AllStar SIP Phone Configuration: Difference between revisions
No edit summary |
No edit summary |
||
| Line 3: | Line 3: | ||
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. | 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 <code>exten => 2522,1,rpt(2522|P)</code> 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 (<code>[sip-phones]</code> in this example) and restart app_rpt. | Most of this example is standard Asterisk PBX configuration. Only the example line <code>exten => 2522,1,rpt(2522|P)</code> 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 (<code>[sip-phones]</code> in this example) and restart app_rpt or do <code>dialplan reload</code>. | ||
Revision as of 02:07, 28 August 2021
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 or do dialplan reload.
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 to your AllStar PBX.
[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.