- sr-users - kamailio.org

[Kamailio-Users] Rewrite $tU
by Spinov Evgeniy 27 Mar '11

27 Mar '11

>Hello, >with the latest version there are alternatives you can use: >> On 12/10/09 5:06 PM, David wrote: >> Hey, >> >> I won't pretend to be an expert in Kamailio, someone will probably >> suggest a better way. But here is how I rewrote my SIP packet's TO >> header before relaying it to the next hop : >> >> # Rewrite RURI >> avp_printf("$rU", "prefix_$rU"); >$rU = "prefix_" + $rU; >> >> # Remove existing header >> remove_hf("To") ; >> >> # Add new header which is copied from RURI >> insert_hf("To: $ru\r\n", "From"); >you can do in one step, so you avoid operations on r-uri: >insert_hf("To: sip:prefix_$rU@$rd\r\n", "From"); >> >> >> DISCLAIMER : Several threads have said that changing the TO header is >>a bad idea, so I only recommend using this as a last resort. >If devices are RFC3261 (SIP 2.0), then it is ok. If there are SIP 1.0 >devices, then you have to preserve To/From header. >For the future, I would add options to change the From/To values via >pseudo-variables. It is the admin to decide to use it or not -- now >people need and do workarounds with subst or append_hf(). >Cheers, >Daniel I use Kamailio 3.0.3, and tried to use both of methods, i.e. insert_ft() and reassigning pseudo variable $rU and all mentioned in this topic. However, handle_subscribe() and dlg_manage() somewhy uses unmodified packet. Is there any workaround to handle $rU or $tU overwrites that for pua_dialoginfo module? I'm doing it like that: $var(orig_uri) = $rU; $rU = $(fU{re.subst,/(.*)_(.*)/\1/}) + "_" + $rU; if (is_method("INVITE")) { dlg_manage(); } if (is_method("SUBSCRIBE")) { handle_subscribe(); } $rU = $var(orig_uri); However, according to debug messages, NOTIFY's which are coming just after INVITE are still without prefixes, but packets on Asterisk, which is after Kamailio, are coming modified. Please, advice. > > David > > > > > > David a écrit : >> Hey, >> >> I do understand that normally the TO header should not be written, my >> problem is that I am using the presence_dialoginfo module which uses the >> TO header and not the RURI, which means that in order to tell the >> difference between customer A's extension 101 and Customer B's extension >> 101, I rewrite the TO header to add context. >> >> This is actually the ame problem that I started discussing here : >> http://lists.kamailio.org/pipermail/users/2009-August/024316.html >> >> Trouble is I have not had time to work on this since then. >> >> Thanks, >> >> David >> >> Olle E. Johansson a écrit : >>> 10 dec 2009 kl. 15.04 skrev Iñaki Baz Castillo: >>> >>> >>>> El Jueves, 10 de Diciembre de 2009, David escribió: >>>>> Hey, >>>>> >>>>> I need to rewrite the $tU variable on a packet before it is >>>>> relayed. It >>>>> is ok if the change is only on the relayed packet and the $tU >>>>> variable >>>>> change. The important thing is that I modify the To header, it seems >>>>> that rewritting $tU would be the easiest. >>>> $tU is not writable, but just readable. >>>> If you want to change the "To" header you should do some "dirty" >>>> replacing operation (see textops module). >>> >>> This is propably something we need to change and make the To and >>> From headers writable. >>> It's not dirty to change them any more. >>> >>> David: In the original SIP 1.0 protocol the To/From headers was part >>> of the dialog identification, thus you where not allowed to change >>> them in a proxy. THis was kept in RFC3261 - sip v2 - but has been >>> removed in an additional RFC a short while ago (seen from SIP history). >>> >>> Regards, >>> /O >>> _______________________________________________ >>> Kamailio (OpenSER) - Users mailing list >>> Users at lists.kamailio.org >>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >>> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users >> >> >> >> _______________________________________________ >> Kamailio (OpenSER) - Users mailing list >> Users at lists.kamailio.org >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Kamailio (OpenSER) - Users mailing list > Users at lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users > -- Daniel-Constantin Mierla * http://www.asipto.com/

