With the advent of Internet telephony, there has been an influx of Internet-based phone companies springing up all over the world! This gives you a large number of choices from which to choose. Many of these service providers allow you to connect your Asterisk-based system to their networks,[66] and some of them are even running Asterisk themselves!
The following configuration should get you connected with an Internet Telephony Service Provider (ITSP),[67] although it is impossible to know the unique configurations each service provider will require from you, and ideally the provider will give you the configuration required to connect your system with its own. However, not all are going to support Asterisk, so we’re going to provide you with a generic configuration which should help you get on your way and, ideally, going in a matter of minutes:
[my_service_provider] type=peer host=10.251.55.100 fromuser=my_unique_idsecret=my_special_secretcontext=incoming_calls dtmfmode=rfc2833 disallow=all allow=gsm allow=ulaw deny=0.0.0.0/0 permit=10.251.55.100/32 insecure=invite
Most of the previous configuration may be familiar to you by now, but in case it’s not, here is a brief rundown.
By defining the type as a peer, we are telling Asterisk not to match
on the [my_service_provider] name, but rather to match
on the IP address in the INVITE message (when the provider is sending us a
call). The host parameter is the IP address that we’ll place our calls to,
and the IP address we’ll be matching on when receiving a call from the
provider.
The fromuser parameter is going
to affect the way our INVITE message is structured when sending the call
to the provider. By setting our username in the fromuser parameter, we will modify the From: and Contact: fields of the INVITE
when sending a call to the provider. This may be required by the provider
if it’s using these fields as part of its authentication routine. You can
see the places Asterisk modifies the header in the next two code
blocks.
Without the fromuser:
Audio is at 66.135.99.122 port 18154 Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.251.55.100:5060: INVITE sip:15195915119@10.251.55.100 SIP/2.0 Via: SIP/2.0/UDP 66.135.99.122:5060;branch=z9hG4bK32469d35;rport From: "asterisk" <sip:asterisk@66.135.99.122>;tag=as4975f3ff To: <sip:15195915119@10.251.55.100> Contact: <sip:asterisk@66.135.99.122> Call-ID: 58e3dfb2584930cd77fe989c00986584@66.135.99.122 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 20 Apr 2007 14:59:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 265
With the fromuser:
Audio is at 66.135.99.122 port 11700 Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.251.55.100:5060: INVITE sip:15195915119@10.251.55.100 SIP/2.0 Via: SIP/2.0/UDP 66.135.99.122:5060;branch=z9hG4bK635b0b1b;rport From: "asterisk" <sip:my_unique_id@66.135.99.122>;tag=as3186c1ba To: <sip:15195915119@10.251.55.100> Contact: <sip:my_unique_id@66.135.99.122> Call-ID: 0c7ad6156f92e70b1fecde903550a12f@66.135.99.122 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 20 Apr 2007 15:00:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 265
The deny and permit
statements are used to deny all incoming calls to this peer except the
IP address defined by the permit parameter. This is
simply a security measure used to make sure nothing else matches on this
peer except traffic coming from the IP address we expect.
At the end is insecure=invite, which may be
required for your provider. This is because the source of the INVITE may
originate from its backend platform, but could be directed through its SIP
proxy server. Basically what this means is that the IP address that the
peer is coming from, and which you are matching on, may not be the IP
address that is in the Contact line: field of the INVITE message when you
are accepting a call from your provider. This tells Asterisk to ignore
this inconsistency and to accept the INVITE anyway.
You may need to set invite=invite,port if the port address is
also inconsistent with what Asterisk is expecting.
Now we need one additional parameter set in the [general] section of our sip.conf file: register. register is going to tell the service provider
where to send calls when it has a call to deliver to us. This is
Asterisk’s way of saying to the service provider, “Hey! If you’ve got a
call for me, send it to me at IP address 10.251.55.100.” The
register parameter takes the following form:
register => username:secret@my.service_provider.tld
Now we just need to configure a simple dialplan to handle our incoming calls and to send calls via the service provider. We’re going to modify the simple dialplan we started building in the the section called “Setting Up the Dialplan for Some Test Calls”” section of this chapter. The italicized sections are the new parts that we’re adding to the dialplan, with everything else existing previously.[68]
[globals] [general] [default] exten => s,1,Verbose(1|Unrouted call handler) exten => s,n,Answer() exten => s,n,Wait(1) exten => s,n,Playback(tt-weasels) exten => s,n,Hangup() [incoming_calls] exten => _X.,1.NoOp() exten => _X.,n,Dial(SIP/1000) [outgoing_calls] exten => _X.,1,NoOp() exten => _X.,n,Dial(SIP/my_service_provider/${EXTEN}) [internal] exten => 1000,1,Verbose(1|Extension 1000) exten => 1000,n,Dial(SIP/1000,30) exten => 1000,n,Hangup() exten => 500,1,Verbose(1|Echo test application) exten => 500,n,Echo() exten => 500,n,Hangup() [phones] include => internal include => outgoing_calls
[66] Be sure to check the policy of any service provider you are looking to connect with, as some of them may not allow you to use a PBX system with its service.
[67] Also known as a VoIP Service Provider (VSP).
[68] We also assume you have configured at least one SIP extension from the previous section.