Kamailio的dispatcher模块我用的很多,很熟
OpenSIPS有同名模块
二者基本功能相同
但真使起来,差别还是挺大的
# OpenSIPS
event_route[E_DISPATCHER_STATUS] {
# log each time a dispatcher destination
# changes its status
xlog("Dispatcher destination has new status");
}
OpenSIPS需要在路由里面修改destination的状态,但Kamailio则不用这么麻烦,模块自己就搞定了
以后抽时间仔细测试下OpenSIPS的这个模块,再完善这篇文章
查了查源码,主要逻辑如下:
evi_param_add_int(list, &group_str, &set->id);
evi_param_add_str(list, &address_str, address);
evi_param_add_str(list, &status_str,
type ? &inactive_str : &active_str);req = get_dummy_sip_msg();
status = evi_raise_event_msg(req, id, params); # 发一个dummy sip message
release_dummy_sip_msg(req);
改了下路由块,如下:
event_route[E_DISPATCHER_STATUS] {
# log each time a dispatcher destination
# changes its status$var(status) = $hdr(status);
xlog("Dispatcher destination has new status, ru = $rU, tu = $tU, fu = $fU, status = $var(status)\n");
avp_print();
}
但还是不行, $rU, $tU, $fU等可以正常取到,$hdr(status)为NULL,$hdr(group)没试,估计也不行
avp_print() 也不行,没有打印出有用的内容
看来OpenSIPS跟Kamailio有很大不同,没那么容易掌握
可能要求助Mailing List了
今天查了下文档,貌似在路由参数里面
https://www.opensips.org/Documentation/Script-CoreVar-3-2
$param(1), $param(2) 等等
OpenSIPS跟Kamailio虽然同源,但细节方面差异巨大
刚才做了测试,路由调整为:
event_route[E_DISPATCHER_STATUS] {
xlog("Dispatcher destination has new status, $param(1), $param(2), $param(3), $param(4)\n");
}
日志为:
Dispatcher destination has new status, 1, sip:172.20.10.6:25080, active, 200 OK probing reply
Dispatcher destination has new status, 1, sip:172.20.10.6:25080, inactive, negative probing reply
跟文档能对应起来了
opensips.cfg内容如下:
####### Global Parameters #########
/* uncomment the following lines to enable debugging */
#debug_mode=yes
log_level=3
xlog_level=3
stderror_enabled=yes
syslog_enabled=yes
syslog_facility=LOG_LOCAL0
udp_workers=4
/* uncomment the next line to enable the auto temporary blacklisting of
not available destinations (default disabled) */
#disable_dns_blacklist=no
/* uncomment the next line to enable IPv6 lookup after IPv4 dns
lookup failures (default disabled) */
#dns_try_ipv6=yes
socket=udp:127.0.0.1:5060 # CUSTOMIZE ME
socket=udp:172.20.10.6:5060 # CUSTOMIZE ME
####### Modules Section ########
#set module path
mpath="/usr/lib/x86_64-linux-gnu/opensips/modules/"
loadmodule "db_mysql.so"
#### SIGNALING module
loadmodule "signaling.so"
#### StateLess module
loadmodule "sl.so"
#### Transaction Module
loadmodule "tm.so"
modparam("tm", "fr_timeout", 5)
modparam("tm", "fr_inv_timeout", 30)
modparam("tm", "restart_fr_on_each_reply", 0)
modparam("tm", "onreply_avp_mode", 1)
#### Record Route Module
loadmodule "rr.so"
/* do not append from tag to the RR (no need for this script) */
modparam("rr", "append_fromtag", 0)
#### MAX ForWarD module
loadmodule "maxfwd.so"
#### SIP MSG OPerationS module
loadmodule "sipmsgops.so"
#### FIFO Management Interface
loadmodule "mi_fifo.so"
modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")
modparam("mi_fifo", "fifo_mode", 0666)
#### USeR LOCation module
loadmodule "usrloc.so"
modparam("usrloc", "nat_bflag", "NAT")
modparam("usrloc", "working_mode_preset", "single-instance-no-db")
#### REGISTRAR module
loadmodule "registrar.so"
modparam("registrar", "tcp_persistent_flag", "TCP_PERSISTENT")
/* uncomment the next line not to allow more than 10 contacts per AOR */
#modparam("registrar", "max_contacts", 10)
loadmodule "textops.so"
#### ACCounting module
loadmodule "acc.so"
/* what special events should be accounted ? */
modparam("acc", "early_media", 0)
modparam("acc", "report_cancels", 0)
/* by default we do not adjust the direct of the sequential requests.