3 10

Presence, RLS, and NAT Traversal
by Peter Dunkley 26 Mar '11

26 Mar '11

Hello, I am using Kamailio and nathelper to connect UAs behind a NAT. However, I want to use presence and RLS in this system as well. The problem I have is that the presence and RLS modules are not nathelper aware and the NOTIFY requests that they generate are targeted at the private network IP address and port of the UA. Has anyone any experience of this? Can anyone help? I had a cunning plan where, in route[NAT], I detected SUBSCRIBE requests that originate from behind a NAT (and do not have myself as the last hop). When I see a request like this I use add_contact_alias(), record_route() and forward it to myself. I then added a handle_ruri_alias() to the WITH_NAT section of route[RELAY]. The theory being that an in-dialog NOTIFY from presence or RLS will use the route-set established by the SUBSCRIBE transaction. This almost works. The presence.winfo NOTIFY from the presence module and the initial presence NOTIFY from the RLS module make it back to the client successfully. However, when this forwarding of the SUBSCRIBE is in place the RLS subscriptions don't happen. I know it is the forward of the SUBSCRIBE that is the trigger behind this problem because simply commenting that out (and allowing the first SUBSCRIBE to get to RLS) makes the RLS subscriptions happen (but in this case the NOTIFYs get back to the client and are 404'd because the request URI is not correct). The only difference between the first SUBSCRIBE and the forwarded SUBSCRIBE is an additional "Via:" header and a "Record-Route:" header. I ran Kamailio with the -E -ddd options and observed that: * modules_k/rls/subscribe.c:rls_get_service_list() is outputting the line "searching document for user sip:@" * This function is called by modules_k/rls/subscribe.c:rls_handle_subscribe() and the output suggests that GET_FROM_PURI(msg)->user and GET_FROM_PURI(msg)->host are empty strings * However, parse_from_header(msg) is called near the top of rls_handle_subscribe() and clearly succeeds to find the "From:" header in the message * As I mentioned above the forwarded SUBSCRIBE is almost identical to the first SUBSCRIBE, and so it does contain a valid "From:" header All I can think of is that Kamailio is becoming confused by the two near-identical SUBSCRIBE requests and that the "cleaning up" of the first (statelessly forwarded) SUBSCRIBE is also "cleaning up" the forwarded SUBSCRIBE that I am currently working with? My route[NAT]: route[NAT] { xlog("L_INFO", "$rm: route[NAT]\n"); #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19") && $si!="MY_REAL_IP" && $si! ="127.0.0.1") { xlog("L_INFO", " Request from NAT'd UAC\n"); if (is_method("REGISTER")) { fix_nated_register(); } else if (is_method("SUBSCRIBE")) { add_contact_alias(); route(RECORD_ROUTE); forward(127.0.0.1, MY_REAL_PORT); exit; } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; } My route[RECORD_ROUTE]: route[RECORD_ROUTE] { # remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); return; } The start of my route[RELAY]: route[RELAY] { xlog("L_INFO", "$rm: route[RELAY]\n"); #!ifdef WITH_NAT if (check_route_param("nat=yes")) { setbflag(FLB_NATB); } if (isflagset(FLT_NATS) || isbflagset(FLB_NATB)) { route(RTPPROXY); } handle_ruri_alias(); #!endif Regards, Peter -- Peter Dunkley Technical Director Crocodile RCS Ltd

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uri unescape transformation?
by Juha Heinanen 26 Mar '11

26 Mar '11

i could not find a transformation or function to unescape an uri, such as this: sip%3Atest%40test.fi did i miss it or is there a need to write one? -- juha

3 3

uri unescape transformation?
by Juha Heinanen 26 Mar '11

26 Mar '11

i could not find a transformation or function to unescape an uri, such as this: sip%3Atest%40test.fi did i miss it or is there a need to write one? -- juha

3 6

issue: fr_inv_timer_avp in Kamailio 3.1
by Min Wang 26 Mar '11

26 Mar '11

Hi I am trying to use fr_inv_timer_avp like the following in Kamailio 3.1: #!KAMAILIO ... modparam("tm", "fr_inv_timer_avp", "$avp(i:2)") modparam("tm", "fr_inv_timer", 180000) .. $avp(i:2)=10 ... # t_set_fr(10000); ... t_on_failure("failroute"); It seems it is not working. the invite will timeout in 3 min, not 10 seconds. If t_set_fr is used, it is working ( 10 seconds timeout). the t_set_fr issue is it can only accept the constant, not avp. I saw there was a discussion in the april 2010, http://lists.sip-router.org/pipermail/sr-users/2010-April/027101.html Not sure fr_inv_timer_avp has been tested or not in Kamailio 3.0/3.1? A bug or my misuage the fr_inv_timer_avp? thanks -- Kind regards / Mit freundlichen Grüßen Min Wang BASIS AUDIONET North America BASIS AUDIONET * 716 N. Bethlehem Pike - Suite 300 * Lower Gwynedd * PA 19002 * USA Phone +1 215 674 8600 ext 451 Fax +1 215 646 4959 wang(a)basis-audionet.com www.basis-audionet.com Europe BASIS AUDIONET GmbH * Jessenstraße 4 * 22767 Hamburg * Germany Phone +49 40 85 31 14-0 Fax +49 40 85 31 14-33 www.basis-audionet.de Registered: Hamburg HR B 85866 Managing Director: Lutz Schaffhausen, Rüdiger Kaun

