I have an Asterisk/Dahdi server that serves as a PSTN software PBX running on Debian with a Digium 1TE134F card. The setup has been up and running for sometime but we noticed that one of our lines is ...
0
votes
1answer
291 views

Asterisk, IAXModem & HylaFax: No carrier detected

I try to get a HylaFax + IAXModem + Asterisk Setup on a Debian 10 server running. The server is directly connected to the Internet, the firewall is disabled (no rules set up). The only thing that ...
0
votes
1answer
840 views

How to configure instant messaging in freepbx 14 & Asterisk 13

i have a trouble to configure instant messaging using freepbx 14 and asterisk 13 , i want that two sip clients can send and receive messages using their soft phones on smartphone and desktop , can ...
0
votes
1answer
129 views

Sip Trunk Peer issue with 2 asterisk servers

I am getting the unauthorized error while trying to place a call: [Sep 3 16:20:19] Asterisk 13.27.0-vici, Copyright (C) 1999 - 2014, Digium, Inc. and others. [Sep 3 16:20:19] Created by Mark ...
0
votes
1answer
58 views

Random hop in IVR

I have an Asterisk 13.23.1 server with freePBX 10.13.66-1. My problem is this, in one IVR when the caller puts a no valid destination (enter a number not configured in the IVR plan) it should go to ...
1
vote
0answers
2k views

Asterisk: chan_sip.c:4321 __sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error for pkt data

I have a setup of Asterisk PBX running on a Raspberry Pi 1 (IP 192.168.2.2) and it runs a few local IP phones just fine. I recently purchased a Grandstream HT813 gateway (IP 192.168.2.3) to connect ...
0
votes
1answer
401 views

DAHDI 3.0.0 not assign spans and cannot generate configuration

I'm setting up a new fresh installation of Debian 10, Asterisk 16 and DAHDI 3.0.0. I have specific the Digium TDM410P 4 port analog card (PSTN) and trying to configure it with asterisk and dahdi. ...
2
votes
2answers
830 views

Asterisk 16 LTS & PJSIP; hello world works but no sound coming from endpoints

I have recently set up an Asterisk server with version 16.4.0 LTS. I am completely new to asterisk yet I have managed to set up the server with the service and it runs smoothly among LAN users and it ...
0
votes
1answer
92 views

Supervisor don't see playback recording in freePBX UCP dashboard

My IT Head surely change the default location where call recordings are being saved, as a result Team Leads can no longer view playback button in UCP, they cannot do call listening. I believe that ...
0
votes
1answer
2k views

Asterisk 16.4 pjsip trunk registration

I'm not an asterisk expert, and I'm stuck at this moment. I'm trying to setup an asterisk box with realtime. Most work, and my endpoints are able to make calls between each other. But I need to setup ...
1
vote
0answers
151 views

Asterisk jitter buffer on receive side of B-Leg (device side)

I've been trying to apply a jitter buffer to the B-Leg or device side of calls for some time now without any success. I've set the jitter buffer to something very large just to test, using an echo ...
0
votes
1answer
246 views

Asterisk how to call by names instead of numbers

I haven't found anywhere clear sample how can I setup alphanumeric alias for internal numbers. For example I am calling now to 504@ip. How can I make it possible to reach 504 number by john@ip by ...
0
votes
2answers
1k views

Asterisk RTP re-invite not working

I’m experiencing the following scenario in Asterisk, and was wondering if you could help: Asterisk source IP sends call to destination signalling IP Destination signalling IP answers call, specifying ...
3
votes
2answers
104 views

Add "+" to Cisco SPA122 (latest firmware)

I use Twilio, and I'm required to use the full ISO spec "+" before all phone calls. MY PBX is Asterisk 13.26.0, connected through private network. I have a Cisco SPA122, with the latest firmware. ...
0
votes
1answer
635 views

Hot to force Asterisk to write date in UTF-8 charset?

CentOS Linux release 7.6.1810 (Core) Kernel: localhost.localdomain 5.0.7-1.el7.elrepo.x86_64 MariaDB: Ver 15.1 Distrib 10.4.4-MariaDB, for Linux (x86_64) using readline 5.1 Asterisk: 16.3.0 built by ...
0
votes
1answer
31 views

Cron job schedule to strem via RTP audio file

I have Asterisk 1.8.28 based PBX and I want to Intercom or Page some audio files at different times daily, I have Cisco SPA508G working as RTP receiver connected to my amplifier and bunch of speakers, ...
0
votes
1answer
2k views

can't run asterisk, no /etc/asterisk/asterisk.conf

i'm trying to install asterisk on my server, I follow to this sample But, in the end of installation, when I call sudo asterisk -vvvr it shows me Unable to connect to remote asterisk (does /var/...
0
votes
1answer
19 views

Is it possible to change files with recordings (answers) in freePBX, Asterisk during talk?

In IVR (freePBX) there are answers recorded that are played when someone is calling , but is it possible to change those answers via REST API or switch recordings "on fly"? I am looking for a way to ...
2
votes
2answers
712 views

Is it possible to decrease the transmit/receive volume in Asterisk below normal?

I am playing around with changing the volume in Asterisk. There are a few applications where I would benefit from this. Most of them are to increase the volume but there are also some scenarios in ...
1
vote
1answer
3k views

asterisk: error while loading shared libraries: libasteriskssl.so.1

I am using Ubuntu 14.04 and followed the instructions to install asterisk from this link Ubuntu 14.04 Asterisk server installation All worked well until I run asterisk and get a shared lib error. ...
0
votes
1answer
322 views

Asterisk sip.conf in MySQL database

I'm able to include a phone inside a database for realtime usage. So, this code (from /etc/asterisk/sip.conf): [phone] type=friend username=phone secret=12345 host=dynamic disallow=all allow=g729 ...
3
votes
1answer
2k views

How to setup Asterisk NAT on docker for Windows

I need to setup a docker container running an Asterisk PBX. The problem is that I receive the wrong ip in sdp replies. In fact Asterisk is using the container internal IP. The second problem is that ...
1
vote
0answers
497 views

Asterisk not adding CDR record in DB for call to unregistered peer

I configured Asterisk CDR backend to PostgreSQL, and everything is working fine for calls between two registered peers. The problem is when a user calls an unregistered peer. I want Asterisk to ...
1
vote
0answers
314 views

On a PBX (Asterisk) determine which phones are connected

I am using Asterisk as my PBX. On my sip.conf I used to have this option enabled: [general] ... qualify=yes more info about qualify can be found in HERE. Thanks to that option I was able to tell if ...
0
votes
2answers
655 views

Converting a 2 channel stereo g729 file to individual single channel pcm encoded file

Whenever I try to convert a G729 encoded two channel file into mono channel pcm encoded wav files I get the following error Input #0, wav, from 'g729stereo.wav': Duration: 00:00:07.07, bitrate: 16 ...
0
votes
2answers
3k views

Asterisk NOT FOUND answer on OPTION request

Remote server send me OPTION package, but my asterisk server send "404 NOT FOUND" response. Is there an option in sip.conf or something else? sip.conf [remote-server] type=friend host=10....
2
votes
1answer
3k views

Asterisk Trying to play back audio file "File /path/to/file.wav does not exist in any format"

i'm trying to rig an application to send out premade calls. I'm trying to get asterisk to play back wav/mp3/gsm files (I compiled it with the proper extensions and they're loaded) but when ever I ...
3
votes
0answers
154 views

Setting CallerID on outgoing ISDN calls? [closed]

Does anybody have any experience of settings the CallerID on outgoing ISDN calls? We are trying to set the caller id to one of the available DDI's on our circuit but without success. They worked on ...