if you enable this parameter, be sure to enable "append_fromtag"
in "rr" module */
modparam("acc", "detect_direction", 0)
loadmodule "dispatcher.so"
modparam("dispatcher", "db_url", "mysql://opensips:opensipsrw@localhost/opensips")
modparam("dispatcher", "ds_ping_interval", 10)
modparam("dispatcher", "ds_probing_mode", 1)
modparam("dispatcher", "dst_avp", "$avp(271)")
loadmodule "avpops.so"
loadmodule "event_route.so"
loadmodule "event_flatstore.so"
modparam("event_flatstore", "max_open_sockets", 500)
loadmodule "proto_udp.so"
####### Routing Logic ########
# main request routing logic
route{
if (!mf_process_maxfwd_header(10)) {
send_reply(483,"Too Many Hops");
exit;
}
if (has_totag()) {
# handle hop-by-hop ACK (no routing required)
if ( is_method("ACK") && t_check_trans() ) {
t_relay();
exit;
}
# sequential request within a dialog should
# take the path determined by record-routing
if ( !loose_route() ) {
# we do record-routing for all our traffic, so we should not
# receive any sequential requests without Route hdr.
send_reply(404,"Not here");
exit;
}
if (is_method("BYE")) {
# do accounting even if the transaction fails
do_accounting("log","failed");
}
# route it out to whatever destination was set by loose_route()
# in $du (destination URI).
route(relay);
exit;
}
# CANCEL processing
if (is_method("CANCEL")) {
if (t_check_trans())
t_relay();
exit;
}
# absorb retransmissions, but do not create transaction
t_check_trans();
if ( !(is_method("REGISTER") ) ) {
if (is_myself("$fd")) {
} else {
# if caller is not local, then called number must be local
if (!is_myself("$rd")) {
send_reply(403,"Relay Forbidden");
exit;
}
}
}
# preloaded route checking
if (loose_route()) {
xlog("L_ERR",
"Attempt to route with preloaded Route's [$fu/$tu/$ru/$ci]");
if (!is_method("ACK"))
send_reply(403,"Preload Route denied");
exit;
}
# record routing
if (!is_method("REGISTER|MESSAGE"))
record_route();
# account only INVITEs
if (is_method("INVITE")) {
do_accounting("log");
}
if (!is_myself("$rd")) {
append_hf("P-hint: outbound\r\n");
route(relay);
}
# requests for my domain
if (is_method("PUBLISH|SUBSCRIBE")) {
send_reply(503, "Service Unavailable");
exit;
}
if (is_method("REGISTER")) {
# store the registration and generate a SIP reply
if (!save("location"))
xlog("failed to register AoR $tu\n");
exit;
}
if ($rU==NULL) {
# request with no Username in RURI
send_reply(484,"Address Incomplete");
exit;
}
if (is_method("INVITE")) {
replace_body("a=sendrecv*\r\n","");
}
# do lookup with method filtering
if (!lookup("location","method-filtering")) {
t_reply(404, "Not Found");
exit;
}
# when routing via usrloc, log the missed calls also
do_accounting("log","missed");
route(relay);
}
route[relay] {
# for INVITEs enable some additional helper routes
if (is_method("INVITE")) {
t_on_branch("per_branch_ops");
t_on_reply("handle_nat");
t_on_failure("missed_call");
}
if (!t_relay()) {
send_reply(500,"Internal Error");
}
exit;
}
branch_route[per_branch_ops] {
xlog("new branch at $ru\n");
}
onreply_route[handle_nat] {
xlog("incoming reply\n");
}
failure_route[missed_call] {
if (t_was_cancelled()) {
exit;
}
# uncomment the following lines if you want to block client
# redirect based on 3xx replies.
##if (t_check_status("3[0-9][0-9]")) {
##t_reply(404,"Not found");
## exit;
##}
}
event_route[E_DISPATCHER_STATUS] {
xlog("Dispatcher destination has new status, $param(1), $param(2), $param(3), $param(4)\n");
}
startup_route {
subscribe_event("E_DISPATCHER_STATUS", "flatstore:/var/log/dispatcher.log");
}
套路就是用osipconfig产生路由脚本,然后查官方手册,加上自己想要的逻辑