New subject: [Serusers] error: mediaproxy/sendMediaproxyCommand(): can't connect to MediaProxy

20 Feb 2005


      Thank you very much. And my log file at /var/log/message lists below:
Charles
Log Start:
-------------------------------------------------------------------------------------------------------------------
Maxfwd module- initializing
AVPops - initializing
permissions - initializing
Default allow file (/usr/local/etc/ser/permissions.allow) not found =>
empty rule set
Default deny file (/usr/local/etc/ser/permissions.deny) not found =>
empty rule set
info: mediaproxy: loaded SIP asymmetric clients file containing 1 entry.
info: mediaproxy: loaded RTP asymmetric clients file containing 0 entries.
INFO: udp_init: SO_RCVBUF is initially 65535
INFO: udp_init: SO_RCVBUF is finally 262142
INFO: fifo process starting: 3947
SER: open_uac_fifo: fifo server up at /tmp/ser_fifo...
PDT:prefix2domain: no prefix found in [1011]
Time:[Mon Feb 21 11:12:49 2005] Method:<INVITE> r-uri:1011@ser.xxx.net.tw
IP:<61.229.13.49> From:sip:1033@ser.xxx.net.tw
To:sip:1011@ser.xxx.net.tw sip:1033@10.18.1.102:1718
SER: BLIND CALL FORWARDING
SER: Look aliases
SER: Look location
SER isflagset (sip)
SER: Look aliases
SER: Look location
SER isflagset (sip)
SER: SIP Call On-Net section route(2)
PDT:prefix2domain: no prefix found in [1011]
Time:[Mon Feb 21 11:12:50 2005] Method:<INVITE> r-uri:1011@ser.xxx.net.tw
IP:<61.229.13.49> From:sip:1033@ser.xxx.net.tw
To:sip:1011@ser.xxx.net.tw sip:1033@10.18.1.102:1718
SER: BLIND CALL FORWARDING
SER: Look aliases
SER: Look location
SER isflagset (sip)
SER: Look aliases
SER: Look location
SER isflagset (sip)
SER: SIP Call On-Net section route(2)
error: mediaproxy/sendMediaproxyCommand(): can't connect to MediaProxy
error: mediaproxy/sendMediaproxyCommand(): can't connect to MediaProxy
SER: SIP Call On-Net section route(2)
error: mediaproxy/sendMediaproxyCommand(): can't connect to MediaProxy
SER: SIP Call On-Net section route(2)
error: mediaproxy/sendMediaproxyCommand(): can't connect to MediaProxy
SER: SIP Call On-Net section route(2)
error: mediaproxy/sendMediaproxyCommand(): can't connect to MediaProxy
SER: SIP Call On-Net section route(2)
error: mediaproxy/sendMediaproxyCommand(): can't connect to MediaProxy
SER: SIP Call On-Net section route(2)
----------------------------------------------------------------------------------------------------------------
On Sun, 20 Feb 2005 21:58:12 -0500, Java Rockx javarockx@gmail.com wrote:
...
Can you include an output of the error messages?
Regards,
Paul
On Mon, 21 Feb 2005 10:12:23 +0800, Charles Wang lazy.charles@gmail.com wrote:
...
Dear ALL:
I start ser 0.9.0. And when I make a sip call or off-net call(to
PSTN), some error(as Subject) about MediaProxy will display in Log
file.
Does anybody have the same problem and have any solution to fix it?
My ser.cfg list below:
debug=10         # debug level (cmd line: -dddddddddd)
fork=yes
log_stderror=no # (cmd line: -E)
listen=xxx.xxx.xxx.xxx
alias=ser.xxx.net.tw
alias=ser
alias=xxx.xxx.xxx.xxx
check_via=no            # (cmd. line: -v)
dns=no                                  # (cmd. line: -r)
rev_dns=no                      # (cmd. line: -R)
port=5060
children=4
fifo_mode=0666
fifo="/tmp/ser_fifo"
fifo_db_url="mysql://ser:heslo@localhost/ser"
# ------------------ module loading ----------------------------------
# ----------------- setting module-specific parameters ---------------
----------(skip)