3 8

RES: event_route and acc_db_request().
by Alexandre Abreu 26 Mar '11

26 Mar '11

Hi Daniel, Using 'xmlrpc' and 'mi' it works. When I finish this project, I will write a small text explaining the steps I did to make this work. Maybe this will benefit more people from the community. Thank you. Alexandre -----Mensagem original----- De: Daniel-Constantin Mierla [mailto:miconda@gmail.com] Enviada em: quinta-feira, 24 de março de 2011 03:51 Para: Alexandre Abreu Cc: 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List' Assunto: Re: RES: RES: [SR-Users] event_route and acc_db_request(). Hello, the method name is 'mi' then the first parameter is dlg_terminate_dlg and the next parameters are the ones required by the dlg_terminate_dlg. The XMLRPC command 'mi' is to execute commands from the old control interface that we named MI: http://kamailio.org/docs/modules/stable/modules/mi_rpc.html#id2945832 Cheers, Daniel On 3/23/11 4:30 PM, Alexandre Abreu wrote: > Hi Daniel, > > Look the request/reply. > > <...> > <?xml version="1.0" encoding="UTF-8"?>. > <methodCall>. > <methodName>mi dlg_terminate_dlg</methodName>. > <params>. > <param><value><string>065fa668d8485f33</string></value></param>. > </params>. > </methodCall>. > > <...> > <?xml version="1.0"?> > <methodResponse> > <fault> > <value> > <struct> > <member> > <name>faultCode</name> > <value><int>500</int></value> > </member> > <member> > <name>faultString</name> > <value><string>Method Not Found</string></value> </member> </struct> > </value> </fault> </methodResponse> > > This is module xmlrpc handling the POST at route[XMLRPC]. I am not > sure if I understand correctly how you suggested the use of 'mi dlg_terminate_dlg' > command. > > Thanks, > Alexandre > > -----Mensagem original----- > De: Daniel-Constantin Mierla [mailto:miconda@gmail.com] Enviada em: > quarta-feira, 23 de março de 2011 11:10 > Para: Alexandre Abreu > Cc: 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List' > Assunto: Re: RES: [SR-Users] event_route and acc_db_request(). > > Hello, > > On 3/23/11 2:36 PM, Alexandre Abreu wrote: >> Hello Daniel. >> >> The module being used is 'mi_xmlrpc.so' > ok, this one creates a new process for handling xmlrpc requests. It is > pretty much clear that is very likely what I suspected first time. > > If you have time, try load instead xmlrpc module and use the command > 'mi dlg_terminate_dlg ...". > > Cheers, > Daniel > >> I am using the Carsten's rtpproxy >> patch. RTPPROXY detects the media timeout and send a notification >> through XMLRPC lib with 'dlg_terminate_dlg callid' to Kamailio. After >> that, I configured the event_route to catch the local generated >> requests for accounting these BYEs. >> >> Thanks, >> Alexandre >> >> -----Mensagem original----- >> De: Daniel-Constantin Mierla [mailto:miconda@gmail.com] Enviada em: >> quarta-feira, 23 de março de 2011 10:17 >> Para: Alexandre Abreu >> Cc: 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - >> Users Mailing List' >> Assunto: Re: RES: [SR-Users] event_route and acc_db_request(). >> >> Hello, >> >> On 3/23/11 1:44 PM, Alexandre Abreu wrote: >>> Hello Daniel. >>> >>> The BYE is generated by kamctl/xmlrpc (dlg_terminate_dlg). >> is xmlrpc via mi_xmlrpc or via xmlrpc module? >> >> I have to double check, but looks very likely to be the fact that acc >> db connection was not initialized for MI/RPC children processes. >> >> Thanks, >> Daniel >> >>> Alexandre >>> >>> De: Daniel-Constantin Mierla [mailto:miconda@gmail.com] Enviada em: >>> quarta-feira, 23 de março de 2011 04:55 >>> Para: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - >>> Users Mailing List >>> Cc: Alexandre Abreu >>> Assunto: Re: [SR-Users] event_route and acc_db_request(). >>> >>> Hello, >>> >>> On 3/17/11 5:45 PM, Alexandre Abreu wrote: >>> Hello. >>> >>> Why acc_db_request() doesnt work on event_route? >>> >>> I will take a look. >>> >>> Is the BYE generated by dialog timeout or you trigger it by > kamctl/xmlrpc? >>> Cheers, >>> Daniel >>> >>> >>> >>> Mar 17 13:15:44 devel kamailio[25209]: INFO:<script>: Routing >>> locally generated BYE to<sip:2000000@192.168.200.114:9297> >>> Mar 17 13:15:44 devel kamailio[25209]: ERROR:<core> [db.c:421]: >>> invalid parameter value Mar 17 13:15:44 devel kamailio[25209]: ERROR: >>> acc [acc.c:391]: error in use_table Mar 17 13:15:44 devel >>> kamailio[25209]: INFO:<script>: Routing locally generated BYE >>> to<sip:2000001@192.168.200.149:7335> >>> Mar 17 13:15:44 devel kamailio[25209]: ERROR:<core> [db.c:421]: >>> invalid parameter value Mar 17 13:15:44 devel kamailio[25209]: ERROR: >>> acc [acc.c:391]: error in use_table >>> >>> event_route[tm:local-request] { >>> >>> xlog("L_INFO", "Routing locally generated $rm >>> to<$ru>\n"); >>> >>> if (is_method("BYE")) >>> acc_db_request("rtp-timeout", acc); } >>> >>> If I change acc_db_request() to acc_log_request() everything works >>> fine, but this BYE should go to database for accounting purposes. >>> >>> I am using GIT version from Kamailio branch 3.1. >>> >>> From 2008 there is a thread that also demonstrate this problem: >>> >>> http://www.mail-archive.com/users@lists.kamailio.org/msg01411.html >>> >>> Unfortunately, in the archives, theres no solution for that. >>> >>> Regards, >>> Alexandre >>> >>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing >>> list sr-users(a)lists.sip-router.org >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>> >>> >> -- >> Daniel-Constantin Mierla >> http://www.asipto.com >> > -- > Daniel-Constantin Mierla > http://www.asipto.com > -- Daniel-Constantin Mierla http://www.asipto.com

4 3

Error when using dispatcher module
by Gary Chen 25 Mar '11

25 Mar '11

Kamailio 3.1.2 I am using Kamailio in front of Freeswitch as simple sip router. To do multi-tenant in Freeswitch I created SIP domain (fs2000.ly.net) and used as sip domain inside my sip phone. This domain does not map to any real IP and no entry in our DNS server. If I sent the call directly to Freeswitch, it works fine and registered with Freeswitch. When I tried to go through Kamailio using dispatcher to route the call to freeswitch, I got this error: ERROR: <core> [resolve.c:1540]: ERROR: sip_hostport2su: could not resolve hostname: "fs2000.ly.net" The strange thing is that sometime the call did go through kamailio and registered with freeswitch. Whenever I restart Kamailio, I got above error and then the error went away by itself. Sometime took a minute and sometime took several hours. And sometime the error come back without restart. Does anybody know why? Why dispatcher want to resolve the hostname when it only lookup dispatcher for destination IP? Gary

1 0

kamailio 3.1: rtpproxy_sock multiple rtpproxies not working?
by Min Wang 24 Mar '11

