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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