[asterisk-users] q: install asterisk + asteris-gui

[Danny Nicholas]

If you're just going to use Asterisk as an internal system, you just need a
simple users.conf, sip.conf and about a 5 line dialplan.

 

Sip.conf

[general]

srvlookup=yes ;allows DNS lookups of server names

naxexpirey=180

defaultexpirey=160

context=default ; Default context for incoming calls

allowoverlap=no ; Disable overlap dialing support. (Default is yes)

bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)

 

bindaddr=192.168.23.95 ; IP address to bind to (0.0.0.0 binds to all)

srvlookup=yes ; Enable DNS SRV lookups on outbound calls

limitonpeers=yes

notifyringing=yes

rtupdate=yes

artcachefriends=yes

notifyhold=yes

incominglimit=1

call-limit=3

 

 

[authentication]

 

[104]

type=peer

context=phones

host=dynamic

fromuser=104

secret=xxxx

canreinvite=yes

directrtpsetup=no

call-limit=3

nat=yes

qualify=yes

register=no

session-timers=accept

session-expires=60

session-minse=120

session-refresher=uac

register => 1001:xxxx at yourpbx.com/1001

defaultip=192.168.23.114

mailbox=1001

disallow=all

allow=ulaw,alaw

 

rinse and repeat for 1002-1005

 

users.conf

[general]

; Full name of a user

fullname = Unknown User

; Starting point of allocation of extensions

userbase = 1001

; Create voicemail mailbox and use use macro-stdexten

hasvoicemail = yes

; Set voicemail mailbox 1001 password to 1234

vmsecret = 1234

; Create SIP Peer

hassip = yes

; Create IAX friend

hasiax = no

; Create Agent friend

hasagent = no

; Create H.323 friend

;hash323 = yes

; Create manager entry

hasmanager = no

; Remaining options are not specific to users.conf entries but are general.

callwaiting = yes

threewaycalling = yes

callwaitingcallerid = yes

transfer = yes

canpark = yes

cancallforward = yes

callreturn = yes

callgroup = 1

pickupgroup = 1

localextenlength = 4

 

[1001]

username=1001

transfer=yes

mailbox=1001

call-limit=3

fullname=user 1

registersip=no

host=dynamic

callgroup=1

context=default

cid_number=1001

hasvoicemail=yes

vmsecret=1234

email=user1 at yourpbx.com

threewaycalling=yes

hasdirectory=no

callwaiting=yes

hasmanager=yes

managerread=system,call,log,verbose,command,agent,user,config

managerwrite=system,call,log,verbose,command,agent,user,config

hasagent=yes

hassip=yes

hasiax=no

secret=xxxx

nat=yes

canreinvite=no

dtmfmode=rfc2833

insecure=no

pickupgroup=1

macaddress=000000001111

autoprov=yes

label=100

linenumber=1

disallow=all

allow=ulaw,gsm

 

repeat for 1002-1005

 

extensions.conf

[default]

Exten => s,1,answer

Exten => s,n,hangup

Exten => _1XXX,1,Dial(SIP/${EXTEN},60.m)

 

  _____  

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of tom
Sent: Wednesday, July 08, 2009 3:38 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] q: install asterisk + asteris-gui

 

thx again,

one last question: as i mentioned, i used freepbx before. now i facing only
the section: 
- users

> my goal right now is to use that asterisk instance just to have intenral
extensions to talk to each other...whats the quickest setup here? i mean i
dont need trunks, dialplans etc, right?

i just need 5 internal extension, eg 1001-1005

thx 
u guys are great!

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