24 Mar '11

hi I have two questions regarding kamailio 3.1 (1) It seems the following format for rtpproxy_sock not working in Kamailio 3.1 modparam("rtpproxy", "rtpproxy_sock", "udp:localhost:12221 udp:localhost:12222") the error is: 0(11881) ERROR: rtpproxy [rtpproxy.c:957]: Name or service not known ////////////////////////// 0(11881) DEBUG: <core> [sr_module.c:871]: DEBUG: init_mod: rtpproxy 0(11881) DEBUG: rtpproxy [rtpproxy.c:531]: url is udp:locahost:22222, len is 18 0(11881) DEBUG: rtpproxy [rtpproxy.c:531]: url is udp:localhost:22222, len is 19 0(11881) DEBUG: <core> [sr_module.c:871]: DEBUG: init_mod: nathelper .... 0(11881) DEBUG: <core> [sr_module.c:828]: DEBUG: init_mod_child (-127): rtpproxy 0(11881) ERROR: rtpproxy [rtpproxy.c:957]: Name or service not known 0(11881) ERROR: <core> [sr_module.c:832]: init_mod_child(): Error while initializing module rtpproxy (/usr/lib/kamailio/modules_k/rtpproxy.so) But if you do the following it is ok modparam("rtpproxy", "rtpproxy_sock","udp:localhost:12221") modparam("rtpproxy", "rtpproxy_sock","udp:localhost:12222") any clue or format changed for rtpproxy_sock? (2) BTW, how do I log stderror into syslog file? -- log_stderror=yes the error will print out in the console, #configure the /etc/syslog.conf for this feature log_facility=LOG_LOCAL1 this log_facility does not log the log_stderror?? Thanks. Kind regards / Mit freundlichen Grüßen Min Wang

2 2

RES: event_route and acc_db_request().
by Alexandre Abreu 24 Mar '11

24 Mar '11

Hello Daniel. The module being used is 'mi_xmlrpc.so' I am using the Carsten's rtpproxy patch. RTPPROXY detects the media timeout and send a notification through XMLRPC lib with 'dlg_terminate_dlg callid' to Kamailio. After that, I configured the event_route to catch the local generated requests for accounting these BYEs. Thanks, Alexandre -----Mensagem original----- De: Daniel-Constantin Mierla [mailto:miconda@gmail.com] Enviada em: quarta-feira, 23 de março de 2011 10:17 Para: Alexandre Abreu Cc: 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List' Assunto: Re: RES: [SR-Users] event_route and acc_db_request(). Hello, On 3/23/11 1:44 PM, Alexandre Abreu wrote: > Hello Daniel. > > The BYE is generated by kamctl/xmlrpc (dlg_terminate_dlg). is xmlrpc via mi_xmlrpc or via xmlrpc module? I have to double check, but looks very likely to be the fact that acc db connection was not initialized for MI/RPC children processes. Thanks, Daniel > Alexandre > > De: Daniel-Constantin Mierla [mailto:miconda@gmail.com] Enviada em: > quarta-feira, 23 de março de 2011 04:55 > Para: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List > Cc: Alexandre Abreu > Assunto: Re: [SR-Users] event_route and acc_db_request(). > > Hello, > > On 3/17/11 5:45 PM, Alexandre Abreu wrote: > Hello. > > Why acc_db_request() doesnt work on event_route? > > I will take a look. > > Is the BYE generated by dialog timeout or you trigger it by kamctl/xmlrpc? > > Cheers, > Daniel > > > > Mar 17 13:15:44 devel kamailio[25209]: INFO:<script>: Routing locally > generated BYE to<sip:2000000@192.168.200.114:9297> > Mar 17 13:15:44 devel kamailio[25209]: ERROR:<core> [db.c:421]: > invalid parameter value Mar 17 13:15:44 devel kamailio[25209]: ERROR: > acc [acc.c:391]: error in use_table Mar 17 13:15:44 devel > kamailio[25209]: INFO:<script>: Routing locally generated BYE > to<sip:2000001@192.168.200.149:7335> > Mar 17 13:15:44 devel kamailio[25209]: ERROR:<core> [db.c:421]: > invalid parameter value Mar 17 13:15:44 devel kamailio[25209]: ERROR: > acc [acc.c:391]: error in use_table > > event_route[tm:local-request] { > > xlog("L_INFO", "Routing locally generated $rm to<$ru>\n"); > > if (is_method("BYE")) > acc_db_request("rtp-timeout", acc); } > > If I change acc_db_request() to acc_log_request() everything works > fine, but this BYE should go to database for accounting purposes. > > I am using GIT version from Kamailio branch 3.1. > > From 2008 there is a thread that also demonstrate this problem: > > http://www.mail-archive.com/users@lists.kamailio.org/msg01411.html > > Unfortunately, in the archives, theres no solution for that. > > Regards, > Alexandre > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing > list sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > -- Daniel-Constantin Mierla http://www.asipto.com

2 3

binary name
by Claudio Furrer 24 Mar '11

24 Mar '11

Hello, What is (or should be) the binary main name when using FLAVOUR=ser, FLAVOUR=sip-router or no FLAVOUR specified? I'm getting "ser" as the name whatever I set to FLAVOUR var at compile time. Is it right? (Makefiles.defs says "ser" in these cases but not sure the intention of the sip-router project). I need to know the correct one to make an ebuild package for Gentoo. Thank you for your answers.. Regards, Claudio

3 7

event_route and acc_db_request().
by Alexandre Abreu 23 Mar '11

23 Mar '11

Hello. Why acc_db_request() doesn't work on event_route? Mar 17 13:15:44 devel kamailio[25209]: INFO: <script>: Routing locally generated BYE to <sip:2000000@192.168.200.114:9297> Mar 17 13:15:44 devel kamailio[25209]: ERROR: <core> [db.c:421]: invalid parameter value Mar 17 13:15:44 devel kamailio[25209]: ERROR: acc [acc.c:391]: error in use_table Mar 17 13:15:44 devel kamailio[25209]: INFO: <script>: Routing locally generated BYE to <sip:2000001@192.168.200.149:7335> Mar 17 13:15:44 devel kamailio[25209]: ERROR: <core> [db.c:421]: invalid parameter value Mar 17 13:15:44 devel kamailio[25209]: ERROR: acc [acc.c:391]: error in use_table event_route[tm:local-request] { xlog("L_INFO", "Routing locally generated $rm to <$ru>\n"); if (is_method("BYE")) acc_db_request("rtp-timeout", "acc"); } If I change acc_db_request() to acc_log_request() everything works fine, but this BYE should go to database for accounting purposes. I am using GIT version from Kamailio branch 3.1. >From 2008 there is a thread that also demonstrate this problem: http://www.mail-archive.com/users@lists.kamailio.org/msg01411.html Unfortunately, in the archives, there's no solution for that. Regards, Alexandre

