Is it possible to make a voice call (VoLTE) using a standard USB-connected (i.e. USB Type-A) 4G / LTE modem / dongle? I am taking about devices such as the [D-Link DWM-222](https://www.dlink.com/uk/en/products/dwm-222-4g-lte-usb-adapter) or the ZTE MF833, for example. As 3G is being switched off in my area, I need this to *really* use VoLTE instead of using "Circuit-switch fallback" (CSFB) to drop back to 3G. I already understand that I will need a sim card capable of "IMS registration". I have found many "similar" question already posted on stackexchange, but they all refer to UMTS (3G) devices. I am aware that boards based on the A7672S can do this, as there is a page and video demonstrating this [here](https://www.engineersgarage.com/a7672-4g-modem-at-command-testing/) . If anyone out there has succeeded in making a VoLTE voice call using a 4G usb-connected modem, preferably from the command line, could you please let me know what model dongle you used? Thank you! As requested in the comments, I am now adding that I would like to be able to make this voice call using any recent Debian Linux (Bookworm, Bullseye or Buster). Also, I do *not* consider this question to be a "request for tutorial". I would simply like to know if anyone out there has succeeded in doing this and if so, what hardware was used. As I am asking a question about ["Using a *nix desktop"](https://unix.stackexchange.com/help/on-topic) , I consider this question to be on-topic.

How can I use SIM slot in HP 8440p Elite Book? I tried every thing on the Internet, but I can't use it. I want to use it to send messages and calls, etc. I mean all SIM functions. I'm using Kali Linux 2.0 which is Debian based. root@Unknown:~# lsusb Bus 002 Device 003: ID 0bb4:0ff9 HTC (High Tech Computer Corp.) Desire / Desire HD / Hero / Thunderbolt (Charge Mode) Bus 002 Device 002: ID 8087:0020 Intel Corp. Integrated Rate Matching Hub Bus 002 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 001 Device 003: ID 090c:037c Silicon Motion, Inc. - Taiwan (formerly Feiya Technology Corp.) 300k Pixel Camera Bus 001 Device 002: ID 8087:0020 Intel Corp. Integrated Rate Matching Hub Bus 001 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub

Ever since installing Zoom Flatpak I've noticed that many web-links (like telephone numbers) that should **not** be associated to ZOOM are now being opened by ZOOM. I'd like to maintain Zoom assocition only for zoommtg and nothing else. Looking inside var/lib/flatpak/exports/share/applications/mimeinfo.cache reveals that zoom is stealing file and URI associations for TEL (telephone) CALLTO as shown below

[MIME Cache]
application/x-zoom=us.zoom.Zoom.desktop;
x-scheme-handler/callto=us.zoom.Zoom.desktop;
x-scheme-handler/msteams=com.microsoft.Teams.desktop;
x-scheme-handler/tel=us.zoom.Zoom.desktop;
x-scheme-handler/zoommtg=us.zoom.Zoom.desktop;
x-scheme-handler/zoomphonecall=us.zoom.Zoom.desktop;
x-scheme-handler/zoomus=us.zoom.Zoom.desktop;

How to stop it from stealing all these other associations?

I noticed that Zoom invitations have a "join by SIP" part, so I duckduckwent SIP and got the impression that I could join a Zoom meeting with Linphone. I created a SIP account, created a Zoom meeting on another computer, and dialed [meeting-id]@[given-ip] with Linphone. But it didn't connect (stopped trying after a couple of seconds). Have I misunderstood something?

I've got a Huawei E173 USB dongle connected to a Raspberry PI. I'm trying to use the device to make and receive calls. For my purpose, it is needed (i) to save audio of the call to file and (ii) to send audio to the telephony call from an input file. As far as I understood (I can't find any official guide), the E173 dongle sets up three new devices (please correct me if I'm wrong): * /dev/ttyUSB0 for commands * /dev/ttyUSB1 for voice data * /dev/ttyUSB2 for notifications In order to enable phone calls, I run the following commands: AT+CLIP=1, AT+CRC=1, AT+CNMI=1,2 on /dev/ttyUSB0. Also, if I run AT^CVOICE? I get ^CVOICE:0,8000,16,20. Finally, when I make/receive calls, I run AT^DDSETEX=2 on /dev/ttyUSB0 to enable audio forward to the /dev/ttyUSB1 port. At this point, I'm stuck. I know there are other similar posts trying to redirect microphone input and speakers output to the dongle, but my aim is to save the conversation to file (during a call, I tried to run cat /dev/ttyUSB1 > filename.raw, but the file can't be opened with standard software, hence, how can I convert it?) and, simultaneously, to send voice from file (which format? I tried to send back the filename.raw to /dev/ttyUSB1 with cat filename.raw > /dev/ttyUSB1, but the entire system gets freezed). How can I save audio to file and send back audio from file? I found some resources on the Internet trying to explain, for similar devices, that audio has to be sent in mono, in digital frequency of 8000 Hz, and digitalized to 16 bit. By looking at the result of the AT^CVOICE? command, such parameters should be applied also in my case. Also, the same resource tells that "audio data should be fed to the modem audio port in batches of 320 bytes every 0.02 seconds". I've tried to apply even such approach, with no luck.

