Workshop - Bewerkingsoverzicht

Xopr: categorie toegevoegd (overduidelijk asterisk)

2015-10-16T10:27:07Z

categorie toegevoegd (overduidelijk asterisk)

Boudewijn: Category:DIY toegevoegd

2014-09-22T21:35:07Z

Category:DIY toegevoegd

SGielen: /* Calling PSTN */

2013-09-16T20:01:10Z

‎Calling PSTN

SGielen op 16 sep 2013 om 19:53

2013-09-16T19:53:35Z

SGielen op 16 sep 2013 om 19:24

2013-09-16T19:24:08Z

SGielen: Start of Workshop

2013-09-16T18:06:04Z

Start of Workshop

Nieuwe pagina

= SIP registration =

Add to <tt>sip.conf</tt>:

<pre>
[sipphone]
type=friend
host=dynamic
context=from-sipphone
secret=geheim
insecure=invite
</pre>

Add to <tt>extensions.conf</tt>:

<pre>
[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
</pre>

Open your Asterisk console using <tt>asterisk -vvvR</tt> (may need to run as root), then:

<pre>
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
</pre>

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 <tt>sip set debug on</tt> 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.

← Oudere versie		Versie van 16 okt 2015 om 10:27
Regel 67:		Regel 67:

	[[Category:DIY]]		[[Category:DIY]]
		+	[[Category:Asterisk]]

← Oudere versie		Versie van 22 sep 2014 om 21:35
Regel 65:		Regel 65:

	Then, tell me the password you chose and I will make sure you can connect.		Then, tell me the password you chose and I will make sure you can connect.
		+
		+	[[Category:DIY]]

@@ Regel 52: / Regel 52: @@
 = 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:
+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 <tt>sip.conf</tt> below <tt>[general]</tt>:
 <pre>

← Oudere versie		Versie van 16 sep 2013 om 19:53
Regel 48:		Regel 48:
	One of the applications is called <tt>Dial</tt>. You use it to forward the call to other channels, for example SIP channels. The top of <tt>sip.conf</tt> explains how to use <tt>Dial</tt> to call SIP channels. Hint: the extension rule starts with <tt>exten => 1234,n,Dial(SIP/</tt>...		One of the applications is called <tt>Dial</tt>. You use it to forward the call to other channels, for example SIP channels. The top of <tt>sip.conf</tt> explains how to use <tt>Dial</tt> to call SIP channels. Hint: the extension rule starts with <tt>exten => 1234,n,Dial(SIP/</tt>...

−	smbp's IP address is 192.168.178.~~170~~. Call the extension "1234" on it to receive the same message. Don't forget to reload your dialplan after changing it using <tt>dialplan reload</tt>.	+	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 <tt>dialplan reload</tt>.
		+
		+	= 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:
		+
		+	<pre>
		+	register => NUMBER:PASSWORD@192.168.178.172
		+	</pre>
		+
		+	For example:
		+
		+	<pre>
		+	register => 31632271690:foobar@192.168.178.172
		+	</pre>
		+
		+	Then, tell me the password you chose and I will make sure you can connect.

← Oudere versie		Versie van 16 sep 2013 om 19:24
Regel 37:		Regel 37:

	As soon as the SIP phone is connected, call 1234 on it. You will hear a thank-you message from Asterisk.		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 (<tt>Answer, Wait, Playback, Hangup</tt>) are <strong>applications</strong>. There are many applications that you can use during a call. To get a list of applications, open your Asterisk CLI and run:
		+
		+	<pre>
		+	yourpbx*CLI> core show applications
		+	</pre>
		+
		+	One of the applications is called <tt>Dial</tt>. You use it to forward the call to other channels, for example SIP channels. The top of <tt>sip.conf</tt> explains how to use <tt>Dial</tt> to call SIP channels. Hint: the extension rule starts with <tt>exten => 1234,n,Dial(SIP/</tt>...
		+
		+	smbp's IP address is 192.168.178.170. Call the extension "1234" on it to receive the same message. Don't forget to reload your dialplan after changing it using <tt>dialplan reload</tt>.