Kamailio作为多个星号服务器的负载均衡器



我有三个运行Ubuntu 14.04的虚拟服务器。我在一台服务器上安装了Kamailio,在另一台服务器上安装了Asterisk。我希望Kamailio服务器作为负载均衡器工作,并将传入呼叫转发给星号服务器(轮询)。

我想先用一个星号服务器来测试它,如果可以的话,我可以添加更多的星号服务器来提高性能。

我像这样添加了我的SIP提供者凭据:

kamctl add test testpasswd

然后像这样将星号服务器添加到dispatcher表中:

INSERT INTO dispatcher (setid,destination,flags,priority,attrs,description) VALUES (1,"sip:10.1.1.3:5060",0,0,"","Asteriskl-I");

我更改了星号服务器上的sip.conf文件,它连接到我的kamailio服务器,这似乎工作。

我的kamailio.cfg文件看起来像这样:

#!KAMAILIO
#
# sample config file for dispatcher module
# - load balancing of VoIP calls with round robin
# - no TPC listening
# - don't dispatch REGISTER and presence requests
#
# Kamailio (OpenSER) SIP Server v3.2
#     - web: http://www.kamailio.org
#     - git: http://sip-router.org
#
# Direct your questions about this file to: sr-users@lists.sip-router.org
#
# Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php
# for an explanation of possible statements, functions and parameters.
#
# Several features can be enabled using '#!define WITH_FEATURE' directives:
#
# *** To run in debug mode:
#     - define WITH_DEBUG
#
####### Global Parameters #########
#!define WITH_DEBUG
#!ifdef WITH_DEBUG
debug=4
log_stderror=yes
#!else
debug=2
log_stderror=no
#!endif
memdbg=5
memlog=5
log_facility=LOG_LOCAL0
fork=yes
children=4
/* comment the next line to enable TCP */
disable_tcp=yes
/* uncomment the next line to disable the auto discovery of local aliases
   based on revers DNS on IPs (default on) */
auto_aliases=no
/* add local domain aliases */
# alias="mysipserver.com"
port=5060
/* uncomment and configure the following line if you want Kamailio to
   bind on a specific interface/port/proto (default bind on all available) */
# listen=udp:127.0.0.1:5060
sip_warning=no
####### Modules Section ########
#set module path
mpath="/usr/local/lib64/kamailio/modules/"
# loadmodule "db_mysql.so"
loadmodule "mi_fifo.so"
loadmodule "kex.so"
loadmodule "tm.so"
loadmodule "tmx.so"
loadmodule "sl.so"
loadmodule "rr.so"
loadmodule "pv.so"
loadmodule "maxfwd.so"
loadmodule "textops.so"
loadmodule "siputils.so"
loadmodule "xlog.so"
loadmodule "sanity.so"
# loadmodule "ctl.so"
loadmodule "mi_rpc.so"
loadmodule "acc.so"
loadmodule "dispatcher.so"
# ----------------- setting module-specific parameters ---------------
# ----- mi_fifo params -----
modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo")
# ----- rr params -----
# add value to ;lr param to cope with most of the UAs
modparam("rr", "enable_full_lr", 1)
# do not append from tag to the RR (no need for this script)
modparam("rr", "append_fromtag", 0)
# ----- acc params -----
modparam("acc", "log_flag", 1)
modparam("acc", "failed_transaction_flag", 3)
modparam("acc", "log_extra",
        "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;src_ip=$si")
# ----- tm params -----
modparam("tm", "fr_timer", 2000)
modparam("tm", "fr_inv_timer", 40000)
# ----- dispatcher params -----
# modparam("dispatcher", "db_url",
#       "mysql://kamailio:123456@localhost/kamailio")
modparam("dispatcher", "table_name", "dispatcher")
modparam("dispatcher", "flags", 2)
modparam("dispatcher", "dst_avp", "$avp(AVP_DST)")
modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)")
modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)")
####### Routing Logic ########
# main request routing logic
route {
        # per request initial checks
        route(REQINIT);
        # handle requests within SIP dialogs
        route(WITHINDLG);
        ### only initial requests (no To tag)
        # CANCEL processing
        if (is_method("CANCEL"))
        {
                if (t_check_trans())
                        t_relay();
                exit;
        }
        t_check_trans();
        # record routing for dialog forming requests (in case they are routed)
        # - remove preloaded route headers
        remove_hf("Route");
        if (is_method("INVITE|SUBSCRIBE"))
                record_route();
        #if (is_method("INVITE"))
        #{
        #       ds_select_domain("1","4");
        #       #sl_send_reply("300","Redirect");
        #       #t_relay();
        #       exit;
        #}
        # account only INVITEs