2 3

RES: RTPPROXY timeout patch.
by Alexandre Abreu 23 Mar '11

23 Mar '11

Hi Carsten, Patch applied. It works now. Thanks, Alexandre. -----Mensagem original----- De: kaiserbock2(a)googlemail.com [mailto:kaiserbock2@googlemail.com] Em nome de Carsten Bock Enviada em: terça-feira, 22 de março de 2011 08:00 Para: Alexandre Abreu Cc: sr-users(a)lists.sip-router.org; RTPproxy Development Assunto: Re: RTPPROXY timeout patch. Hi Alexandre, it took quite a lot of searching, but now it's done. It was a general RTPProxy/PThread issue. The problem seems to be, that the PThread is created before the fork of the main process. Then the Thread waits forever... Attached an changed main.c + rtpp_defines.h which modifies this behaviour and starts the Thread after the fork. I will create an according patch and send it to the RTPProxy-Dev-Mailinglist as well. Have fun, Carsten 2011/3/22 Carsten Bock <lists(a)bock.info>: > Hi Alexandre, > > it seem to be a general RTP-Proxy/PThread issue. It seems not to be > related to the XML-RPC notification; the thread, which processes the > timeouts, seems to waits forever... I will check... > > Carsten > > 2011/3/21 Alexandre Abreu <alexandre.abreu(a)redt.com.br>: >> Hi Carsten, >> >> After more testing, I just realized that the notification works only if I >> use "-f" parameter in RTPPROXY. If I omit this parameter (to run in >> background mode), the session timeout but there's no notification sent >> (BYE). Why's that? >> >> Thanks, >> Alexandre. >> >> -----Mensagem original----- >> De: Alexandre Abreu [mailto:alexandre.abreu@redt.com.br] >> Enviada em: quarta-feira, 16 de março de 2011 14:41 >> Para: 'Carsten Bock' >> Cc: 'sr-users(a)lists.sip-router.org' >> Assunto: RES: RTPPROXY timeout patch. >> >> Carsten, >> >> Here it goes: >> >> [root@devel rtpproxy-carsten]# ./rtpproxy -T 10 -f -F -i -l 192.168.200.90 >> -s udp:*:7722 -d DBUG ERR INFO -n tcp:127.0.0.1:7723 >> INFO:main: rtpproxy started, pid 22495 >> rtpproxy: >>> Running Timeout-Process >> >> DBUG:handle_command: received command "22428_8 Uc0,8,101 080b5d23d1603667 >> 192.168.200.114 6380 0073852a;1" >> INFO:handle_command: new session 080b5d23d1603667, tag 0073852a;1 requested, >> type strong >> INFO:handle_command: new session on a port 48662 created, tag 0073852a;1 >> INFO:handle_command: pre-filling caller's address with 192.168.200.114:6380 >> DBUG:doreply: sending reply "22428_8 48662 192.168.200.90" >> DBUG:handle_command: received command "22436_8 Lc0,8,101 080b5d23d1603667 >> 192.168.200.149 7614 0073852a;1 c758967a;1 >> xmlrpc:http://127.0.0.1:8000/RPC2" >> INFO:handle_command: lookup on ports 48662/58604, session timer restarted >> INFO:handle_command: setting custom timeout handler >> (xmlrpc:http://127.0.0.1:8000/RPC2) >> INFO:handle_command: pre-filling callee's address with 192.168.200.149:7614 >> DBUG:doreply: sending reply "22436_8 58604 192.168.200.90" >> INFO:process_rtp: session timeout >> INFO:remove_session: RTP stats: 982 in from callee, 4 in from caller, 986 >> relayed, 0 dropped >> INFO:remove_session: RTCP stats: 5 in from callee, 1 in from caller, 6 >> relayed, 0 dropped >> INFO:remove_session: session on ports 48662/58604 is cleaned up >> ERR:do_timeout_notification: Timeout socket is: èîÈÄÐÊ@ÍÈÄÈÄ`ê >> DBUG:reconnect_timeout_handler: reconnecting timeout socket >> ERR:reconnect_timeout_handler: can't create timeout socket: Address family >> not supported by protocol >> ERR:do_timeout_notification: unable to send timeout notification >> >> Very weird chars in the debug of what Timeout socket is... >> >> But the custom timeout handler are being printed correctly from the config >> file. >> INFO:handle_command: setting custom timeout handler >> (xmlrpc:http://127.0.0.1:8000/RPC2) >> Here I am running CentOS 5.2 32-bit. >> >> Thanks, >> Alexandre >> >> -----Mensagem original----- >> De: kaiserbock2(a)googlemail.com [mailto:kaiserbock2@googlemail.com] Em nome >> de Carsten Bock Enviada em: quarta-feira, 16 de março de 2011 14:18 >> Para: Alexandre Abreu >> Cc: sr-users(a)lists.sip-router.org >> Assunto: Re: RTPPROXY timeout patch. >> >> Hi Alexandre, >> >> My version of RTP-Proxy is following: >> >> bock@bock-tde:~/ims/rtpproxy$ ./rtpproxy -v Basic version: 20040107 >> Extension 20050322: Support for multiple RTP streams and MOH Extension >> 20060704: Support for extra parameter in the V command Extension 20071116: >> Support for RTP re-packetization Extension 20071218: Support for forking >> (copying) RTP stream Extension 20080403: Support for RTP statistics querying >> Extension 20081102: Support for setting codecs in the update/lookup command >> Extension 20081224: Support for session timeout notifications Extension >> 20090810: Support for automatic bridging Extension 20100819: Support for >> timeout notifications using XML-RPC towards Kamailio/sip-router.org >> >> Please find attached a modified rtpp_notify.c file. I have just added some >> tiny debug output in order to see some points. >> Now you should see the following Debug-Outputs: >> >> rtpproxy: >>> Running Timeout-Process >> >> Then the notifier process is running. That would be good. If not, that's the >> reason why it is not working. When the request comes in, you should see the >> following: >> >> INFO:handle_command: setting custom timeout handler >> (xmlrpc:http://localhost:8000/RPC2) >> >> Then the Timeout-Socket was properly set, that would be good as well. >> Now the timeout: >> >> INFO:process_rtp: session timeout >> [...] >> ERR:do_timeout_notification: Timeout socket is: >> xmlrpc:http://localhost:8000/RPC2 >> >> That would be great, because then the Timeout towards the Kamailio should be >> triggerd. >> If these parts are ok, then there must be some issue either in the XML-RPC >> client library or in the communication between the RTP-Proxy and Kamailio. I >> hope you did a trace on the XML-RPC Port both on the RTPproxy and on the >> Kamailio? What distro are you using? My tests were only on Ubuntu and >> Debian, which are quite similar. >> >> Hope we find this issue, >> >> Kind regards, >> Carsten >> >> P.S.: I think i removed that check for the port for testing, that's why my >> version accepted the socket without port... (now i'm using "-n >> tcp:127.0.0.1:9999") >> >> 2011/3/16 Alexandre Abreu <alexandre.abreu(a)redt.com.br>: >>> Carsten, >>> >>> Indeed. Very strange. >>> >>> Are we