Asterisk: The Future of Telephony, 2nd Edition

Connecting Two Asterisk Boxes Together via SIP

There may come a time 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.

Configuring a Local Firewall

If you’re running 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 of RTP ports in the rtp.conf file located in /etc/asterisk. An excellent book on iptables firewalls is Linux Firewalls by Steve Suehring and Robert Ziegler (Novell Press):

# iptables -I RH-Firewall-1-INPUT -p udp --dport 5060 -j ACCEPT
# iptables -I RH-Firewall-1-INPUT -p udp --dport 10000:20000 -j ACCEPT
#service iptables save

Be aware that this will allow all UDP traffic from any source access to ports 5060 and 10,000 through 20,000.

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 Figure 4-5). 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.

Figure 4-5. SIP trunking topology

First, let’s configure our Asterisk boxes.

Configuring Our Asterisk Boxes

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 sip.conf file 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.

Here is the configuration for the Toronto box:

[general]
register => toronto:welcome@192.168.1.101/osaka

[osaka]
type=friend
secret=welcome
context=osaka_incoming
host=dynamic
disallow=all
allow=ulaw

And the configuration for the Osaka box:

[general]
register => osaka:welcome@192.168.2.202/toronto

[toronto]
type=friend
secret=welcome
context=toronto_incoming
host=dynamic
disallow=all
allow=ulaw

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.

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:

[Apr 22 18:52:32] WARNING[23631]: chan_sip.c:8117 check_auth: username mismatch, 
                                  have <toronto>, digest has <s>

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.

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 [osaka] authorization block, and on the Osaka box, we have the [toronto] block. We define the type as a friend, which allows us to both receive and place calls from the other Asterisk box. The secret is the password the other system should use when authenticating. The context is where incoming calls are processed in the dialplan (extensions.conf). We set the host parameter to dynamic, 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 disallow and allow parameters control the codecs we wish to use with the other end.

If you save the file and reload the SIP channel on both Asterisk boxes (sip reload from the Asterisk console), you should see something like the following, which will tell you the remote box successfully registered:

*CLI>     -- Saved useragent "Asterisk PBX" for peer toronto

You should see the status of the Host change from (Unspecified) to the IP address of the remote box when you run sip show peers:

*CLI> sip show peers
Name/username        Host            Dyn Nat ACL Port     Status               
toronto/osaka        192.168.2.202    D          5060     Unmonitored

You can verify that your own registration was successful by running sip show registry from the Asterisk console:

*CLI> sip show registry
Host                   Username    Refresh State      Reg.Time                 
192.168.1.101:5060     osaka       105 Registered     Sun, 22 Apr 2007 19:13:20

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.

SIP Phone Configuration

See the Configuring an FXS Channel for an Analog Telephone” section of this chapter for more information about configuring SIP phones with Asterisk. Below is the configuration for two SIP phones in the sip.conf 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 sip.conf file on each respective server.

Toronto sip.conf:

[2000]
type=friend
host=dynamic
context=phones

Osaka sip.conf:

[1000]
type=friend
host=dynamic
context=phones

You should now have extension 1000 registered to Toronto, and extension 1001 registered to Osaka. You can verify this with the sip show peers command from the Asterisk console. Next, we’re going to configure the dialplan logic that will allow us to call between the extensions.

Configuring the Dialplan

Now we can configure a simple dialplan for each server allowing us to call between the two phones we have registered: one to Toronto, the other to Osaka. In the Working with Interface Configuration Files” section of this chapter, we asked you to create a simple extensions.conf 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.

Toronto extensions.conf:

[globals]

[general]
autofallthrough=yes

[default]

[incoming_calls]

[phones]
include => internal
include => remote

[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

Osaka extensions.conf:

[globals]

[general]
autofallthrough=yes

[default]

[incoming_calls]

[phones]
include => internal
include => remote

[internal]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

Once you’ve configured your extensions.conf file, you can reload it from the Asterisk console with the dialplan reload command. Verify your dialplan loaded with the dialplan show command.

And that’s it! You should be able to place calls between your two Asterisk servers now.