Hello *,
I would like to announce to the IPFire community, that the IPFire.org
VoIP services have been relaunched with loads of bug fixes and some
exciting new features!
Everyone of the IPFire team members has an account on our
infrastructure to be able to log on to the systems, etc. Tied to that
is our Jabber server and SIP server. The latter of which has always had
some trouble with some phones under certain conditions, with calls
being dropped or one party not hearing anything. That was annoying and
led to many problems communicating to each other. I personally like to
use this service since we are all so close to each other, but often in
many different countries so that using a landline is very expensive.
So here is the boring details:
We are using Kamailio and Asterisk to provide the telephone services.
Kamailio is acting as a SIP proxy and registrar and Asterisk is
handling the dialplan and upstream connections to some SIP providers.
The relaunch was basically to replace SEMS (which we used before) by
Asterisk and to rewrite the Kamailio configuration file from scratch.
That took a long time and led to many problems itself, but finally we
have come to a point where there are no bigger known issues any more.
Yay!
To make a long story short: Here we are now with our brand new system.
And to make it even more awesome it has gotten some new features.
First of all: There is a connection to the plain old telephony network
now. That means that every SIP extension call be called from a landline
phone. Even with this nice SIP service it is sometimes a little bit
inconvenient to have a SIP phone around. This is no longer needed since
your normal phone is enough now.
The number is:
+49 2363 6035 XXX
Where XXX is the extension of each developer. You can decide if you
want to share that publicly or not. That is up to you. We are not
getting charged for any inbound calls. Outbound calls are not available
for everyone, but I have the option to activate those on a per-user
basis.
Extensions had four digits before. Since it was very expensive to get a
number block that was large enough I had to change them to three
digits only. Please update your configuration.
Incoming calls will query a phonebook so that the name of the caller is
shown when known.
The same connection allows us to call any services like the conference
server from a landline which is available under:
+49 2363 6035 900
This will allow people who do not have an account or who don't have it
set up to join the monthly conference call or any other conference
without any setup.
SIP over TLS is supported and outgoing calls can use SRTP to encrypt
the voice traffic. However, Asterisk is not able to initiate any calls
with optional SRTP, so that only one call leg can use the encryption
now.
On top of all of this, there is some standard PBX services available
like mailboxes, music and echo test for testing and you can also park
calls, etc.
Further documentation is here:
http://wiki.ipfire.org/en/community/talk.ipfire.org/start
It includes some guides how to configure phones and some technical
details about which codecs are supported, etc.
So, please make good use out of this service. It is free and it is
supposed to bring the IPFire team closer together to be able to work
quicker and better!
If you find any problems with your phone, please let me know.
Best,
-Michael
