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<html xmlns="http://www.w3.org/1999/xhtml"><head><meta http-equiv="Content-Type" content="text/html; charset=UTF-8" /><title>Connecting Two Asterisk Boxes Together via SIP</title><link rel="stylesheet" href="styles.css" type="text/css" /><meta name="generator" content="DocBook XSL Stylesheets V1.69.1" /><link rel="start" href="index.html" title="Asterisk™: The Future of Telephony" /><link rel="up" href="asterisk-CHP-4.html" title="Chapter 4. Initial Configuration of Asterisk" /><link rel="prev" href="I_sect14_tt599.html" title="Connecting to a SIP Service Provider" /><link rel="next" href="I_sect14_tt645.html" title="Configuring an IAX Softphone" /></head><body><div class="navheader"><table width="100%" summary="Navigation header"><tr><th colspan="3" align="center">Connecting Two Asterisk Boxes Together via SIP</th></tr><tr><td width="20%" align="left"><a accesskey="p" href="I_sect14_tt599.html">Prev</a> </td><th width="60%" align="center">Chapter 4. Initial Configuration of Asterisk</th><td width="20%" align="right"> <a accesskey="n" href="I_sect14_tt645.html">Next</a></td></tr></table><hr /></div><div class="sect1" lang="en" xml:lang="en"><div class="titlepage"><div><div><h2 class="title" style="clear: both"><a id="connecting_two_asterisk"></a>Connecting Two Asterisk Boxes Together via SIP</h2></div></div></div><p>There may come a time<a id="ch04_siptwo" class="indexterm"></a> when you have a pair of Asterisk boxes, and you’d like to
pass calls between them. Luckily this isn’t very difficult, although it
does have some oddities that we need to deal with, but from the
configuration viewpoint it isn’t really all that difficult.</p><div class="sidebar"><a id="I_sidebar4_tt624"></a><p class="title"><b>Configuring a Local Firewall</b></p><p>If you’re running <a id="I_indexterm4_tt625" class="indexterm"></a>iptables on the same machine as the Asterisk box, then you
can run the following commands to open port 5060 for SIP signaling, and
ports 10,000 through 20,000 for the RTP traffic. You can also narrow the
range <a id="I_indexterm4_tt626" class="indexterm"></a>of RTP ports in the <span class="emphasis"><em>rtp.conf</em></span> file
located in <span class="emphasis"><em>/etc/asterisk</em></span>. An excellent book on
iptables firewalls <a id="I_indexterm4_tt627" class="indexterm"></a><a id="I_indexterm4_tt628" class="indexterm"></a><a id="I_indexterm4_tt629" class="indexterm"></a>is <span class="emphasis"><em>Linux Firewalls</em></span> by Steve Suehring
and Robert Ziegler (Novell Press):</p><p><a id="I_programlisting4_tt630"></a></p><pre class="programlisting"># <strong class="userinput"><code>iptables -I RH-Firewall-1-INPUT -p udp --dport 5060 -j ACCEPT</code></strong>
# <strong class="userinput"><code>iptables -I RH-Firewall-1-INPUT -p udp --dport 10000:20000 -j ACCEPT</code></strong>
# <strong class="userinput"><code>service iptables save</code></strong></pre><p>Be aware that
this will allow all UDP traffic from any source access to ports 5060 and
10,000 through 20,000.</p></div><p>Our topology will consist of a SIP phone (Alice) registered to
Asterisk A (Toronto), and a separate SIP phone (Bob) registered to
Asterisk B (Osaka). At the end of this section, you will be able to set up
a call from Alice to Bob (and vice versa) through your pair of Asterisk
boxes (see <a href="connecting_two_asterisk.html#figure_4_5" title="Figure 4.5. SIP trunking topology">Figure 4.5, “SIP trunking topology”</a>). This is a common scenario when
you have two physical locations, such as a company with multiple offices
that wants a single logical extension topology.</p><div class="figure"><a id="figure_4_5"></a><p class="title"><b>Figure 4.5. SIP trunking topology</b></p><div class="mediaobject"><a id="I_mediaobject4_tt631"></a><img src="figs/web/ast2_0405.png" alt="SIP trunking topology" /></div></div><p>First, let’s configure our Asterisk boxes.</p><div class="sect2" lang="en" xml:lang="en"><div class="titlepage"><div><div><h3 class="title"><a id="id4121686"></a>Configuring Our Asterisk Boxes</h3></div></div></div><p>We have a pair of Asterisk boxes that we’re going to call Toronto
and Osaka and that we’re going to have register to each other. We’re
going to use the most basic <span class="emphasis"><em>sip.conf</em></span> file
<a id="I_indexterm4_tt632" class="indexterm"></a>that will work in this scenario. Just like the SIP phone
configuration earlier in this chapter, it’s not necessarily the best way
to do it, but it’ll work.</p><p>Here is the configuration for the Toronto box:</p><p><a id="I_programlisting4_tt633"></a></p><pre class="programlisting">[general]
register => toronto:welcome@192.168.1.101/osaka
[osaka]
type=friend
secret=welcome
context=osaka_incoming
host=dynamic
disallow=all
allow=ulaw
</pre><p>And the configuration for the Osaka box:</p><p><a id="I_programlisting4_tt634"></a></p><pre class="programlisting">[general]
register => osaka:welcome@192.168.2.202/toronto
[toronto]
type=friend
secret=welcome
context=toronto_incoming
host=dynamic
disallow=all
allow=ulaw
</pre><p>Many of the previous options may be familiar to you by now,
but let’s take a look at them further just in case they are not.</p><p>The second line of the file tells our Asterisk box to register to
the other box, with the purpose of telling the remote Asterisk box where
to send calls when it wishes to send a call to our local Asterisk box.
