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37455 discussions

[Kamailio-Users] t_newtran/t_reply vs. sl_send_reply
by mayamatakeshi 31 Jul '08

31 Jul '08

Hello, sorry if this is a stupid question, but in the openser.cfg that comes with the source, I see the following: if (!lookup("location")) { switch ($retcode) { case -1: case -3: t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } Could you tell me why we use t_newtran/t_reply for cases -1,-3 and sl_send_reply for case -2. I read docs for modules sl and tm but I still don't get it why we shoud created a transaction in one situation and not in the other. Regards, takeshi

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[Kamailio-Users] CDRTool: no price showing
by Brian Del Shasta 30 Jul '08

30 Jul '08

I'm attempting to understand how call rating works within CDRTool. I am able to link destinations to the calls, but unable to generate prices. I'm using CDRTool 6.4.1 with FreeRadius/MySQL and a Cisco gateway. What I'm trying to setup seems simple: I have one set rate of $.08/min for any call beginning with a 1 (I'll exclude 800 numbers later). This rate should be applied globally with no differentiation between caller party or destination... Any help is greatly appreciated...

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[Kamailio-Users] pua_bla and presence
by Gustavo Mistrinelli 30 Jul '08

30 Jul '08

Hey, I have the same problem , using the same openser configuration file, only adding pua_bla "server_address" parameter on 1.3.2 bla stop working. I installed 2 openser once with 1.3.0 and the other with openser 1.3.2, then start openser using full debug log level to see what happens with each one. I'm using Polycom 650 hardphones Basic console error log ------------------------------------- ERROR:pua_bla:bla_handle_notify: content length= 0 WARNING:tm:t_unref: script writer didn't release transaction http://pastebin.com/m4a1b7969 (OpenSER 1.3.2 debug logs) Full debug log ----------------------- http://pastebin.com/m177c219a (OpenSER 1.3.0 full log) http://pastebin.com/m36f3ffca (OpenSER 1.3.2 full log) The main difference that I saw comparing both logs are remarked in pastbin, and refer to: "DBG:pua:print_subs: pres_uri" Please see below both configuration parameters Openser Parameters ---------------------------------- # Presence server module - common API loadmodule "presence.so" modparam("presence", "server_address", "sip:192.168.4.10") modparam("presence", "max_expires", 3600) # Presence server module - presence & watcher info and XCAP loadmodule "presence_xml.so" modparam("presence_xml", "force_active", 1) # Common API for presence user agent client loadmodule "pua.so" # BLA extension for PUA loadmodule "pua_bla.so" modparam("pua_bla", "default_domain", "192.168.4.10") modparam("pua_bla", "header_name", "Sender") modparam("pua_bla", "server_address", "sip:192.168.4.10") (Only for OpenSER 1.3.2 I'm sending configuration of affected routes (Main, Register, Presence) so you can see if there is an error. route { # Sanity Check Section #..... # Loose Route Section if (loose_route()) { xlog("L_INFO", "$mi Loose Routing"); if(!has_totag()) { xlog("L_INFO", "OPENSER_NAME: $mi Initial loose-routing rejected - \ M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("403", "Initial Loose-Routing Rejected"); route(ACC_FORBIDDEN_ROUTE); return; } if (is_method("CANCEL") || is_method("ACK")) { xlog("L_INFO", "$mi CANCEL/ACK"); route(BASE_LOCAL_ROUTE); return; } if (is_method("SUBSCRIBE") && $rd == "OPENSER_IP") { # in-dialog subscribe requests route(PRESENCE_ROUTE); return; } if(is_method("BYE")) { route(ACC_BYE_ROUTE); xlog("L_INFO", "(ACC) $mi Setting account_flag"); setflag(ACC_FLAG); route(BASE_LOCAL_ROUTE); return; } }; # Call Type Processing Section if (method=="INVITE") { route(INVITE_ROUTE); return; }; if (method=="REGISTER") { route(REGISTER_ROUTE); return; } if(is_method("PUBLISH|SUBSCRIBE")) { route(PRESENCE_ROUTE); return; } if(is_method("NOTIFY")) { if($hdr(event)=="dialog;sla") { route(PRESENCE_ROUTE); return; } } } route[REGISTER_ROUTE] { # ----------------------------------------------------------------- # REGISTER Message Handler # ----------------------------------------------------------------- # This cool register script came from http://openser.org/docs/scripting.html xlog("L_NOTICE", "OPENSER_NAME: $mi Register Route - \ M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); if ((!www_authorize("OPENSER_DOMAIN", "device")))) { xlog("L_INFO", "OPENSER_NAME: $mi Register Authentication Failed - \ M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); www_challenge("OPENSER_DOMAIN", "0"); sl_send_reply("401", "Unauthorized"); exit; } else xlog("L_INFO", "OPENSER_NAME: $mi Register Authentication OK -\ M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); if ( !(src_ip=="LOCAL_PAIR_IP" && src_port==LOCAL_PAIR_PORT) ) { if($hdr(User-Agent) =~ "PolycomSoundPointIP-SPIP_650-UA") { xlog("L_NOTICE", "$mi Polycom BLA set FLAG \ M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); bla_set_flag(); } if (!save("location")) { xlog("L_NOTICE", "$mi $fu save(location) Failed\n"); sl_reply_error(); return(0); }; } consume_credentials(); return; } route[PRESENCE_ROUTE] { # ------------------------------------------------------------------- # Presence Route # ------------------------------------------------------------------- xlog("L_INFO","$mi: Presence Route"); if (!t_newtran()){ sl_reply_error(); exit; }; if(is_method("PUBLISH")){ if($hdr(Sender)!= NULL) { handle_publish("$hdr(Sender)"); } else { handle_publish(""); } t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } else if(is_method("NOTIFY")) { if(bla_handle_notify()) { t_reply("200", "OK"); t_release(); } exit; } Ok, I hope that provided information help to see if there is a problem on my configuration or a is a bug on OpenSER 1.3.2 Thanks in advance, Cheers, Gustavo Mistrinelli

