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commented
8 years ago changed title to match roadmap.
more help for this Asterisk discovery ticket:
* *install asterisk* (on linux, get the asterisk package in your package manager
(ie synaptic on gnome))
* *go to /etc/asterisk an look at extensions.conf* there, at the end of the file
you have a section called [default] or something like that.
There is help on the extensions.conf syntax and more on :
http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf
#PredefinedExtensionNames
* look at the extensions written there: exten => someNumber, somePriority,
someAsteriskCommand ..
imagine you have a line like exten => 1234,s,what
* *install a softphone* through your package manager (kphone is cool and available
from the Ubuntu package manager, ekiga is buggy, don't use it, x-lite is used
by ECE
students here in ing5 as I've heard and has linux & windows versions)
* make sure asterisk runs (ie. pstree | grep asterisk for example), if it's not
running, you can start by typing: /etc/init.d/asterisk start
* asterisk is listening on port 5060 normally (you can check that with: sudo
netstat -ntpa | grep asterisk)
* *edit sip.conf* to add an SIP user(=phone) (add what's below to the file)
[101] #101 is a username, change to something ascii if you want
type=friend
secret=welcome #this is the passwor for user 101 (or whatever name has been
defined
in the [] above)
qualify=yes ; Qualify peer is not more than 2000 mS away
nat=no ; This phone is not natted
host=dynamic ; This device registers with us
canreinvite=no ; Asterisk by default tries to redirect
context=home
port=5061 ; Uncomment this line if Ekiga and Asterisk are on the same host
#Note for last line,
#(a softphone is like a server in some way, because it can receive calls, so it
has
to listen to some port, asterisk listens to some port as well (for SIP this
port is
5060 by default)... we must have those two ports different, we decide to make
asterisk listen on a different port, and we won't touch the softphone config).
#We ask asterisk to listen on 5061 instead of default 5060 port because our
softphone
maybe stupid and it's maybe harder to setup the softphone to listen on/call
from a
different port.
For info on SIP open ports, see
http://www.chebucto.ns.ca/~rakerman/port-table.html
and look for SIP (ctrl+f).
* save the sip.conf file and restart asterisk: type /etc/init.d/asterisk restart
(note there is a way to reload the sip.conf file without restarting asterisk
also...)
* you can check that asterisk is listening on port 5061 (sudo netstat -ntpa | grep
5061)
* you can get into asterisk's interactive console by typing "asterisk -vvvvr" (at
actually any number of v for verboseness, tutorials on the net put 4 or 5 v ,
but put
at least the r to have access to asterisk's command line)
* run kphone (or the softphone you took), create a phone/connection with username
to 101 (or whatever you defined in sip.conf between [brackets]) and password to
"welcome" (without the "")
you want to log into the server localhost on port 5061 (maybe you have to type
localhost:5061 instead of having two fields).
- call the number 1234@localhost
- you hear thing and can typed keys on the DTMF big telephone buttons.
- the line exten => 1234,.... of the extensions.conf file was actually played
when
you called. Notice that PlayBackground(someFileNameWithNo.Extension) plays a
file in
a non-modal/background way (in short in this case, doesn't block keyboard). the
file
played comes from */usr/share/asterisk/sounds/ where you have lots of .gsm sound
files*. You can play those gsm files in console with the command named "play"
(except
that you have to install it first... with command "sudo apt-get install sox" as
proposed by the shell when you try to type "play").
root@jonathan:/usr/share/asterisk/sounds# sudo apt-get install sox
...... #sox which contains the play program gets downloaded from the web and
installed on your computer
#now you can use play to play many types of files including gsm in console
root@jonathan:/usr/share/asterisk/sounds# play vm-goodbye.gsm
# you hear a girl saying "Good bye !"
Input File : 'vm-goodbye.gsm'
Sample Size : 8-bit (1 byte)
Sample Encoding: GSM
Channels : 1
Sample Rate : 8000
Time: 00:01.02 [00:00.00] of 00:00.00 (0.00%) Samples out: 49.0k Clips: 0
Done.
Original comment by jonathan.schroder@gmail.com on 20 Jan 2009 at 6:41
- Changed title: Install Asterisk + an SIP softphone, call a local Asterisk sever over SIP with the softphone
Original issue reported on code.google.com by
jonathan.schroder@gmail.comon 20 Jan 2009 at 6:31