Page 127 - Asterisk™: The Future of Telephony

Matching on Username Instead of IP Address
Some service providers may insteadSession Initiation Protocol be sending their calls to
you via multiple IP addresses, requiring you to create a separate peer account for each
IP address. If you don’t know each of these IP addresses, you may need to match on
the username instead. The format for the service provider definition needs to only
change slightly, but the biggest change to note is that you will need to set the [serv-
ice_provider_header] as the username your service provider is going to send the call
to. We have also changed the type from a peer to a friend, which from the viewpoint
of Asterisk creates both a type user and type peer, where the type user will be matched
before the peer:
[my_unique_id]
type=friend
host=10.251.55.100
fromuser=my_unique_id
secret=my_special_secret
context=incoming_calls
dtmfmode=rfc2833
disallow=all
allow=gsm
allow=ulaw
insecure=invite
Note that we’ve removed the deny and permit parameters since we may not know the
IP addresses the calls will be coming from. If you do happen to know them and still
wish to match them, you can add back in the deny and permit(s) for the IP addresses.

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 param-
eter, 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

Connecting to a SIP Service Provider | 99