Remember how we mentioned a little oddity in the configuration? Notice
that at the end of the registration line we tag on a forward slash and
the username of the remote Asterisk box? What this does is tell the
remote Asterisk box what digest name to use when it wants to set up a
call. If you forget to add this, then when the far end tries to send you
a call, you’ll see the following at your Asterisk CLI:</p><p><a id="I_programlisting4_tt635"></a></p><pre class="programlisting">[Apr 22 18:52:32] WARNING[23631]: chan_sip.c:8117 check_auth: username mismatch,
have <toronto>, digest has <s></pre><p>So
by adding the forward slash and username, we tell the other end what to
place in the Digest username of the Proxy Authorization field in the SIP
INVITE message.</p><p>The rest of the file is the authorization block we use to control
the incoming and outgoing calls from the other Asterisk box. On the
Toronto box, we have the <code class="literal">[osaka]</code> authorization block,
and on the Osaka box, we have the <code class="literal">[toronto]</code> block. We
define the type as a <code class="literal">friend</code>, which allows us to both
receive and place calls from the other Asterisk box. The
<code class="literal">secret</code> is the password the other system should use
when authenticating. The <code class="literal">context</code> is where incoming
calls are processed in the dialplan
(<span class="emphasis"><em>extensions.conf</em></span>). We set the
<code class="literal">host</code> parameter to <code class="literal">dynamic</code>, which
tells our Asterisk box that the other endpoint will register to us,
thereby telling us what IP address to set up calls when we want to send
a call to the other end. Finally, the <code class="literal">disallow</code> and
<code class="literal">allow</code> parameters control the codecs we wish to use
with the other end.</p><p>If you save the file and reload the SIP channel on both Asterisk
boxes (<strong class="userinput"><code>sip reload</code></strong> from the
Asterisk console), you should see something like the following, which
will tell you the remote box successfully registered:</p><p>
<a id="I_programlisting4_tt636"></a></p><pre class="programlisting">*CLI> -- Saved useragent "Asterisk PBX" for peer toronto</pre><p>
</p><p>You should see the status of the Host change from
(<code class="literal">Unspecified</code>) to the IP address of the remote box
when you run <code class="literal">sip show peers</code>:</p><p><a id="I_programlisting4_tt637"></a></p><pre class="programlisting">*CLI> <strong class="userinput"><code>sip show peers</code></strong>
Name/username Host Dyn Nat ACL Port Status
toronto/osaka 192.168.2.202 D 5060 Unmonitored</pre><p>You
can verify that your own registration was successful by running
<code class="literal">sip show</code><code class="literal">registry</code> from the Asterisk
console:</p><p><a id="I_programlisting4_tt638"></a></p><pre class="programlisting">*CLI> <strong class="userinput"><code>sip show registry</code></strong>
Host Username Refresh State Reg.Time
192.168.1.101:5060 osaka 105 Registered Sun, 22 Apr 2007 19:13:20</pre><p>Now
that our Asterisk boxes are happy with each other, let’s configure a
couple of SIP phones so we can call between the boxes.</p></div><div class="sect2" lang="en" xml:lang="en"><div class="titlepage"><div><div><h3 class="title"><a id="id4121946"></a>SIP Phone Configuration</h3></div></div></div><p>See the <a href="asterisk-CHP-4-SECT-6.html" title="Configuring an FXS Channel for an Analog Telephone">the section called “Configuring an FXS Channel for an Analog Telephone”</a>” section of this
chapter for more information about configuring SIP phones with Asterisk.