After switching to linux, one thing i miss is something called phonetray, which displayed caller ID to my screen. I wondered if there is a way for Linux to do something similar. I found a blog discussing NCID and call blocking here . Is there a way to use NCID and have the caller ID display for a brief time on screen?

I've downloaded a file encoded in RTTY 45.45 baud (mp3 output of linked video). *I'm not sure if the final output should be audio or text.* I've also installed minimodem which I'm reading can convert data into audio with option --tx and out with --rx. For example converting a picture into audio and back with: cat pic.jpg | minimodem --tx 9600 -f audio.wave minimodem --rx 9600 -f audio.wave > pic2.jpg I tried the following two commands, one expecting audio output and another expecting text: minimodem --rx rtty -f youtube.mp3 > out.mp3 minimodem --rx rtty -f youtube.mp3 > out.txt Neither of these seem to work. Can anyone advise me on how to decode this story?

**How can I redirect the microphone of one computer to listen to it on another computer via ssh? Which is the right device or which is the right command line?** Some years ago it was easy and fun to redirect sound from a remote microphone to a local computer or vice versa – it was an easy telephone. There are [some](http://www.commandlinefu.com/commands/view/350/output-your-microphone-to-a-remote-computers-speaker) [instructions](http://ubuntuforums.org/showthread.php?t=1328338) for it, but none of them seem to work on newer computers/linux distros. I don’t even have a /dev/audio on my computer (Fedora 17). I think that it may have something to do with pulse audio. Or don’t I need pulse audio for this simple telephone? Which is the right device? I can see all my sound devices when I start alsamixer and press the F6 key. But I don’t know which are the devices in my /dev tree.

I want to use my laptop (Debian 8.4) as a bluetooth headset for my smartphone (OnePlus One, Android). This means not just audio of my phone, also incoming calls. I already managed to play normal audio over the laptop speakers, but no telephone data. I saw there is a headset profile for bluetooth. Does it work automatically? I already had worked with PBAP and MAP, but in the bluetooth headset specification i found no helping answers. I set the bluetooth class of my laptop to 0x40040C in /etc/bluetooth/main.conf and also set it via hciconfig hci0 class 0x40040C The class I get from this website . To connect it I use bluetoothctl. But when I change the modus (eg. discoverable on), the laptop always changes its bluetooth class back to laptop. When I am searching for devices at my phone, the laptop appears with a headset icon, but when I pair, the icon changes to a regular headphone icon. In the settings there is also 'Media Audio'. What I think I need is 'Telephone Audio'. So the question is, how can I achieve this? It would be great if I heave no extra (graphical) tools to install and even better when there is some code to get it work in QT. All examples I have found do not work, are for Windows or are too old. In my case receiving a message per bluetooth that a call is incoming would be enough (like a simple smartwatch, just showing). With my current solution with media audio, I hear just the phone ringing, on laptop and phone simultaneously, but just if I set the telephone sound on my phone from silent or vibration to any volume.

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" recording plays. I think it should be possible to set the recording using custom time zones, but I'm not sure that is the best way to do it. I also don't know how to control "rings before voicemail" based on a schedule. Is this possible?

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 such a link opens my default dialer and places the call. When links are not formatted via this protocol, my browser's Google voice plugin usually auto-detects the phone number and gives me a chance to call via Google voice. It would be nice if I could also do this for tel: formatted numbers, or better yet, as I don't always have a voice session open, set tel: formatted numbers be handled by yate. **How can I fix my browser's handling of the tel: protocol links such that it opens in yate or google voice?** - In Chromium I get a warning saying that xdg-open will be used to handle the link, but when I click 'Launch Application' nothing happens. - In Firefox, clicking on the link leads me to a blank page with the url tel:18003744432 - In Konqueror clicking on the link produces this error: Access by untrusted page to tel:18003744432 denied. I'm using Linux Mint 16 KDE x64.

As soon as I start TeamSpeak other applications (e.g. FireFox, VLC Player, ...) gets muted. Additionally, these applications get muted again (after I un-muted them manually) from time to time while TeamSpeak is running. I know that I had the same problem with older versions of Mumble. I remember some Mumble-dev stating that Mumble once was registered as "phone application", thus the PulseAudio or so mutes other applications while "phone applications" are running. Newer versions of Mumble do not have this issue, because they are running as "Games" or something like that. However, like always, the TeamSpeak devs are not helping at all. Hence, I thought maybe I can force PulseAudio to stop this stupid auto-muting. Does somebody know if PulseAudio can be configured to stop auto-muting applications when "phone applications" are running? Or generally turning off any auto-mute functionality? I want to *always* control volume/mute myself.

Is there is an application/service which synchronize contacts from CardDav to LDAP? My plan is to synchronize the contacts which are used in ownCloud to my PBX so I have it on my desktop phone which is linked to the PBX via ISDN.