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[OpenSER-Users] OpenSER as NAT traversal proxy HELP!
by Joris Dobbelsteen 30 Jul '08

30 Jul '08

Dear, I'm really trying to use OpenSER as a NAT traversal SIP proxy, since my home phone keeps breaking voice channels (the box was not intended behind NAT and I'm, of course, using a configuration that no so well supported). What is the idea: SIP transactions should travel this way: ZyXEL UA <-> SIP Proxy <-> NAT Firewall (iptables) <-> {Internet} RTP should travel this way: ZyXEL UA <-> NAT Firewall & RTPProxy <-> {Internet} My current test is using X-Lite with voipbuster, but that doesn't really work. It seems that registers are functioning, at least X-Lite reports itself being registered. Voice calls always end up in timeouts, so something is really going wrong here, it might be authentication problems? An added problem is that I have just sufficient knowledge of SIP to see what it is doing, without really knowing what to expect exactly. Furthermore I have virtually no knowledge of OpenSER. I've quite a hard time even grasping the configuration I typed in. This is not really helpful What I do know: * SIP Proxy traffic is flowing. * SIP INVITES don't work at all. * SIP to RTP is communication, but I don't know if RTP is actually flowing. I stole most of the configuration from the "04 NAT Traversal" slides of the "Italy 2007 Admin course", to which there is link on the documentation site. I adapted it to make it work with the debian supplied OpenSER 1.1. How do I get this all working? What am I getting wrong? I really really appeciate any help I can get to get it working! - Joris Config is this: # ----------- global configuration parameters ------------------------ debug=4 # debug level (cmd line: -dddddddddd) fork=yes # Set to no to enter debugging mode log_stderror=no # (cmd line: -E) Set to yes to enter debugging mode check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) advertised_address="82.168.191.xx" advertised_port=5060 port=5060 children=4 fifo="/tmp/openser_fifo" # # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database mpath="/usr/lib/openser/modules/" loadmodule "mysql.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "nathelper.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "auth.so" loadmodule "auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line #modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # 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) # -- nathelper params --- modparam("nathelper", "rtpproxy_sock", "udp:192.168.10.6:22222") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) #modparam("nathelper", "sipping_bflag", 7) modparam("nathelper", "sipping_from", "sip:pinger@82.168.191.xx") # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # NAT detection route(2); # 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(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS #if(uri=~"@tls_domain1.net") { # t_relay("tls:domain1.net"); # exit; #} else if(uri=~"@tls_domain2.net") { # t_relay("tls:domain2.net"); # exit; #} route(1); }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("sip.familiedobbelsteen.nl", "subscriber")) { www_challenge("sip.familiedobbelsteen.nl", "0"); exit; }; if (isflagset(5)) { # set branch flag -- when someone will call this user # INVITE will have branch flag 6 set after loopup("location") setflag(6); # if you want OPTIONS natpings uncomment next # setflag(7); }; save("location"); exit; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (subst_uri('/(sip:.*);nat=yes/\1/i')) { setflag(6); }; if (isflagset(5) || isflagset(6)) { route(3); }; if (!t_relay()) { sl_reply_error(); }; exit; } route[2] { force_rport(); if(nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); }; setflag(5); }; } route[3] { if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")) { force_rtp_proxy("", "82.168.191.xx"); t_on_failure("2"); }; if (isflagset(5)) search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); t_on_reply("1"); } failure_route[2] { if (isflagset(6)||isflagset(5)) { unforce_rtp_proxy(); }; } onreply_route[1] { if ((isflagset(5) || isflagset(6)) && status =~ "(183)|(2[0-9][0-9])") { force_rtp_proxy(); }; search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); if (isflagset(6)) { fix_nated_contact(); }; exit; }