        if (is_method("INVITE"))
        {
                setflag(1); # do accounting
        }
        # handle presence related requests
        route(PRESENCE);
        # handle registrations
        route(REGISTRAR);
        if ($rU==$null)
        {
                # request with no Username in RURI
                sl_send_reply("484","Address Incomplete");
                exit;
        }
        # dispatch destinations
        route(DISPATCH);
        route(RELAY);
}
route[RELAY] {
        if (!t_relay()) {
                sl_reply_error();
        }
        exit;
}
# Per SIP request initial checks
route[REQINIT] {
        if (!mf_process_maxfwd_header("10")) {
                sl_send_reply("483","Too Many Hops");
                exit;
        }
        if(!sanity_check("1511", "7"))
        {
                xlog("Malformed SIP message from $si:$spn");
                exit;
        }
}
# Handle requests within SIP dialogs
route[WITHINDLG] {
        if (has_totag()) {
                # sequential request withing a dialog should
                # take the path determined by record-routing
                if (loose_route()) {
                        if (is_method("BYE")) {
                                setflag(1); # do accounting ...
                                setflag(3); # ... even if the transaction fails
                        }
                        route(RELAY);
                } else {
                        if (is_method("SUBSCRIBE") && uri == myself) {
                                # in-dialog subscribe requests
                                route(PRESENCE);
                                exit;
                        }
                        if ( is_method("ACK") ) {
                                if ( t_check_trans() ) {
                                        # non loose-route, but stateful ACK;
                                        # must be ACK after a 487 or e.g. 404 from upstream server
                                        t_relay();
                                        exit;
                                } else {
                                        # ACK without matching transaction ... ignore and discard.
                                        exit;
                                }
                        }
                        sl_send_reply("404","Not here");
                }
                exit;
        }
}
# Handle SIP registrations
route[REGISTRAR] {
        if(!is_method("REGISTER"))
                return;
        #sl_send_reply("404", "No registrar");
        #t_relay();
        if(!ds_select_dst("1", "4"))
        {
                sl_send_reply("404", "No registrar");
                exit;
        }
        forward();
        exit;
}
# Presence server route
route[PRESENCE] {
        if(!is_method("PUBLISH|SUBSCRIBE"))
                return;
        sl_send_reply("404", "Not here");
        exit;
}
# Dispatch requests
route[DISPATCH] {
        # round robin dispatching on gateways group '1'
        if(!ds_select_dst("1", "4"))
        {
                send_reply("404", "No destination");
                exit;
        }
        xlog("L_DBG", "--- SCRIPT: going to <$ru> via <$du>n");
        t_on_failure("RTF_DISPATCH");
        return;
}
# Sample failure route
failure_route[RTF_DISPATCH] {
        if (t_is_canceled()) {
                exit;
        }
        # next DST - only for 500 or local timeout
        if (t_check_status("500")
                        or (t_branch_timeout() and !t_branch_replied()))
        {
                if(ds_next_dst())
                {
                        t_on_failure("RTF_DISPATCH");
                        route(RELAY);
                        exit;
                }
        }
}

如果我将星号框直接连接到我的SIP提供程序,它可以完美地工作。但如果我将它连接到kamailio服务器,而kamailio服务器连接到SIP提供商,它不会。

我在谷歌上搜索了几个小时,尝试了很多东西,我真的不知道接下来我可以尝试什么…如果有人能帮助我,我将非常高兴!

非常感谢,并致以最诚挚的问候

I added my SIP provider credentials like this:

kamctl add test testpasswd -这是错误的。

查看以下链接,了解如何在使用用户名/密码身份验证的Kamailio上设置SIP中继的详细信息:

http://lists.sip-router.org/pipermail/sr-users/2015-September/090001.html

相关内容

  • 没有找到相关文章

最新更新