Questions tagged [voip]

Voice over IP (voice over Internet Protocol, VoIP) is a method and group of technologies for the delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet.

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VoIP without HTTP?

I want to be able to make phone calls to ordinary telephone phone numbers from my (Unix/Linux) laptop without using a web browser (or other proprietary software such as "Skype for Desktop"). ...
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Openwrt and sipproxy why I am unable to perform a call via a softphone?

I have setup in my router openwrt and I try to perform a call using my provider's VOIP settings that I retrieved for its router that provided. My network settings are the following: config interface '...
1 vote
1 answer
692 views

FreePBX No connection to Asterisk

I am new to asterisk and freePBX. I set up a server years ago but it was very basic and used for less than a month. I’m now launching a new company and found myself in need of a VOIP system so I’m ...
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2 votes
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Asterisk: how to change the CLI color prompt?

The Asterisk wiki says that the color of the CLI prompt can be changed with %Cn[;n]. ie: export ASTERISK_PROMPT="%C31[%H]: " but when I use the export above, I get broken prompt [1;31m[voip]...
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Skype for linux won't launch

I am trying to launch the newest skype versions (~)8.57.0.116-r1 (~)8.58.0.93 on my Gentoo and it won't launch, the only error being this line in dmesg traps: skypeforlinux[17278] trap int3 ip:...
2 votes
1 answer
5k views

How do I fix this error: "ipset v6.11: Hash is full, cannot add more elements"

When I run the update script for voipbl manually, I get this error from ipset: ipset v6.11: Hash is full, cannot add more elements. I am running it manually because some IPs that are on the blacklist ...
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VOIP worked with Mint 19.1 & kernel 4.4, fails w/ kernel 4.8

With Linux Mint 19.1, kernel 4.15, I connect to the Internet using WiFi, and want to share the Ethernet connection with a Grandstream IP Phone. I have done: click | network connection | wired ...
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28 views

Change asterisk voicemail settings based on a schedule?

I need to set up an asterisk voicemail inbox that: Works normally during business hours. After business hours, calls to this line go straight to voicemail after a special "we're not in right now" ...
1 vote
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159 views

Debian configure OpenSIPS for Websocket and UDP

I'm trying to configure OpenSIPS with OverSIP so that I could make a SIP call between a webrowser and a sip client app like Yate or Linphone. The call between two client apps (Yate/Linphone) works ...
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2 votes
1 answer
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How do I prevent Viber from turning my microphone volume too high?

I am using Linux Mint 17, and I installed Viber to use as an alternative to Skype. I got the Viber client that is available on their website: https://www.viber.com/products/linux/ Whenever I start a ...
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2 votes
3 answers
2k views

How to make Skype conference calls and screen sharing in Linux

Now that Skype got discontinued for Linux and only the 5 beta version is avalable, Linux users have a problem, which is how to communicate with Mac and Windows users. There have been lately a lot of ...
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Horrible input sound on any distro

I've been toying around with several distros in my endless effort to ditch windows and stick to linux but I haven't been able to work this out. I used to have a desktop where everything just worked ...
1 vote
1 answer
198 views

Changing asterisk call-in directory recording?

I'm not sure what to call this, so I cannot Google it. When you call your Asterisk line and go to the directory, it asks you to enter the first three letters of your party's last name. When you do so, ...
2 votes
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913 views

Asterisk and NAT: SIP and IAX registration failed on remote connection behind NAT

I set up two asterisk servers (on Fedora) in different networks. My goal is to make a call from softphone (on windows lite with ip: 192.168.20.3) to the asterisk server 2 which is in the other network ...
1 vote
0 answers
316 views

Asterisk: SIP and IAX registration failed on remote connection [duplicate]

I set up two asterisk servers (on Fedora) in different networks. My goal is to make a call from softphone (on windows lite with ip: 192.168.20.3) to the asterisk server 2 which is in the other network ...
1 vote
0 answers
184 views

shorewall - VoIP - jitter buffer

I am using shorewall as my firewall and traffic shaping platform; however, I'm wondering what more I can do to help mitigate upstream jitter issues. I have a fair amount of bandwidth, but the ...
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Asterisk: Can call but can't hear voice

I set up a simple asterisk server on Fedora. My goal is to make two android phones call each other. Actually they are connected via wifi, and I use Zoiper and Jitsi softphone. Fedora is on a ...
1 vote
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how autopause clementine on skype call?

