Workshop
SIP registration
Add to sip.conf:
[sipphone] type=friend host=dynamic context=from-sipphone secret=geheim insecure=invite
Add to extensions.conf:
[from-sipphone] ; Here will be the dialplan rules that are relevant ; when we call a number from the SIP phone. exten => 1234,1,Answer exten => 1234,n,Wait(1) exten => 1234,n,Playback(demo-thanks) exten => 1234,n,Hangup
Open your Asterisk console using asterisk -vvvR (may need to run as root), then:
yourpbx*CLI> sip reload yourpbx*CLI> dialplan reload yourpbx*CLI> sip show peers Name/username Host Dyn Forcerport ACL Port Status sipphone/sipphone (Unspecified) D N 0 Unmonitored
Now, configure your phone to connect to your Asterisk instance (over UDP). Watch your Asterisk log. When it connects, you will see a message. If it doesn't, use sip set debug on to find out whether you're receiving INVITE messages and see their contents.
As soon as the SIP phone is connected, call 1234 on it. You will hear a thank-you message from Asterisk.
Calling smbp
The rules in your dialplan (Answer, Wait, Playback, Hangup) are applications. There are many applications that you can use during a call. To get a list of applications, open your Asterisk CLI and run:
yourpbx*CLI> core show applications
One of the applications is called Dial. You use it to forward the call to other channels, for example SIP channels. The top of sip.conf explains how to use Dial to call SIP channels. Hint: the extension rule starts with exten => 1234,n,Dial(SIP/...
smbp's IP address is 192.168.178.172. Call the extension "1234" on it to receive the same message. Don't forget to reload your dialplan after changing it using dialplan reload.
Calling PSTN
smbp can forward to the public network if you authenticate to it. Claim one of the numbers on the slide. Then, add a line like this to your sip.conf below [general]:
register => NUMBER:PASSWORD@192.168.178.172
For example:
register => 31632271690:foobar@192.168.178.172
Then, tell me the password you chose and I will make sure you can connect.