# -- mediaproxy params --
modparam("mediaproxy", "natping_interval", 30)
modparam("mediaproxy", "sip_asymmetrics",
"/usr/local/etc/ser/sip-asymmetric-clients")
modparam("mediaproxy", "rtp_asymmetrics",
"/usr/local/etc/ser/rtp-asymmetric-clients")
# -- usrloc params --
modparam("usrloc", "db_mode", 1)
modparam("usrloc", "timer_interval", 60)
modparam("usrloc", "desc_time_order", 1)
# -- auth params --
modparam("auth_db", "calculate_ha1", yes)
modparam("auth_db", "password_column", "password")
# -- rr params --
# add value to ;lr param to make some broken UAs happy
modparam("rr", "enable_full_lr", 1)
# -- db_url params --
modparam("acc|auth_db|domain|group|permissions|speeddial|uri_db|usrloc|pdt",
"db_url", "mysql://ser:heslo@localhost/ser")
# -- use_domain params --
modparam("auth_db|group|registrar|speeddial|uri_db|usrloc", "use_domain", 0)
# -- permissions params --
modparam("permissions", "db_mode", 1)
modparam("permissions", "trusted_table", "trusted")
# -- accounting params --
modparam("acc", "db_flag", 1)
modparam("acc", "db_missed_flag", 1)
modparam("acc", "log_fmt", "cdfimorstup")
modparam("acc", "log_level", 1)
modparam("acc", "failed_transactions", 1)
modparam("acc", "report_cancels", 1)
modparam("acc", "report_ack", 0)
# -- domain params --
modparam("domain", "db_mode", 1)
# ------------- exec parameters
modparam("exec", "setvars", 1)
modparam("exec", "time_to_kill", 10)
# -- registration params --
modparam("registrar", "nat_flag", 2)
modparam("registrar", "min_expires", 60)
modparam("registrar", "max_expires", 86400)
modparam("registrar", "default_expires", 3600)
modparam("registrar", "append_branches", 1)
# -- avp params --
modparam("avpops", "avp_url", "mysql://ser:heslo@localhost/ser")
modparam("avpops", "avp_table", "usr_preferences")
#modparam("avpops", "use_domain", "1")
modparam("avpops", "uuid_column", "uuid")
modparam("avpops", "username_column", "username")
modparam("avpops", "domain_column", "domain")
modparam("avpops", "attribute_column", "attribute")
modparam("avpops", "value_column", "value")
modparam("avpops", "type_column", "type")
modparam("avpops", "avp_aliases",
"voicemail=i:500;calltype=i:700;fwd_no_answer_type=i:701;fwd_busy_type=i:702")
# To use more than one tables example
#modparam("avpops", "db_scheme",
"scheme1:table=subscriber;uuid_column=uuid;value_column=first_name")
# -- tm params --
modparam("tm", "fr_timer", 15)
modparam("tm", "fr_inv_timer", 22)
modparam("tm", "wt_timer", 5)
modparam("tm", "fr_inv_timer_avp", "inv_timeout")
# -- pdt params --
modparam("pdt", "db_table", "prefix_domain")
modparam("pdt", "prefix", "")
modparam("pdt", "hsize_2pow", 2)
modparam("pdt", "sync_time", 300)
modparam("pdt", "clean_time", 600)
# -- logging params
modparam("xlog", "buf_size", 8192)
# -- group params --
modparam("group", "table", "grp")
modparam("group", "user_column", "username")
modparam("group", "domain_column", "domain")
modparam("group", "group_column", "grp")
# -------------------------  request routing logic -------------------
# main routing logic
route {
    # ------------------------------------------------------------------------
    # Sanity Check Section
    # ------------------------------------------------------------------------
    if (!mf_process_maxfwd_header("10")) {
            sl_send_reply("483", "Too Many Hops");
            break;
    };

