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This PPA will continue to be updated with new upstream releases and can be found at https://launchpad.net/~gnutelephony/ archive/ppa.

The minimum requirement is to configure your sipwitch server as both the registrar and outbound proxy.When you update the sipwitch package through apt, the service is automatically stopped and restarted for you, but this will terminate any active phone calls.You can also change sipwitch configuration files at any time and reload them into the currently running server without any downtime.You would still enter the password in the authentication part of Twinkle.But at least the server config file will not have passwords, only digests, just like the /etc/shadow file has for user logins. It normally will automatically start when you reboot and can be managed from the standard desktop "services" applet.SIP Witch can be setup to operate over multiple nodes.

This allows one to have very large site installations, where each server runs part of a dialing plan. This is not likely a scenario for initial testing, so I am mostly referencing here that the capability exists.

If you are in the "sipwitch" group in /etc/group, then you will not need to be root to do this.

The list of commands available can be found with "man sipwitch". Of particular use is: sipwitch calls - lists active calls in progress (no output if none) sipwitch registry - lists registered extensions sipwitch status - single line state table sipwitch stats - running call stats (always useful even if idle...) sipwitch reload - reloads configuration after config files changed There are three primary log files: /var/log/- basic error and activation logging /var/log/sipwitch.stats - call statistics every hour /var/log/sipwitch.calls - record of each call These are automatically rotated based on /var/logrotate.d/sipwitch There is also a /etc/cron.hourly/sipwitch which generates hourly statistics.

We will later add PLUGINS="forward" here when we setup for use as a secure calling domain controller for Asterisk or Free Switch.

Then you should add a sipwitch "group" in /etc/group: This matches the GROUP="sipwitch" line found in /etc/default/sipwitch.

For example, a simple provisioning record for /etc/sipwitch.d/might look like: In this, "Id" is the logical user id (can be from email or login id), there is a secret, which is a password that is also used in the user agent, an extension # in the range of the dialing plan this entry is associated with, and a display name which appears when that user agent calls another.

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