AllStar SIP Phone Configuration: Difference between revisions
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{{Warning|adv-topics/sip-phone/}} | |||
= 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. | 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>. | ||
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load => chan_sip.so ; Session Initiation Protocol (SIP) | load => chan_sip.so ; Session Initiation Protocol (SIP) | ||
</pre> | </pre> | ||
Edit sip.conf to allow your sip phone to login to your AllStar PBX. | Edit sip.conf to allow your sip phone to login to your AllStar PBX. | ||
<pre> | <pre> | ||
In General section, use this to mask reason for rejecting (light hacker protection) . | |||
alwaysauthreject=yes | |||
Change to no if you try debuging sip phones so you can see the real reason for the reject. | |||
Settings for each sip extension. If not specified per sip, anything from General section will apply. | |||
[210] | [210] | ||
type=friend | type=friend | ||
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; Extension 213 - Cordless Phones ATA | ; Extension 213 - Cordless Phones ATA | ||
; Extension 1000 - Voice Mail | ; Extension 1000 - Voice Mail | ||
exten => 210,1,Dial(SIP/210,60,rT) | exten => 210,1,Dial(SIP/210,60,rT) | ||
exten => 211,1,Dial(SIP/211,60,rT) | exten => 211,1,Dial(SIP/211,60,rT) | ||
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exten => 1000,2,Hangup | exten => 1000,2,Hangup | ||
Make / add in extensions.conf for routing (considered dialplan in asterisk) | |||
; Allow SIP calls to local nodes | ; Allow SIP calls to local nodes | ||
exten => 25330,1,rpt(25330|P) | exten => 25330,1,rpt(25330|P) | ||
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</pre> | </pre> | ||
The 'P' tells asterisk to connect to node in 'phone control mode'. | |||
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. | 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. | ||
[[Category:How to]] | [[Category:How to]] | ||
Latest revision as of 16:33, 18 October 2024
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.
In General section, use this to mask reason for rejecting (light hacker protection) . alwaysauthreject=yes Change to no if you try debuging sip phones so you can see the real reason for the reject. Settings for each sip extension. If not specified per sip, anything from General section will apply. [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 Make / add in extensions.conf for routing (considered dialplan in asterisk) ; Allow SIP calls to local nodes exten => 25330,1,rpt(25330|P) exten => 25331,1,rpt(25331|P) exten => 2522,1,rpt(2522|P)
The 'P' tells asterisk to connect to node in 'phone control mode'. 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.