    if (msg:len > max_len) {
            sl_send_reply("513", "Message Overflow");
            break;
    };

    # ------------------------------------------------------------------------
    # NOTIFY Keep-Alive Section
    # ------------------------------------------------------------------------
    if ((method=="NOTIFY") && search("^Event: keep-alive")) {
            sl_send_reply("200", "OK");
            break;
    };

    # ------------------------------------------------------------------------
    # OPTIONS Section
    # ------------------------------------------------------------------------
    if (method=="OPTIONS") {
            options_reply();
            break;
    };

    # ------------------------------------------------------------------------
    # NAT Test Section #1
    # ------------------------------------------------------------------------
    if (method=="REGISTER" && client_nat_test("3")) {
            fix_contact();
            force_rport();
            setflag(2);
    };

    # ------------------------------------------------------------------------
    # Registration Section
    # ------------------------------------------------------------------------
    if (method=="REGISTER") {

            # allow all requests from user 700 - the Click2Dial controller
            if (!isflagset(14)) {
                    if (!is_from_local()) {
                            sl_send_reply("403", "Unknown Domain");
                            break;
                    };
                    if (!www_authorize("", "subscriber")) {
                            www_challenge("", "0");
                            break;
                    };
                    if (!check_to()) {
                            sl_send_reply("401", "Unauthorized");
                            break;
                    };

                    # To - Use To username and (optionally) domain to check
                    if (is_user_in("To", "demo-disabled")) {
                            sl_send_reply("403", "Your evaluation period has expired");
                            break;
                    };

                    # To - Use To username and (optionally) domain to check
                    if (is_user_in("To", "disabled")) {
                            sl_send_reply("403", "Your account has been disabled");
                            break;
                    };
            };

            # snom sip phones use this header to start their
            # keep-alive mechanism for NAT bindings
            append_to_reply("P-NAT-Refresh: 15\r\n");

            if (!save("location")) {
                    sl_reply_error();
            };
            break;
    };

    # ------------------------------------------------------------------------
    # Accounting Section # 1
    #
    # NOTE: We test for flag 14 because we do not want to record Click2Dial
    #       entries
    # ------------------------------------------------------------------------
    if ((method=="INVITE" || method=="BYE") && !isflagset(14)) {
            setflag(1);
    };

    # ------------------------------------------------------------------------
    # NAT Tear-Down Section
    # ------------------------------------------------------------------------
    if ((method == "BYE" || method == "CANCEL")) {
            end_media_session();
    };

    # ------------------------------------------------------------------------
    # Record Route Section
    #
    # we record-route all messages -- to make sure that subsequent messages
    # will go through our proxy; that's particularly good if upstream and
    # downstream entities use different transport protocol
    # ------------------------------------------------------------------------
    if (!method=="REGISTER") {
            record_route();
    };

    # ------------------------------------------------------------------------
    # Loose Route Section
    #
    # Grant route routing if route headers present
    # ------------------------------------------------------------------------
    if (loose_route()) {
            route(2);
            break;
    };

    prefix2domain();

    # ------------------------------------------------------------------------
    # NAT Test Section #1
    # ------------------------------------------------------------------------
    if (client_nat_test("3") && !search("^Record-Route:")) {
            force_rport();
            fix_contact();
    };

    # ------------------------------------------------------------------------
    # PSTN Section
    # ------------------------------------------------------------------------
    if (method=="INVITE") {  ## Deny PSTN to 0204 and 095 on Tawan
            if ((uri=~"^sip:0204[0-9]*@") || (uri=~"^sip:095[0-9]*@")) {
                    sl_send_reply("503", "Service Unavailable");
                    break;
            };
    };

    # ------------------------------------------------------------------------
    # Alias Routing Section
    # ------------------------------------------------------------------------
    lookup("aliases");
    if (!uri==myself) {
            route(2);
            break;
    };

    # ------------------------------------------------------------------------
    # Load ACL Section
    # ------------------------------------------------------------------------
    if (method=="INVITE" && !isflagset(14)) {
            xlog("L_ERR", "Time:[%Tf] Method:<%rm> r-uri:<%ru>\n");
            xlog("L_ERR", "IP:<%is> From:<%fu> To:<%tu> %ct\n");
            if (is_user_in("Request-URI", "voicemail")) {
                    setflag(31);
            };
            if (is_from_local() || is_uri_host_local()) {

                    # Check user from grp table
                    if (is_user_in("From", "int")) {
                      log(1, "SER: a INT user\n");
                            setflag(29);
                    };
                    if (is_user_in("From", "free-pstn")) {
                      log(1, "SER: a FREE-PSTN user\n");
                            setflag(28);
                    };
                    if (avp_db_load("$from/username", "s:callidblock")) {
                            if (avp_check("s:callidblock", "eq/y/i")) {
                                    setflag(25);
                            };
                    };
            };
            if (avp_db_load("$ruri/username", "s:anoncallrej")) {
                    if (avp_check("s:anoncallrej", "eq/y/i")) {
                      log(1, "SER: a ANON-CALL-REJ user\n");
                            setflag(24);
                    };
            };
    };