With kde-connect is possible: when come a call, clementine autopause; with VLC is possible: on Skype call the music is autopaused. How can get this in clementine? Autopause music on skype call? I see ...
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2 answers
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Broken package problem installing ring on Mint 18.1

I have a Linux Mint 18.1 Serena installed, and the apt-get install ring won't do the trick. it returns the following: Some packages could not be installed. This may mean that you have requested an ...
1 vote
2 answers
888 views

Voice-over-IP with a VPN

I use an IPFire (linux) router as a VPN client. My voice-over-IP is working properly without the VPN active. However, when the VPN is up, my SIP client (Twinkle) can't register with my VoIP provider. ...
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3 votes
0 answers
835 views

how do I make a SIP phone ring with sipsak?

How can I use sipsak to simulate a call so that there's an actual ring, or possibly even a sound is played? thufir@mordor:~$ thufir@mordor:~$ sudo sipsak -v --usrloc-mode --from sip:demo_alice@...
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1 vote
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How can sipsak simulate a call with usrloc?

Building on a stackoverflow question, how can I actually get the device to ring? thufir@mordor:~$ sudo sipsak -vv -s sip:6003@localhost No SRV record: _sip._tcp.localhost No SRV record: _sip._udp....
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1 vote
1 answer
578 views

Linphone (3.9.1) compiling error with ./prepare.py --package in OS X El Capitan (10.11.4)

I had clone the linphone-desktop project and follow all the steps provided both on the README and in the README.macos from the linphone repository to install all the dependencies, by the way I'm using ...
1 vote
1 answer
377 views

ip_local_port_range kernel parameter confusion

On my system we have following setting: [root@voip ~]# cat /proc/sys/net/ipv4/ip_local_port_range 32768 61000 We have asterisk running on it which is using RTP port range 12000-13000 in case we ...
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Asterisk v13 on Kali Linux: No RTP engine was found. Do you have one loaded?

I installed Asterisk and tried to make a call with zoiper but I get an error [call failure 401 forbidden] and Asterisk return this message : Connected to Asterisk UNKNOWN__and_probably_unsupported ...
1 vote
0 answers
789 views

Another problem installing Asterisk on kali linux

I am trying to install Asterisk [version 13] on Kali linux. I first install DAHDI, but I won't install libPRI. So when I install asterisk, I use: ./configure and make menuselect Till now it'...
2 votes
2 answers
11k views

Unable to install dahdi - /usr/bin/install: cannot stat ‘’: No such file or directory

I am trying to install dahdi in my centos 7 system using this link but having some errors [root@localhost ~]# cd /usr/src/dahdi-linux-complete* [root@localhost dahdi-linux-complete-2.11.0+2.11.0]# ...
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How can software that doesn't natively support JACK be made to work with it?

Audacity, for example, can be switched to JACK if needed. What about software that can't but can be switched to alsa?
1 vote
0 answers
219 views

Debian ARM on chroot Mumble server error

I use Debian running in a chroot next Android. $ uname -a Linux localhost 3.0.64-CM-g7e7a10d #5 SMP PREEMPT Wed Apr 8 17:34:47 BRT 2015 armv7l GNU/Linux I installed mumble server apt-get install ...
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Error: you do not appear to have the sources for the 2.6.32-042stab102.9 kernel installed