Below is the configuration for two SIP phones in the
<span class="emphasis"><em>sip.conf</em></span> file for each server, which we’ll be
referencing from the dialplan in the next section, thereby giving us two
endpoints to call between. Append this configuration to the end of the
<span class="emphasis"><em>sip.conf</em></span> file on each respective server.</p><p>Toronto <span class="emphasis"><em>sip.conf</em></span>:</p><p><a id="I_programlisting4_tt639"></a></p><pre class="programlisting">[1000]
type=friend
host=dynamic
context=phones</pre><p>Osaka <span class="emphasis"><em>sip.conf</em></span>:</p><p><a id="I_programlisting4_tt640"></a></p><pre class="programlisting">[1001]
type=friend
host=dynamic
context=phones</pre><p>You should now have extension 1000 registered
to Toronto, and extension 1001 registered to Osaka. You can verify this
with the <code class="literal">sip show peers</code> command from the Asterisk
console. Next, we’re going to configure the dialplan logic that will
allow us to call between the extensions.</p></div><div class="sect2" lang="en" xml:lang="en"><div class="titlepage"><div><div><h3 class="title"><a id="id4122019"></a>Configuring the Dialplan</h3></div></div></div><p>Now we can configure a simple dialplan<a id="I_indexterm4_tt641" class="indexterm"></a> for each server allowing us to call between the two
phones we have registered: one to Toronto, the other to Osaka. In the
<a href="asterisk-CHP-4-SECT-2.html" title="Working with Interface Configuration Files">the section called “Working with Interface Configuration Files”</a>” section of this chapter, we
asked you to create a simple <code class="filename">extensions.conf</code> file. We are going to build
up a dialplan based on this simple configuration. The dialplan for each
server will be very similar to the other one, but for clarity we will
show both. The new lines we’re adding to the file will be
italicized.</p><p>Toronto <code class="filename">extensions.conf</code>:</p><a id="I_programlisting4_tt642"></a><pre class="programlisting">[globals]
[general]
autofallthrough=yes
[default]
[incoming_calls]
[phones]
include => internal
<span class="emphasis"><em>include => remote
</em></span><span class="emphasis"><em>[internal]
exten => _2XXX,1,NoOp()
exten => _2XXX,n,Dial(SIP/${EXTEN},30)
exten => _2XXX,n,Playback(the-party-you-are-calling&is-curntly-unavail)
exten => _2XXX,n,Hangup()
[remote]
exten => _1XXX,1,NoOp()
exten => _1XXX,n,Dial(SIP/osaka/${EXTEN})
exten => _1XXX,n,Hangup()
[osaka_incoming]
include => internal</em></span></pre><p>Osaka <code class="filename">extensions.conf</code>:</p><a id="I_programlisting4_tt643"></a><pre class="programlisting">[globals]
[general]
autofallthrough=yes
[default]
[incoming_calls]
[phones]
include => internal
<span class="emphasis"><em>include => remote</em></span>
[internal]
<span class="emphasis"><em>exten => _1XXX,1,NoOp()
exten => _1XXX,n,Dial(SIP/${EXTEN},30)
exten => _1XXX,n,Playback(the-party-you-are-calling&is-curntly-unavail)
exten => _1XXX,n,Hangup()
[remote]
exten => _2XXX,1,NoOp()
exten => _2XXX,n,Dial(SIP/toronto/${EXTEN})
exten => _2XXX,n,Hangup()
[toronto_incoming]
include => internal</em></span></pre><p>Once you’ve configured your <code class="filename">extensions.conf</code> file, you can reload it from
the Asterisk console with the <code class="literal">dialplan reload</code>
command. Verify your dialplan loaded with the <code class="literal">dialplan
show</code> command.</p><p>And that’s it! You should be able to place calls between your two
Asterisk servers now.<a id="I_indexterm4_tt644" class="indexterm"></a></p></div></div><div class="navfooter"><hr /><table width="100%" summary="Navigation footer"><tr><td width="40%" align="left"><a accesskey="p" href="I_sect14_tt599.html">Prev</a> </td><td width="20%" align="center"><a accesskey="u" href="asterisk-CHP-4.html">Up</a></td><td width="40%" align="right"> <a accesskey="n" href="I_sect14_tt645.html">Next</a></td></tr><tr><td width="40%" align="left" valign="top">Connecting to a SIP Service Provider </td><td width="20%" align="center"><a accesskey="h" href="index.html">Home</a></td><td width="40%" align="right" valign="top"> Configuring an IAX Softphone</td></tr></table></div><div xmlns="" id="svn-footer"><hr /><p>You are reading <em>Asterisk: The Future of Telephony</em> (2nd Edition for Asterisk 1.4), by Jim van Meggelen, Jared Smith, and Leif Madsen.<br />
This work is licensed under the <a href="http://creativecommons.org/licenses/by-nc-nd/3.0/">Creative Commons Attribution-Noncommercial-No Derivative Works License v3.0</a>.<br />
To submit comments, corrections, or other contributions to the text, please visit <a href="http://oreilly.com/catalog/9780596510480/">http://www.oreilly.com/</a>.</p></div></body></html>
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