    # ------------------------------------------------------------------------
    # Anonymous Call Rejection Section # 24
    # ------------------------------------------------------------------------
    if (isflagset(24) && (method=="INVITE") &&

search("^(f|F)rom:.*(a|A)nonymous")) {
                route(8);
                break;
        };
    # ------------------------------------------------------------------------
    # Speed Dialing Section
    # ------------------------------------------------------------------------
    if ((method=="INVITE") && (uri=~"^sip:[0-9]{2}@.*")) {
            sd_lookup("speed_dial");
    };

    # ------------------------------------------------------------------------
    # 002 International Call Section # 29
    # ------------------------------------------------------------------------
    if (method=="INVITE" && uri=~"^sip:002[0-9]*@") {
            if (isflagset(29)) {
                    log(1, "SER: an International Call route(6)\n");
                    route(6);
            } else {
                    sl_send_reply("503", "Service Unavailable");
            };
            break;
    };

    # ------------------------------------------------------------------------
    # 0XXXXXXXXX Domestic Call Section # 29
    # ------------------------------------------------------------------------
    if (method=="INVITE" && uri=~"^sip:0[0-9]{9}@") {
            if (isflagset(29)) {
              log(1, "SER: a Domestic Call route(6)\n");
                    route(6);
            } else {
                    sl_send_reply("503", "Service Unavailable");
            };
            break;
    };

    # ------------------------------------------------------------------------
    # URI Compare Section
    #
    # Here we compare the "from" and "to" to see if the caller is dialing
    # their own extension. If so then we route to voicemail(31) if needed
    # ------------------------------------------------------------------------
    if (method=="INVITE") {
            avp_write("$from", "i:34");
            if (avp_check("i:34", "eq/$ruri/i")) {
                    if (isflagset(31)) {
                            route(5);
                            break;
                    } else {
                            sl_send_reply("486", "Busy");
                            break;
                    };
            };
    };

    # ------------------------------------------------------------------------
    # Do Not Disturb Section
    # ------------------------------------------------------------------------
    if (avp_db_load("$ruri/username", "s:donotdisturb")) {
            if (avp_check("s:donotdisturb", "eq/y/i")) {
                    route(5);
                    break;
            };
    };

    # ------------------------------------------------------------------------
    # Blind Call Forwarding Section
    # ------------------------------------------------------------------------
    if (method=="INVITE") {
      log(1, "SER: BLIND CALL FORWARDING\n");
            # here we must store the current (aka original) R-URI because if
            # we set call forwarding and the forwarded number is busy then we
            # need to use this original R-URI to determine which voicemail
            # box we should go to
            if (isflagset(31)) {
                    avp_write("$ruri", "$voicemail");
            };
            if (avp_db_load("$ruri/username", "s:callfwd")) {
                    avp_pushto("$ruri", "s:callfwd");

                    # lookup the call fowarding number to see if it is a served
                    # sip number or a PSTN number

                    # check forwarding number rules
                    log(1, "SER: Check Forwarding Number Rules\n");
                    route(1);

                    if (avp_check("$calltype", "eq/-/i")) {
                            log(1, "SER: 503 Service Unavailable 1\n");
                            sl_send_reply("503", "Service Unavailable");
                            break;
                    };

                    # test for domestic PSTN gateway
                    if (avp_check("$calltype", "eq/dom/i")) {
                      log(1, "SER: Start a domestic PSTN call route(3)\n");
                            route(3);
                            break;
                    };

                    # test for international PSTN gateway
                    if (avp_check("$calltype", "eq/int/i")) {
                            log(1, "SER: Start an international PSTN call route(6)\n");
                            route(6);
                            break;
                    };
            };
    };

    # ------------------------------------------------------------------------
    # Call Routing Section
    # ------------------------------------------------------------------------
    if (!lookup("location")) {

