I went through a lot of pain before I could get it working. There is a Asterisk 1.4 version on the debian repo but I wanted to use a newer version. I found this blog post and followed it to the letter:
http://rutmandal.info/infotalk/2009/03/ ... debian-40/
Once you have compiled from source (it takes a while!) open asterisk.conf and edit the top line accordingly:
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[directories](!) ; remove the (!) to enable this
http://www.voip-info.org/wiki/view/Asterisk+non-root
After you've done that reboot Bubba. I had to do that for everything to work.
You should be able to do this now (as 'root' or 'asterisk' if you've changed permissions per the step above):
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bubba:/# asterisk -r
Asterisk 1.6.0.6, Copyright (C) 1999 - 2008 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 1.6.0.6 currently running on bubba (pid = 2529)
Verbosity is at least 9
bubba*CLI>
You can add verbosity with:
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asterisk -rvvv
After you've got asterisk up and running you now have to update sip.conf and extensions.conf (according to your SIP providers instructions). If you're using Skype for SIP take a look at this post:
http://forum.skype.com/index.php?showtopic=487451
I modified my confs slightly.
This is what I added to the top of sip.conf:
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disallow=all
allow=ulaw
allow=alaw
allow=g729
register => 9905100000XXXX:LxXTF999QXXXX@sip.skype.com/9905100000XXX
externip = XX.XX.XX.XX
[100]
type=peer
host=dynamic
secret=100
context=users
mailbox=100@default
[SkypeForSIP]
type=friend
dtmfmode=rfc2833
host=sip.skype.com
username=9905100000XXXX
fromuser=9905100000XXXX
secret=LxXTF999QXXXX
canreinvite=no
allow=ulaw
allow=alaw
allow=g729
insecure=invite
fromdomain=sip.skype.com
context=users
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[users]
include => local
[local]
exten => _0.,1,Dial(SIP/SkypeForSIP/+44${EXTEN:1})
exten => _00.,1,Dial(SIP/SkypeForSIP/+${EXTEN:2})
exten => 9905100000XXXX,1,Dial(SIP/100, 20)
exten => 9905100000XXXX,2,VoiceMail(100@mb_config)
exten => 9905100000XXXX,3,PlayBack(vm-goodbye)
exten => 9905100000XXXX,4,Hangup()
exten => *97,1,VoiceMailMain(100@mb_conf)
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[mb_config]
100 => 100, 100, youremail@yourdomain.com
I'm using a Cisco IP phone and it works great. Also checkout 'Telephone' (code.google.com/p/telephone/) soft IP phone if you're on a Mac.
Hope this helps.
Ben