Let’s imagine that we’ve upgraded our VoIP network which was formerly based on geographically distributed Asterisks.
We configured OpenSIPS servers as registrars, connected them together in a full-sharing usrloc cluster and now we need to route calls between endpoints not directly, but through Asterisks – to handle our calls in a familiar way (CDR records, call recording via MixMonitor, some AGI scripts, etc).
We have to create something like that:
This is a code snippet of OpenSIPS with IP address 10.145.213.63:
# initial INVITE
if(is_method("INVITE") && !has_totag())
{
t_on_failure("1");
# call from registered user ->
# forward to mediaserver for call recording, etc.
if(is_registered("location"))
{
$ru="sip:" + $oU + "@" + "10.145.213.63:5067";
route(relay);
}
# call from Asterisk? -> change domain part before doing lookup
if($sp=="5067")
{
$rd="taxsee.com";
}
$var(lookup_flags) = "m";
if(cluster_check_addr("1", "$si")) {
xlog("si: $si . $rm from cluster, doing local lookup only\n");
} else {
xlog("si: $si . $rm from outside, doing global lookup\n");
$var(lookup_flags) = $var(lookup_flags) + "g";
}
if(!lookup("location", "$var(lookup_flags)"))
{
t_reply("404", "Not Found");
exit;
}
if(has_body("application/sdp"))
{
rtpengine_offer("RTP/AVP replace-origin replace-session-connection ICE=remove");
}
} # initial INVITE end
route(relay);
A SIP peer to this OpenSIPS in Asterisk sip.conf looks like this:
[opensips]
type=peer
context=office
host=10.145.213.63
port=5060And a dialplan for CDR/MixMonitor/etc:
context office
{
_XXX =>
{
NoOp(imagine this is CDR, MixMonitor, AGI);
Dial(SIP/opensips/${EXTEN});
Hangup();
};
};UPD: assuming your SIP acoounts DB has moved from Asterisk to OpenSIPS cluster, its desirable not just processing REGISTER requests and doing authentication, but also adding caller ids to your SIP accounts.
Tags: opensips