            # if flag 31 (ie voicemail) is set and we made it here this means
            # the user's phone is not registered anywhere. We'll forward to
            # voicemail after this block because we need to check the call
            # forward settings first
            if (isflagset(31)) {
                    # flag 19 means the user has voicemail but is not online
                    # so we need to remember to send to voicemail if call
                    # forwarding is not enabled
                    setflag(19);
            };

            if (method=="INVITE") {
                    if (does_uri_exist()) {
                            # subscriber record found, but they're offline
                            log(1, "SER: Temporarily Unavailable\n");
                            sl_send_reply("480", "Temporarily Unavailable");
                            break;
                    };

                    if (uri=~"^sip:0[0-9]{9}@") {
                            # Send to PSTN Gateway
                            if (isflagset(28) || (is_user_in("From", "int")) ||

(is_user_in("From", "free-pstn"))) {
                                  log(1, "SER: Start a PSTN call\n");
                                        route(3);
                                } else {
                                        log(1, "SER: 503 Service Unavailable 2\n");
                                        sl_send_reply("503", "Service Unavailable");
                                };
                                break;
                        };
                        sl_send_reply("404", "User Not Found");
                        break;
                };
        };
    # ------------------------------------------------------------------------
    # Call Forwarding Section
    # ------------------------------------------------------------------------
    if (method=="INVITE") {

            # save R-URI in a temp AVP for later use
            avp_write("$ruri", "i:99");

            # only load the forward no answer option if voice mail is not enabled
            if (!isflagset(31)) {
                    if (avp_db_load("$ruri/username", "s:fwdnoanswer")) {
                            route(1);
                    };
            };
            if (avp_db_load("$ruri/username", "s:fwdbusy")) {
                    route(1);
            };
            avp_pushto("$ruri", "i:99");
    };
    if (isflagset(19)) {
            # send to voicemail
            route(5);
    } else {
            route(2);
    };

}
route[1] {
    # Here we have route checks for all the call forwarding stuff.
    # The return values are passed as AVP $calltype as follows:
    #
    # "-"   = R-URI is not allowed
    # "dom" = R-RURI is a domestic call
    # "int" = R-RURI is an international call
    # "sip" = R-RURI is a sip call

    avp_write("sip", "$calltype");


# Call rejected
        if ((uri=~"^sip:0204[0-9]*@") ||
            (uri=~"^sip:095[0-9]*@") ||
            (uri=~"^sip:[0-9]{3}@")) {
                break;
        };
  log(1, "SER: Look aliases\n");
        lookup("aliases");
    log(1, "SER: Look location\n");
    if (!lookup("location")) {
            if (uri=~"^sip:0[0-9]{9}@") {
                    # test for domestic PSTN number
                    if (isflagset(28)) {
                            log(1, "SER isflagset 28 (domestic)\n");
                            avp_write("dom", "$calltype");
                    };
            } else if (uri=~"^sip:002[0-9]*@") {
                    # test for international PSTN number
                    if (isflagset(29)) {
                            log(1, "SER isflagset 29 (int)\n");
                            avp_write("int", "$calltype");
                    };
            };
            break;
    };
    log(1, "SER isflagset (sip)\n");
    avp_write("sip", "$calltype");

}
route[2] {
        log(1, "SER: SIP Call On-Net section route(2)\n");
        if ((method=="INVITE") && !allow_trusted()) {
                if (!proxy_authorize("", "subscriber")) {
                        proxy_challenge("", "0");
                        break;
                } else if (!check_from()) {
                        log(1, "Spoofed SIP call attempt");
                        sl_send_reply("403", "Use From=ID");
                        break;
                } else if (!(is_from_local() || is_uri_host_local())) {
                        sl_send_reply("403", "Please register to use our service");
                        break;
                };
        };
        if (uri=~"[@:](192.168.|10.|172.(1[6-9]|2[0-9]|3[0-1]).)" &&
!search("^Route:")){
                sl_send_reply("479", "We don't forward to private IP addresses");
                break;
        };
        if (isflagset(25)) {
                replace("^From:(.*)>" , "From: "Anonymous"
sip:someone@anonymous.invalid");
        };
        if (method=="INVITE" || method=="ACK") {
                use_media_proxy();
        };
        t_on_failure("1");
        t_on_reply("1");
        if (!t_relay()) {
                if (method=="INVITE" || method=="ACK") {
                        end_media_session();
                };
                sl_reply_error();
        };
}
route[3] {
        log(1, "SER: Demestic Call Off-Net section route(3)\n");
    # All Domestic Calls Go To CISCO 5300
    if (method=="INVITE") {
            if (!proxy_authorize("", "subscriber")) {
                    proxy_challenge("", "0");
                    break;
            } else if (!check_from()) {
                    log(1, "Spoofed SIP call attempt");
                    sl_send_reply("403", "Use From=ID");
                    break;
            } else if (!(is_from_local() || is_uri_host_local())) {
                    sl_send_reply("403", "Please register to use our service");
                    break;
            };
            # enable caller id blocking for PSTN calls
            if (isflagset(25)) {
                    append_rpid_hf();
            };
    };
    # SIP->PSTN calls get 45 seconds to timeout
    log(1, "SER: Connecting to PSTN.....\n");
    avp_write("i:45", "inv_timeout");
    rewritehost("61.220.190.243");
    if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" &&

