Sip_phone_extension

Sip Phone extension to access your node directly

Configuring a SIP phone on my network 

An entry needed to be added to sip.conf and extensions.conf and make sure in modules.conf the sip channel driver is enabled.
The phone needed to be configured, and that was accomplished through the web based portal using the local ip address of the phone.
 
Now I can just dial the extension that I configured and gain access to my node from my desktop sip phone.   Very handy feature.     I used my node numbers as the extensions to dial

SIP.CONF

[41899] ; Change to your node number or any extension number you want
deny=0.0.0.0/0.0.0.0
username=UUUUUU ;USERNAME you create
secret=SSSSSSSS ; SECRET PASSWORD you create
dtmfmode=rfc2833
canreinvite=no
context=radio-control
host=dynamic ; unless you set it to a static ip it is dynamic / dhcp
trustrpid=yes
sendrpid=no
type=friend
nat=no
port=5060
qualify=yes
qualifyfreq=60
transport=udp
encryption=no
callgroup=
pickupgroup=
dial=SIP/NNNNN ; NODE NUMBER
mailbox=NNNNN@device ;N NODE NUMBER
permit=0.0.0.0/0.0.0.0
callerid=XXXXX ; pick your own cid name CALLSIGN-NODE#

EXTENSIONS.CONF

[radio-control] ;W2YMM ADDED FOR SIP PHONE
exten => 41899,1,Answer
exten => 41899,n,Wait(2)
exten => 41899,n,Playback(rpt/node)
exten => 41899,n,Playback(digits/4) ;say node digits one at a time
exten => 41899,n,Playback(digits/1)
exten => 41899,n,Playback(digits/8)
exten => 41899,n,Playback(digits/9)
exten => 41899,n,Playback(digits/9)
exten => 41899,n,Rpt,41899|P|${CALLERID(name)} ;

The extension number you pick in sip.conf will need to match the exten=> here

Also make sure in /etc/asterisk/modules.conf you have sip set to load,                    

Comment out noload=chan_sip.so if its in there.

;noload=chan_sip.so

NOTE 41899 IS MY NODE NUMBER/EXTENSION, CHANGE DIGITS TO YOURS

The configuration is fairly straight forward.

You’ll need

  • Ip address of your Allstar Node
  • Extension Number
  • Password

Here are some screen shots   (This happens to be from my node 40822 not 41899)

Phone configs will vary depending on the manufacturer, I have this working on  Polycom, Cisco, and Yealink.

sip_config_allstar1sip_config_allstar2

The way I have this setup, I pick up the receiver, or put on speaker phone, dial my extension and the phone connects directly to my node, the node number is announced on the telephone.   To talk I press *99 and to receive I press #.

You could change it to a vox setup by editing the line in extension.conf

exten => 41899,n,Rpt,41899|P   (Change the P to a Pv)  But if you do that, you’ll have to utilize the mute button on your phone.  Vox works well as long as you don’t have a high background noise level.

 

To do some troubleshooting you can log into your CLI and issue some commands

sip show peers

sip show peer 41899      (Change to your extension #)

help sip show           for additional options

I needed to reboot the node after the configuration was completed

Hope this helps you get started. W2YMM