I want to turn off buffering of SIP calls in freeswitch pbx software. Freeswitch holds RTP data from clients in buffer and sends it every 20ms. I want to freeswitch pass throught packets without holding. How to configure it? ----- EDIT (additional info) ----- I have two SIP clients and FreeSwitch PBX. Voice 8 kHz sample rate, A-Law coding (8 bytes per sample, no compression) When I call directly from one client to another the tcpdump output on one client is: 00:00:00.000475 IP 10.64.177.34.5440 > 10.64.0.42.5082: UDP, length 172 00:00:00.031599 IP 10.64.177.34.5440 > 10.64.0.42.5082: UDP, length 172 00:00:00.032012 IP 10.64.177.34.5440 > 10.64.0.42.5082: UDP, length 172 00:00:00.000315 IP 10.64.177.34.5440 > 10.64.0.42.5082: UDP, length 172 00:00:00.031775 IP 10.64.177.34.5440 > 10.64.0.42.5082: UDP, length 172 00:00:00.000384 IP 10.64.177.34.5440 > 10.64.0.42.5082: UDP, length 172 00:00:00.031499 IP 10.64.177.34.5440 > 10.64.0.42.5082: UDP, length 172 00:00:00.031986 IP 10.64.177.34.5440 > 10.64.0.42.5082: UDP, length 172 00:00:00.000475 IP 10.64.177.34.5440 > 10.64.0.42.5082: UDP, length 172 00:00:00.031578 IP 10.64.177.34.5440 > 10.64.0.42.5082: UDP, length 172 00:00:00.031936 IP 10.64.177.34.5440 > 10.64.0.42.5082: UDP, length 172 00:00:00.000419 IP 10.64.177.34.5440 > 10.64.0.42.5082: UDP, length 172 00:00:00.031573 IP 10.64.177.34.5440 > 10.64.0.42.5082: UDP, length 172 But when I connect from one client to another using pbx as middle point I got: 00:00:00.020013 IP 10.64.72.24.30230 > 10.64.0.42.5078: UDP, length 172 00:00:00.019969 IP 10.64.72.24.30230 > 10.64.0.42.5078: UDP, length 172 00:00:00.020017 IP 10.64.72.24.30230 > 10.64.0.42.5078: UDP, length 172 00:00:00.019984 IP 10.64.72.24.30230 > 10.64.0.42.5078: UDP, length 172 00:00:00.020078 IP 10.64.72.24.30230 > 10.64.0.42.5078: UDP, length 172 00:00:00.020016 IP 10.64.72.24.30230 > 10.64.0.42.5078: UDP, length 172 00:00:00.019850 IP 10.64.72.24.30230 > 10.64.0.42.5078: UDP, length 172 00:00:00.020045 IP 10.64.72.24.30230 > 10.64.0.42.5078: UDP, length 172 00:00:00.020012 IP 10.64.72.24.30230 > 10.64.0.42.5078: UDP, length 172 00:00:00.019974 IP 10.64.72.24.30230 > 10.64.0.42.5078: UDP, length 172 00:00:00.020054 IP 10.64.72.24.30230 > 10.64.0.42.5078: UDP, length 172 00:00:00.019996 IP 10.64.72.24.30230 > 10.64.0.42.5078: UDP, length 172 00:00:00.019972 IP 10.64.72.24.30230 > 10.64.0.42.5078: UDP, length 172 The average time in both cases is aproximate 20ms (slighter less in direct case), but non regular data portions seems better for client because there is no gaps in heard voice (in speaker or headphones). I suppose that regular period portions of data cause problems due to clock drift problem. **So I want to turn off this feature in FreeSwitch, so the data will come in original timestamps.**

My laptop (a Thinkpad T440) has a SIM card slot which is meant to be used for Internet, though I haven't tested that yet since I don't have any SIM cards with Internet contracts. However, is it possible to use my laptop as a phone with a SIM card? Or would I need a different antenna for that? EDIT: Oh, and I'm running Arch on it (in case that makes a difference).

Is there is any Telephony API for Linux like we have "Tapi.h" for Windows?

I've installed minimodem on a server running Ubuntu. I've read through the documentation but still having trouble. I want to input a text file and use frequency shift keying to output an audio file.

I want to send sSMS periodically to certain Mobile numbers (Indian mobile numbers.) Is there a way I can send an SMS with my own cellphone number or by creating an account on a site like way2sms?

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 using my softphone through my PSTN line? Help me to find the right configuration into my inbound roules or trunks.

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: 1. A quiet Linux computer is on all the time in both places. 2. The user at one computer should be able to switch to an already running application and call the other party in the other country. 3. The computer in the other country should ring. 4. The person in the other country hearing the ring, should be able to answer the call. Again - nothing more complicated than pressing a button in an application. 5. Video option is a bonus but not a must. I unfortunately have little experience with VoIP. But I could do a fair job setting up firewalls as it might be necessary in such scenario. I am also used to various Linux distros. The question really is what applications could fit my purpose and if I must run a server (voip?) in one of the countries. As for users - less choice is better, if I could I would use this just for communication between these 2 "computer stations". Maybe even with telephone headsets :) Many thanks for your suggestions.