!search("^Route:")){
                sl_send_reply("479", "We don't forward to private IP addresses");
                break;
        };
        if (method=="INVITE" || method=="ACK") {
                use_media_proxy();
        };
        if (isflagset(31)) {
                t_on_failure("1");
        };
        t_on_reply("1");
        if (!t_relay()) {
                if (method=="INVITE" || method=="ACK") {
                        end_media_session();
                };
                sl_reply_error();
        };
}
route[4] {
        log(1, "SER: Voice Mail section route(4)\n");
    # voicemail route #1
    #
    # this path this executed during these conditions:
    #
    # cond 1) the called number is in the location table
    #                     but the callee did not answer the phone
    #                     (ie, failover to voicemail)

    if (isflagset(25)) {
            replace("^From:(.*)>" , "From: \"Anonymous\"

sip:someone@anonymous.invalid");
        };
        rewritehostport("99.99.99.100:5060");
        append_branch();
        t_on_reply("1");
        if (!t_relay()) {
                if (method=="INVITE" || method=="ACK") {
                        end_media_session();
                };
                sl_reply_error();
        };
}
route[5] {
        log(1, "SER: Voice Mail section route(5)\n");
    # voicemail route #2
    #
    # this path this executed during these conditions:
    #
    #       cond 1) the called number is not in the location table
    #       cond 2) the from_uri == to_uri (ie, caller==callee)

    if (method=="INVITE" || method=="ACK") {
            use_media_proxy();
    };
    rewritehostport("99.99.99.100:5060");
    t_on_reply("1");
    if (!t_relay()) {
            if (method=="INVITE" || method=="ACK") {
                    end_media_session();
            };
            sl_reply_error();
    };

}
route[6] {
        log(1, "SER: International Call Off-Net section route(6)\n");
    # All International Calls Go To CISCO 5300
    if (method=="INVITE") {
            if (!proxy_authorize("", "subscriber")) {
                    proxy_challenge("", "0");
                    break;
            } else if (!check_from()) {
                    log(1, "Spoofed SIP call attempt");
                    sl_send_reply("403", "Use From=ID");
                    break;
            } else if (!(is_from_local() || is_uri_host_local())) {
                    sl_send_reply("403", "Please register to use our service");
                    break;
            };
            # enable caller id blocking for PSTN calls
            if (isflagset(25)) {
                    append_rpid_hf();
            };
    };
    # SIP->PSTN calls get 45 seconds to timeout
    avp_write("i:45", "inv_timeout");
    rewritehost("61.220.190.243");
    if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" &&

!search("^Route:")){
                sl_send_reply("479", "We don't forward to private IP addresses");
                break;
        };
        if (method=="INVITE" || method=="ACK") {
                use_media_proxy();
        };
        t_on_reply("1");
        if (!t_relay()) {
                if (method=="INVITE" || method=="ACK") {
                        end_media_session();
                };
                sl_reply_error();
        };
}
route[7] {
  log(1, "SER: Caller Blocked section route(7)\n");
    # caller blocked announcment
    #
    # this path this executed if a caller has been blocked

    if (method=="INVITE" || method=="ACK") {
            use_media_proxy();
    };
    rewriteuri("sip:699@99.99.99.100:5060");
    t_on_reply("1");
    if (!t_relay()) {
            if (method=="INVITE" || method=="ACK") {
                    end_media_session();
            };
            sl_reply_error();
    };