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[OpenSER-Users] reply route failure
by Fero Vlkolinsky 30 Jul '08

30 Jul '08

Hello, I am getting the folowing error on communication with one of our application servers: tm:t_should_relay_response: pick_branch failed (lowest==-1) for code 404 This is in situation where I have two triggerings to the AS (orig and term) and term subscriber is not registered to OpenSER. OpenSER thus returns 404 not found which is then sent back to OpenSER to be forwarded to orig leg of the AS. I had the very same behavior with other AS in our lab and found out that it was because of tag added to To header in AS for the 404 response. So i removed the tag and the call worked fine. However with this AS it doesn't seem to work. Is there a way to hack pick_branch so that I see what are the existing branches and why it does not match any? Or is there any easier way to find out the reason? Thanks a lot, Fero --- reklama ----------------------------------------------------- Hľadáš DOM – BYT – POZEMOK – KANCELÁRIU ? http://www.reality.sk

2 1

[Kamailio-Users] Add a table to openser DB
by Diego Costantini 30 Jul '08

30 Jul '08

Hi, I need to use something like this (openser on a pc with 2 network cards): http://www.voip-info.org/wiki/index.php?page=SER+example+outboundproxy since I couldn't find any documentation to properly create a "location_external" table, I followed this procedure: http://www.tutorialspoint.com/mysql/mysql-clone-tables.htm Now I have the table I need, but when I start openser I get this error: Jul 30 15:49:32 [8374] ERROR:usrloc:register_udomain: Invalid table version (use openser_mysql.sh reinstall) Jul 30 15:49:32 [8374] ERROR:registrar:domain_fixup: failed to register domain Jul 30 15:49:32 [8374] ERROR:core:fix_actions: fixing failed (code=-1) at cfg line 451 Jul 30 15:49:32 [8374] CRITICAL:core:fix_expr: fix_actions error Jul 30 15:49:32 [8374] ERROR:core:main: failed to fix configuration with err code -1 Jul 30 15:49:32 [8374] NOTICE:presence:destroy: destroy module ... How can I make Openser happy to accept my new table? Diego

2 1

[Kamailio-Users] Getting SNMPStats to work
by Alex Massover 30 Jul '08

30 Jul '08

Hello! I'm trying to setup SNMPStats on Debian Sid. I have: dev5:~# cat /etc/openser/openser.cfg | grep snmp loadmodule "snmpstats.so" modparam("snmpstats", "snmpgetPath", "/usr/bin/") modparam("snmpstats", "sipEntityType", "proxyServer") modparam("snmpstats", "MsgQueueMinorThreshold", 1234) modparam("snmpstats", "MsgQueueMajorThreshold", 5678) modparam("snmpstats", "dlg_minor_threshold", 500) modparam("snmpstats", "dlg_minor_threshold", 600) dev5:~# cat /etc/snmp/snmpd.conf | grep agent master agentx agentXSocket tcp:localhost:705 dev5:~# cat /etc/snmp/snmpstats.conf agentXSocket tcp:localhost:705 dev5:~# snmpd -f -D -Dagentx -x tcp:localhost:705 2>&1 | less No log handling enabled - turning on stderr logging registered debug token agentx, 1 netsnmp_assert !"registration != duplicate" failed agent_registry.c:535 netsnmp_subtree_load() netsnmp_assert !"registration != duplicate" failed agent_registry.c:535 netsnmp_subtree_load() netsnmp_assert !"registration != duplicate" failed agent_registry.c:535 netsnmp_subtree_load() agentx_register_app_config_handler: registering .conf token for "agentxsocket" agentx_register_app_config_handler: registering .conf token for "agentxperms" agentx_register_app_config_handler: registering .conf token for "agentxRetries" agentx_register_app_config_handler: registering .conf token for "agentxTimeout" /etc/snmp/snmpd.conf: line 115: Warning: Unknown token: system.sysDescr.0. Turning on AgentX master support. agentx/config: port spec: tcp:localhost:705 agentx/master: initializing... agentx/master: initializing... DONE NET-SNMP version 5.4.1 agentx/master: unexpected callback op 4 agentx/master: handle pdu (req=0x365b5eac,trans=0x0,sess=0x0) agentx/master: open 0x86b37e8 agentx/master: opened 0x86c2028 = 6 with flags = a0 agentx/master: send response, stat 0 (req=0x365b5eac,trans=0x0,sess=0x0) But it doesn't work: dev5:~# snmpget -v2c -c public localhost openserSIPEntityType.0 OPENSER-SIP-COMMON-MIB::openserSIPEntityType.0 = No Such Object available on this agent at this OID Am I missing something? -- Best Regards, Alex Massover

