AllStar SIP Phone Configuration

AllStar SIP Phone Configuration

This example configuration shows how to set up a sip phone extension and allow it to connect with an AllStar node on the same server.

Most of this example is standard Asterisk PBX configuration. Only the example line exten => 2522,1,rpt(2522|P) is unique to app_rpt. So if you already have your PBX working, all you'll need to do is add the above line to your dialplan phone context ([sip-phones] in this example) and restart app_rpt.

Edit modules.conf to be sure the sip module is loaded. The sip port on your router need not (should not) be open.

load => chan_sip.so      ; Session Initiation Protocol (SIP)

Edit sip.conf to allow your sip phone to login.

[210]
type=friend
host=dynamic
username=210
secret=<your secret passwd>  ; make it yours 
dtmfmode=rfc2833
mailbox=210                  ; Mailbox for message waiting indicator
context=sip-phones           ; Points to the stanza in extensions.conf
callerid="Tim Line 1" <210>

Edit extensions.conf to add your line(s) to the dialplan. The last 3 lines are the magic part that allows connection to your node. Please be sure to change the example node numbers to yours.

[sip-phones]
; Extension 210 - Tim's line 1
; Extension 211 - Tim's line 2
; Extension 212 - Garage phone
; Extension 213 - Cordless Phones ATA
; Extension 1000 - Voice Mail
exten => 210,1,Dial(SIP/210,60,rT)
exten => 211,1,Dial(SIP/211,60,rT)
exten => 212,1,Dial(SIP/212,60,rT)
exten => 213,1,Dial(SIP/213,60,rT)
exten => 1000,1,Voicemailmain(210)
exten => 1000,2,Hangup

; Allow SIP calls to local nodes
exten => 25330,1,rpt(25330|P)
exten => 25331,1,rpt(25331|P)
exten => 2522,1,rpt(2522|P)

Now you can dial your node number and go off-hook. Your are connected your node. Dial you node around just as if you were on the radio. You must use *99 to make PTT and # to unkey.