}
route[8] {
        log(1, "SER: Anonymous Call Rejection section route(8)\n");
    # anonymous call rejection announcment
    #
    # this path this executed for anonymous callers

    if (method=="INVITE" || method=="ACK") {
            use_media_proxy();
    };
    rewriteuri("sip:698@99.99.99.100:5060");
    t_on_reply("1");
    if (!t_relay()) {
            if (method=="INVITE" || method=="ACK") {
                    end_media_session();
            };
            sl_reply_error();
    };

}
onreply_route[1] {
        # Not all 2xx messages have a content body so here we
        # make sure our Content-Length > 0 to avoid a parse error
        if (status=~"(180)|(183)|2[0-9][0-9]") {
                if (!search("^Content-Length:\ 0")) {
                        use_media_proxy();
                };
        };
        if (client_nat_test("1")) {
                fix_contact();
        };
}
failure_route[1] {
        log(1, "SER: Failure Route section failure_route(1)\n");
    # if caller hung up then don't sent to voicemail
    if (t_check_status("487")) {
            break;
    };
    if (isflagset(26) && t_check_status("486")) {
            # forward busy is flag 26
            if (avp_pushto("$ruri", "s:fwdbusy")) {
                    log(1, "SER: fork to fwdbusy\n");
                    avp_delete("s:fwdbusy");
                    append_branch();
                    resetflag(26);

                    # test for domestic PSTN gateway
                    if (uri=~"^sip:0[0-9]{9}@") {
                    # if (avp_check("$fwd_busy_type", "eq/dom/i")) {
                            # test for domestic PSTN gateway
                            log(1, "SER: Busy Failure and Jump to route(3)\n");
                            route(3);
                    } else if (uri=~"^sip:002[1-9][0-9]*@") {
                    # } else if (avp_check("$fwd_busy_type", "eq/int/i")) {
                            # test for international PSTN gateway
                            log(1, "SER: Busy Failure and Jump to route(6)\n");
                            route(6);
                    } else {
                            # default to sip call
                            log(1, "SER: Busy Failure and Jump to route(2)\n");
                            route(2);
                    };
                    break;
            };
    };

    # here we can have either voicemail __OR__ forward no answer
    if (isflagset(27) && t_check_status("408")) {
            # forward no answer is flag 27
            if (avp_pushto("$ruri", "s:fwdnoanswer")) {
                    log(1, "SER: fork to fwdnoanswer\n");
                    avp_delete("s:fwdnoanswer");
                    append_branch();
                    resetflag(27);

                    if (uri=~"^sip:0[0-9]{9}@") {
                    # if (avp_check("$fwd_no_answer_type", "eq/dom/i")) {
                            # test for domestic PSTN gateway
                            log(1, "SER: No Answer Failure and Jump to route(3)\n");
                            route(3);
              } else if (uri=~"^sip:002[1-9][0-9]*@") {
                    # } else if (avp_check("$fwd_no_answer_type", "eq/int/i")) {
                            # test for international PSTN gateway
                            log(1, "SER: No Answer Failure and Jump to route(6)\n");
                            route(6);
                    } else {
                            # default to sip call
                            log(1, "SER: No Answer Failure and Jump to route(2)\n");
                            route(2);
                    };
                    break;
            };
    } else if (isflagset(31) && avp_pushto("$ruri", "$voicemail")) {
            avp_delete("$voicemail");
            log(1, "SER: No Answer Failure and Jump to route(4)\n");
            route(4);
            break;
    };

}

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Re: [Serusers] error: mediaproxy/sendMediaproxyCommand(): can't connect to MediaProxy

My ser.cfg list below:

Subset of ser.cfg about route[3:

Dump using ngrep -d eth0 -W byline port 5060

Subset of ser.cfg about route[3:

Dump using ngrep -d eth0 -W byline port 5060

Subset of ser.cfg about route[3:

Dump using ngrep -d eth0 -W byline port 5060

Subset of ser.cfg about route[3:

Dump using ngrep -d eth0 -W byline port 5060

Subset of ser.cfg about route[3:

Dump using ngrep -d eth0 -W byline port 5060