Connecting Two Asterisk Boxes Together via SIP

There may come a time when you have a pair of Asterisk boxes, and youd like to pass calls between them. Luckily this isnt very difficult, although it does have some oddities that we need to deal with, but from the configuration viewpoint it isnt really all that difficult. Configuring a Local Firewall If youre 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, SIP trunking topology). 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, lets configure our Asterisk boxes.

Configuring Our Asterisk Boxes

We have a pair of Asterisk boxes that were going to call Toronto and Osaka and that were going to have register to each other. Were 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, its not necessarily the best way to do it, but itll 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 lets 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, youll 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 toronto/osaka 192.168.2.202 Dyn Nat ACL Port D 5060 Status Unmonitored

You can verify that your own registration was successful by running sip showregistry from the Asterisk console:
*CLI> sip show registry Host Username 192.168.1.101:5060 osaka 19:13:20 Refresh State 105 Registered Reg.Time Sun, 22 Apr 2007

Now that our Asterisk boxes are happy with each other, lets configure a couple of SIP phones so we can call between the boxes.

SIP Phone Configuration

See the the section called 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 well 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:
[1000] type=friend host=dynamic context=phones

Osaka sip.conf:
[1001] 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, were 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 the section called 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 were 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 youve 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 thats it! You should be able to place calls between your two Asterisk servers now.