running the same RTPPROXY version? How can you start using '-n >>> tcp:127.0.0.1' without specifying a port? >>> >>> [root@devel rtpproxy-carsten]# ./rtpproxy -T 10 -f -F -i -l >>> 192.168.200.90 -s udp:*:7722 -d DBUG ERR INFO -n tcp:127.0.0.1 >>> rtpproxy: can't parse host:port in TCP address >>> rtpproxy: can't start notification thread >>> >>> [root@devel rtpproxy-carsten]# ./rtpproxy -T 10 -f -F -i -l >>> 192.168.200.90 -s udp:*:7722 -d DBUG ERR INFO -n tcp:127.0.0.1:7723 >>> INFO:main: rtpproxy started, pid 21169 >>> >>> DBUG:handle_command: received command "17828_9 Uc0,8,101 >>> 4b10ce04de4f8818 >>> 192.168.200.114 6380 9c4b6265;1" >>> INFO:handle_command: new session 4b10ce04de4f8818, tag 9c4b6265;1 >>> requested, type strong >>> INFO:handle_command: new session on a port 43750 created, tag >>> 9c4b6265;1 >>> INFO:handle_command: pre-filling caller's address with >>> 192.168.200.114:6380 >>> DBUG:doreply: sending reply "17828_9 43750 192.168.200.90 " >>> DBUG:handle_command: received command "17847_9 Lc0,8,101 >>> 4b10ce04de4f8818 >>> 192.168.200.149 7386 9c4b6265;1 dd69ab1d;1 >>> xmlrpc:http://localhost:8000/RPC2" >>> INFO:handle_command: lookup on ports 43750/55796, session timer >>> restarted >>> INFO:handle_command: setting custom timeout handler >>> (xmlrpc:http://localhost:8000/RPC2) >>> INFO:handle_command: pre-filling callee's address with >>> 192.168.200.149:7386 >>> DBUG:doreply: sending reply "17847_9 55796 192.168.200.90 " >>> INFO:process_rtp: session timeout >>> INFO:remove_session: RTP stats: 548 in from callee, 5 in from caller, >>> 553 relayed, 0 dropped >>> INFO:remove_session: RTCP stats: 3 in from callee, 1 in from caller, 4 >>> relayed, 0 dropped >>> INFO:remove_session: session on ports 43750/55796 is cleaned up >>> DBUG:reconnect_timeout_handler: reconnecting timeout socket >>> ERR:reconnect_timeout_handler: can't create timeout socket: Address >>> family not supported by protocol >>> ERR:do_timeout_notification: unable to send timeout notification >>> >>> Above the error message. >>> >>> [root@devel ~]# md5sum rtpproxy-carsten.tar.gz >>> c02b1e2ac57d39562e86bcfc4ee592b8 rtpproxy-carsten.tar.gz >>> >>> Thanks, >>> Alexandre. >>> >>> -----Mensagem original----- >>> De: kaiserbock2(a)googlemail.com [mailto:kaiserbock2@googlemail.com] Em >>> nome de Carsten Bock Enviada em: quarta-feira, 16 de março de 2011 >>> 13:03 >>> Para: Alexandre Abreu >>> Cc: sr-users(a)lists.sip-router.org >>> Assunto: Re: RTPPROXY timeout patch. >>> >>> Hi Alexandre, >>> >>> That is strange: >>> >>> I run the RTP-Proxy like this (directly from the TAR-File, i sent you) >>> and Kamailio with attached config-file. >>> >>> bock@bock-tde:~/ims/rtpproxy$ sudo ./rtpproxy -T 10 -f -F -i -l >>> 127.0.0.1 -s udp:*:22222 -d DBUG -n tcp:127.0.0.1 >>> rtpproxy: Timer started. >>> INFO:main: rtpproxy started, pid 4203 >>> [... Kamailio connects to RTP-Proxy...] >>> DBUG:handle_command: received command "4259_8 >>> UAc98,97,99,104,3,0,8,9,96 56f0f83a-5373-46a1-b6f6-9ce2f93e68d5 >>> 195.71.4.203 4008 5371f039-40d0-4944-aae7-6f75071a2f8c;1" >>> INFO:handle_command: new session 56f0f83a-5373-46a1-b6f6-9ce2f93e68d5, >>> tag 5371f039-40d0-4944-aae7-6f75071a2f8c;1 requested, type strong >>> INFO:handle_command: new session on a port 45508 created, tag >>> 5371f039-40d0-4944-aae7-6f75071a2f8c;1 >>> INFO:handle_command: pre-filling caller's address with >>> 195.71.4.203:4008 >>> DBUG:doreply: sending reply "4259_8 45508 127.0.0.1 " >>> DBUG:handle_command: received command "4259_9 LAc98,96 >>> 56f0f83a-5373-46a1-b6f6-9ce2f93e68d5 195.71.4.203 4000 >>> 5371f039-40d0-4944-aae7-6f75071a2f8c;1 >>> 9915df0c-30fc-49c5-aa8a-c86b4242c094;1 >>> xmlrpc:http://localhost:8000/RPC2" >>> INFO:handle_command: lookup on ports 45508/45238, session timer >>> restarted >>> INFO:handle_command: setting custom timeout handler >>> (xmlrpc:http://localhost:8000/RPC2) >>> INFO:handle_command: pre-filling callee's address with >>> 195.71.4.203:4000 >>> DBUG:doreply: sending reply "4259_9 45238 127.0.0.1 " >>> INFO:process_rtp: session timeout >>> ERR:rtpp_notify_schedule: XMLRPC xmlrpc:http://localhost:8000/RPC2 >>> INFO:remove_session: RTP stats: 0 in from callee, 0 in from caller, 0 >>> relayed, 0 dropped >>> INFO:remove_session: RTCP stats: 0 in from callee, 0 in from caller, 0 >>> relayed, 0 dropped >>> INFO:remove_session: session on ports 45508/45238 is cleaned up >>> ERR:do_timeout_notification: Timeout socket: >>> xmlrpc:http://localhost:8000/RPC2 >>> >>> And it works for me: >>> >>> U 2011/03/16 16:50:14.350721 127.0.0.1:5060 -> 127.0.0.1:15061 BYE >>> sip:2@127.0.0.1:15061;ob SIP/2.0. >>> Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bK06e9.245e2dd7.0. >>> To: sip:2@localhost;tag=5371f039-40d0-4944-aae7-6f75071a2f8c. >>> From: sip:1@localhost;tag=9915df0c-30fc-49c5-aa8a-c86b4242c094. >>> CSeq: 7905 BYE. >>> Call-ID: 56f0f83a-5373-46a1-b6f6-9ce2f93e68d5. >>> Content-Length: 0. >>> User-Agent: kamailio (3.2.0-dev2 (x86_64/linux)). >>> Max-Forwards: 70. >>> . >>> >>> U 2011/03/16 16:50:14.350801 127.0.0.1:5060 -> 127.0.0.1:15060 BYE >>> sip:1@127.0.0.1:15060;transport=UDP;ob SIP/2.0. >>> Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bK06e9.345e2dd7.0. >>> To: sip:1@localhost;tag=9915df0c-30fc-49c5-aa8a-c86b4242c094. >>> From: sip:2@localhost;tag=5371f039-40d0-4944-aae7-6f75071a2f8c. >>> CSeq: 7905 BYE. >>> Call-ID: 56f0f83a-5373-46a1-b6f6-9ce2f93e68d5. >>> Content-Length: 0. >>> User-Agent: kamailio (3.2.0-dev2 (x86_64/linux)). >>> Max-Forwards: 70. >>> . >>> [...] >>> >>> Maybe, you can add some more debug-info from RTP-Proxy... >>> And can you verify, that the RTP-Proxy is not trying to send the request? >>> >>> Kind regards, >>> Carsten >>> >>> 2011/3/16 Alexandre Abreu <alexandre.abreu(a)redt.com.br>: >>>> Hi Carsten, >>>> >>>> Even with your RTPPROXY tarball I was unable to get this working. >>>> Session remains after RTPPROXY timeout. >>>> I am using KAMAILIO 3.1 branch from GIT and as I told you, I moved >>>> the rtpproxy/ from GIT-MASTER to the Kamailio branch (waiting your >> backport). >>> Is >>>> there anything else regarding this feature that should also should be >>> moved >>>> (beyond rtpproxy/)? >>>> >>>> Thanks, >>>> Alexandre >> >> > > > > -- > Carsten Bock > http://www.ng-voice.com > mailto:carsten@ng-voice.com > -- Carsten Bock http://www.ng-voice.com mailto:carsten@ng-voice.com

