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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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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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0answers
50 views

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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0answers
14 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" ...
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0answers
63 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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1answer
254 views

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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0answers
44 views

What's wrong with my curl?

I'm trying to download some data from a VoIP phone on the network using the curl command without the certificate being validated, I've tried this command but it's just writing the text "Unauthorized ...
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3answers
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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1answer
1k views

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 ...
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1answer
83 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, ...
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0answers
602 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 ...
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0answers
207 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 ...
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0answers
120 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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1answer
483 views

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 ...
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0answers
26 views

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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2answers
2k views

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 ...
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2answers
541 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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0answers
497 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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0answers
757 views

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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1answer
406 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
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1answer
286 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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1answer
1k views

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 ...
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0answers
572 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'...
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2answers
8k 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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1answer
17 views

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?
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0answers
187 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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1answer
643 views

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 ...
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1answer
669 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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1answer
319 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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1answer
917 views

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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0answers
234 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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0answers
1k 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 ...
5
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1answer
506 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 ...
3
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0answers
249 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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3answers
1k views

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 ...
3
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1answer
298 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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0answers
744 views

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 ...
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2answers
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
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1answer
559 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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1answer
1k views

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
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1answer
903 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 ...
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7answers
3k 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 ...