I've just bought a new VPS with Cent OS 6.6 installed. I'm attempting to install Asterisk 11 on this VPS via command line remotely. I've used the directions here however I get this error: you do not ...
1 vote
1 answer
960 views

SIPp "hello world" message to Asterisk

How do I run diagnostics against Asterisk? Asterisk is running on tleilax; and doge is on the same network (My network topology isn't optimal). Specifically, I want to do something like: sipp 345@...
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2 votes
1 answer
392 views

ekiga as an asterisk client

is this the typical Asterisk topology? On my home system, I have the Asterisk server, tleilax, and the client, doge, connecting into my router -- so that they're both on the same network. The IP ...
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0 votes
1 answer
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how to connect a softphone to asterisk

To give context, here's my network: On the tleilax server, there's clearly a result from sipsak on doge: tleilax*CLI> tleilax*CLI> core show version Asterisk 1.8.29.0-vici built by abuild @ ...
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1 vote
0 answers
317 views

Make a PSTN call from softphone using IP PBX module

I just bought a PBX IP Module running Linux IP PBX 2.6 Unitekk. I succeed to run VoIP call in my LAN network and also getting the PSTN call into my softphone (XLITE). Can I make an external call ...
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2 votes
0 answers
2k views

how to do a sip call from sipp

I am newest user using SIPp , I am trying to made a sip call from SIPp program on Linux but i fail on it . i want a command to call phone number (14042835351) from sip domain (80.81.82.83) with same ...
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5 votes
1 answer
586 views

Easy voice talk between 2 computers in 2 countries?

My plan is to enable 2 sets of unsophisticated users (children and their grandparents) in 2 different countries to talk easily. In practice I see it as: A quiet Linux computer is on all the time in ...
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3 votes
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307 views

VoIP dialer for SIP

I have some Java, so am biased in that direction, but not overly so. None of the Java suggestions: https://stackoverflow.com/questions/16471795/sip-client-for-java https://stackoverflow.com/...
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6 votes
3 answers
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How can I set yate to be my default tel: protocol handler?

On this site I clicked on the support link and a phone number popped up. It was formatted via the tel: protocol, and as such was underlined and highlighted like a web link. On my phone, clicking on ...
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3 votes
1 answer
349 views

Asterisk, GSM-FXO gateway example of usage

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 ...
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3 votes
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How to setup QoS on router (prioritize VOIP and web browsing over downloading)?

I've got Asus WL500GP-v2 router with renewed Oleg's firmware installed (http://code.google.com/p/wl500g/ , the stable version). I would like to setup some Quality of Service to ensure all users are ...
5 votes
2 answers
1k views

Jingle CLI client / Jitsi replacement / VOIP

I am looking for a Jingle client for the command line. Currently, I use Jitsi, but my dreams of not needing the GUI for more than just web-browsing are not realized due to my reliance on this program....
2 votes
1 answer
632 views

(IP Telephony, VoIP and Video Conferencing) Ekiga configuration for LAN with same subnet

I am looking for an Open Source Voice Chat Application, like Skype, but within an isolated intranet (no Internet is available). Is there any application which can run on Scientific Linux or CentOS? ...
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1 vote
1 answer
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How do I get my phone to ring using Linphone and MagicJack?

I have a VOIP number setup with Linphone and IPKall. I am using my old magicJack to connect my handset. The handset works for voice input and output, but when I receive a call, it only rings in the ...
3 votes
1 answer
1k views

Audio file as an audio input device

I want to play an audio file over VoIP in order that people can listen to it with a phone call. One way to do this would be to set up the audio file as an audio device where the audio starts playing ...
8 votes
7 answers
4k views

Secure FOSS alternative to Skype on Linux & OpenBSD?

Criteria: Makes audio/video calls Encrypts the whole traffic (using good encryption) Is cross-platform (including Windows 7, etc.) Runs on modern Linux distributions (Fedora, Ubuntu, etc.) Runs on ...
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