2 1

Authentication using a PSTN phone
by Miguel Tadeu 23 Mar '11

23 Mar '11

Hi, I have a service that requires a user to authenticate, in order to bill him at the end of the service. I also need real- time check of the balance during a call, in order to signal when the amount is too low to continue the call. Is this possible with SER? Thanx, Miguel Tadeu

2 1

Question about time display in SIREMIS 2.0
by Ricardo Martinez 22 Mar '11

22 Mar '11

Hello. Is there a way to adjust the timezone to display the charts and the logs in the SIREMIS 2.0? I have my linux machine with my local timezone, but the display in the charts and the logs are not in synch with my local time. Thanks in advance. Regards, Ricardo Martinez.-

2 3

ClueCon 2011 Registration Open, Call For Speakers
by Michael Collins 22 Mar '11

22 Mar '11

Greetings fellow open source VoIP users and developers! We are happy to announce that ClueCon 2011 registration is now open. The event will be held August 9-11, 2011 at the Sofitel Hotel in downtown Chicago. Additionally, we are now accepting proposals for speaking topics. If you or your organization would like to speak at ClueCon this year please contact us with your ideas. Visit us at http://www.cluecon.com for additional details on registration, speaking, or sponsorship opportunities. Thank you to all the members of the open source telephony projects who have supported ClueCon over the years. We hope to see you all again this coming August! Michael S Collins ClueCon Team Member http://www.cluecon.com

