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A phone dialer and call handler.


Calls is licensed under the GPLv3+.


To build Calls you need to first install the build-deps defined by the debian/control file

If you are running a Debian based distribution, you can easily install all those the dependencies making use of the following command

sudo apt-get build-dep .


We use the meson and thereby Ninja. The quickest way to get going is to do the following:

meson . _build
ninja -C _build
ninja -C _build install

If you don't want to pollute your filesystem please be aware, that you can also use --prefix=~/install.

Build the documentation

If you want to build the documentation you have to configure the meson project with -Dgtk_doc=true

meson . _build -Dgtk_doc=true
ninja -C _build
ninja -C _build calls-doc

You can also browse the documentation online

Running from the source tree

The most comfortable way to run from the source tree is by using the provided run script which sets up the environment for you:


Note: Invoking the run script might terminate after a few seconds and instead only activates an already running instance of gnome-calls. In this case you want to systemctl --user stop calls to stop it.


We want to encourage users to submit high quality bug reports. We understand that it can be daunting to make sense of a misbehaving application so following are some tips that can help to pin down the root of a problem.

Enable verbose logging by invoking Calls with -vvv argument (i.e. _build/run -vvv). Observe the output and check for anything suspicious.

If you have found something weird in the logs you cangrep the sources to find where the warning is emitted and - if you're so inclined - start exploring from there:

  • check surrounding code
  • add debugging output
  • set breakpoints and inspect stack traces and local variables

You can also easily run under gdb if you invoke like

CALLS_GDB=1 _build/run -vvv

In the case of crashes you should provide a backtrace ideally with debugging symbols (for debian based distros you have to add a suitable debugging suite to your apt sources; see link below).

If your system is using systemd you may find this guide useful: With coredumpctl coredumps can easily be analyzed at a later date.

For backend specific debugging, please see the sections below.

Call provider backends

Calls uses libpeas to support runtime loadable plugins which we call "providers". Calls currently ships four different plugins:

  • mm: The ModemManager plugin used for cellular modems
  • sip: The SIP plugin for VoIP
  • dummy: A dummy plugin
  • ofono: The oFono plugin used for cellular modems (not in active development)

By default Calls will load the mm and sip plugins. If you want to load other plugins you may specify the -p <PLUGIN> argument (you can pass multiple -p arguments) when invoking calls, f.e.

_build/run -p sip -p dummy
/usr/bin/gnome-calls -p mm

Every plugins uses the following concepts:

  • CallsProvider: The principal abstraction of a library allowing to place and receive calls.
  • CallsOrigin: Originates calls. Represents a single modem or VoIP account.
  • CallsCall: A call.

There is a one to many relation between provider and origins and between origins and calls. F.e. you have one SIP provider managing multiple SIP accounts (=origins) each of which can have multiple active calls (not yet implemented).


This is the default backend for cellular calls. It uses libmm-glib to talk to ModemManager over DBus. It currently only supports one modem and one active call at a time.


You can monitor the ModemManager messages on the DBus as follows:

gdbus monitor --system --dest org.freedesktop.ModemManager1

For complete debugging logs you can set ModemManager's log verbosity to DEBUG as follows:

mmcli -G DEBUG

and inspect the logs on a systemd based system with:

journalctl -u ModemManager.service

For more information see here


This plugin uses the sofia-sip library for SIP signalling and GStreamer for media handling. It supports multiple SIP accounts and currently one active call at a time (subject to change).


You can print the sent and received SIP messages by setting the environment variable TPORT_LOG=1. To test the audio quality you can use one of the various public reachable echo test services, f.e. echo@conference.sip2sip.info. Please note that the SIP plugin currently doesn't support DTMF, which is used for some test services for navigating through a menu.

If one or both sides can't hear any audio at all it is likely that the audio packets are not reaching the desired destination.


This plugin is mostly useful for development purposes and work on the UI as it allows simulating both outgoing and incoming calls. To trigger an incoming call you should send a USR1 signal to the calls process:

kill -SIGUSR1 $(pidof gnome-calls)


This plugin is not in active development anymore, so your mileage may vary. See here for more information.