1 0

[Kamailio-Users] OpenSER + Asterisk (Voicemail) Issue....
by Gerard A. Matthew 30 Jul '08

30 Jul '08

I'm having this little issue when implementing the voicemail feature. My openser.cfg looks like this in the failure route: if(!t_was_cancelled()) { revert_uri(); rewritehostport("voicemail.mydomain.com:5061"); append_branch(); #PREVENT SOME CRAZY VOICEMAIL LOOP xlog("L_INFO", "INFO: CALL TO VOICEMAIL"); setflag(10); route(1); } On my asterisk end after the time out, i'm viewing the following: SELECT * FROM sipusers WHERE name = 'XXXXXX' i.e XXXXXX = the PSTN number i'm using to call into the IP phone that's connected to OpenSER There seems to be a simple mixup with the number that is sent to asterisk. Obviously there is no user with the PSTN number, however there is one with the called number. Any idea as to what wold be causing this? Have I provided enough information? Regards, Gerard.

2 1

[OpenSER-Users] Polycom Phones produce no wringing when dialing pots lines on the pstn
by Alex G 30 Jul '08

30 Jul '08

I'm sure i'm not the first to experience this. My polycom phones when calling out a pstn gateway recieve far end ringing when dialing cellphones in the US, but when i try to call a pots line i get no ringing, only silence until the end point answers the call. Does anyone have a solution for this? From what i can see in the sip captures, the gateway sends a 180 ringing on the pots line calls, but when i call a cellphone i recieve a 183 Session Progress where the phones do provide the ringing. A polycom to polycom call shows the same 180 ringing, but in this case they do provide ringing. Just to be clear, the ringing i am talking about is the rinigng you here in the earpiece after a dial. Any insight or help would be very appreciated. I have sip captures that i can post, but opted not to incase someone has seen this and knows the answer first. Thanks so much!

3 2

Re: [Serusers] use_dns_cache and call forwarding
by caio 29 Jul '08

29 Jul '08

Hi Miklos, This happens only when I try to do a call to a subcriber having forwarding enabled. Normal calls success ok. I'm not sure about the append_branch() usage and how to work with 'on busy' and 'no answer' forwarding features. Thanks your comments.. Claudio On Thu, Jul 24, 2008 at 4:58 AM, Miklos Tirpak <miklos(a)iptel.org> wrote: > Hi, > > On 07/23/2008 05:22 PM, caio wrote: >> >> Hi guys, >> I have a problem with call forwarding (on busy or not answer).., the >> call fall into the failure_route(1) block, and AVP checking and pushto >> does its job. >> >> Here a snipped of code of failure_route(1) (flag 27 means call fwd if >> no answer): >> >> if (isflagset(27) && t_check_status("408")) { # if fwd no >> answer is set and reply msg is 408 >> >> if (avp_pushto("$ruri", "s:fwdnoanswer")) { >> avp_delete("s:fwdnoanswer"); >> resetflag(27); >> >> avp_print(); >> >> log(1, "LOG: --> appending new branch..."); >> if (!append_branch()){ >> t_reply("500", "Too many branches?!"); >> drop; >> } >> >> log(1, "LOG: --> Calling route_6..."); >> route(6); >> break; >> >> Route(6) finally calls route(1), where t_relay is called. >> But have a dns failure..and the call never is forwarded. >> >> == ser.log == >> >> DEBUG: mk_proxy: doing DNS lookup... >> get_record: lookup(_sip._udp.wsa.lab, 33) failed_ >> sip_resolvehost: no SRV record found for wsa.lab, trying 'normal' >> lookup..._ >> check_via_address(190.244.33.5, 190.244.33.5, 0)_ >> ERROR: udp_send: sendto(sock,0x2ab32599c720,910,0,0x2ab325999e68,16): >> Operation not permitted(1)_ >> msg_send: ERROR: udp_send failed_ >> ERROR: t_forward_nonack: sending request failed_ >> DEBUG: add_to_tail_of_timer[4]: 0x2ab325999e80_ >> DEBUG: add_to_tail_of_timer[0]: 0x2ab325999ea0_ >> ERROR: w_t_relay (failure mode): forwarding failed_ >> >> A question regarding this trouble.....Can I disable use_dns_cache ? >> SER version is 0.9.7. > > there is no DNS cache in version 0.9.x as I recall, but anyway, your problem > does not seem to be related to DNS, but to the UDP message sending: SER > fails to send out the UDP packet, "Operation not permitted". Is there no > firewall or routing problem at the proxy side? > > Miklos > >> What do you recommend? >> If you need my ser.cfg I can paste it... >> >> Thanks.. >> > -- caio -- caio

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