1 0

Failure route triggers a 500 internal server error
by Miguel Baptista 22 Mar '11

22 Mar '11

Hi all, I am using kamailio 3.1.2 and I have an issue with failure route. Every-time kamailio enters the failure route an internal error is triggered and kamailio sends a 500 SIP error message: /SIP/2.0 500 I'm terribly sorry, server error occurred (1/TM)/. Just for testing purpose I "downgraded" my failure route configuration to a very simple one. But even so I am always getting the same error and it doesn't matter if it is a busy, a cancel, a moved temporarily... If I don't arm the failure route I don't get any error. Any idea what might be wrong? Is this a bug? This is the relevant part of my configuration : */route[RELAY] {/* / if (is_method("INVITE")) {/ / t_on_failure("FAILTWO");/ / }/ / if (!t_relay()) {/ / sl_reply_error();/ / }/ / exit;/ /}/ /*failure_route[FAILTWO] {* xlog("L_INFO", "FAILUREFROMEXTERNAL: Entered\n"); if (t_check_status("302")) { xlog("L_INFO", "FAILUREFROMEXTERNAL: 302 detected\n"); } else { xlog("L_INFO", "FAILUREFROMEXTERNAL: other detected\n"); } xlog("L_INFO", "FAILUREFROMEXTERNAL: Exiting\n");/ } Here is the Debug output from a busy message got by kamailio and the internal error generated: /Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:366]: qm_malloc(0x82bcfe0, 860) called from <core>: receive.c: receive_msg(120) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:406]: qm_malloc(0x82bcfe0, 860) returns address 0x837e300 frag. 0x837e2e8 (size=860) on 1 -th hit Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: <core> [parser/msg_parser.c:640]: SIP Reply (status): Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: <core> [parser/msg_parser.c:642]: version: <SIP/2.0> Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: <core> [parser/msg_parser.c:644]: status: <486> Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: <core> [parser/msg_parser.c:646]: reason: <Busy Here> Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:366]: qm_malloc(0x82bcfe0, 32) called from <core>: parser/msg_parser.c: parse_headers(341) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:406]: qm_malloc(0x82bcfe0, 32) returns address 0x837f29c frag. 0x837f284 (size=32) on 1 -th hit Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:366]: qm_malloc(0x82bcfe0, 116) called from <core>: parser/msg_parser.c: get_hdr_field(130) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:406]: qm_malloc(0x82bcfe0, 116) returns address 0x837e68c frag. 0x837e674 (size=116) on 1 -th hit Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:366]: qm_malloc(0x82bcfe0, 32) called from <core>: parser/parse_via.c: parse_via(2168) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:406]: qm_malloc(0x82bcfe0, 32) returns address 0x837f2ec frag. 0x837f2d4 (size=32) on 1 -th hit Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: <core> [parser/parse_via.c:1287]: Found param type 232, <branch> = <z9hG4bK54f1.2f69b061.0>; state=16 Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: <core> [parser/parse_via.c:2300]: end of header reached, state=5 Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: <core> [parser/msg_parser.c:515]: parse_headers: Via found, flags=2 Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: <core> [parser/msg_parser.c:517]: parse_headers: this is the first via Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: <core> [receive.c:145]: After parse_msg... Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: tm [t_lookup.c:1081]: DEBUG: t_check_msg: msg id=19 global id=18 T start=0xffffffff Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:366]: qm_malloc(0x82bcfe0, 32) called from <core>: parser/msg_parser.c: parse_headers(341) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:406]: qm_malloc(0x82bcfe0, 32) returns address 0x837f33c frag. 0x837f324 (size=32) on 1 -th hit Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:366]: qm_malloc(0x82bcfe0, 116) called from <core>: parser/msg_parser.c: get_hdr_field(130) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:406]: qm_malloc(0x82bcfe0, 116) returns address 0x837e820 frag. 0x837e808 (size=224) on 1 -th hit Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:366]: qm_malloc(0x82bcfe0, 32) called from <core>: parser/parse_via.c: parse_via(2168) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:406]: qm_malloc(0x82bcfe0, 32) returns address 0x837f38c frag. 0x837f374 (size=32) on 1 -th hit Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: <core> [parser/parse_via.c:1287]: Found param type 232, <branch> = <z9hG4bK1f04fb19>; state=6 Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:366]: qm_malloc(0x82bcfe0, 32) called from <core>: parser/parse_via.c: parse_via(2168) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:406]: qm_malloc(0x82bcfe0, 32) returns address 0x837f6e0 frag. 0x837f6c8 (size=32) on 1 -th hit Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: <core> [parser/parse_via.c:1287]: Found param type 235, <rport> = <5060>; state=16 Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: <core> [parser/parse_via.c:2300]: end of header reached, state=5 Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: <core> [parser/msg_parser.c:515]: parse_headers: Via found, flags=22 Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: <core> [parser/msg_parser.c:528]: parse_headers: this is the second via Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:366]: qm_malloc(0x82bcfe0, 32) called from <core>: parser/msg_parser.c: parse_headers(341) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:406]: qm_malloc(0x82bcfe0, 32) returns address 0x837f730 frag. 0x837f718 (size=32) on 1 -th hit Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:366]: qm_malloc(0x82bcfe0, 32) called from <core>: parser/msg_parser.c: parse_headers(341) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:406]: qm_malloc(0x82bcfe0, 32) returns address 0x837f820 frag. 0x837f808 (size=32) on 1 -th hit Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:366]: qm_malloc(0x82bcfe0, 224) called from <core>: parser/msg_parser.c: get_hdr_field(170) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:406]: qm_malloc(0x82bcfe0, 224) returns address 0x837e9c8 frag. 0x837e9b0 (size=224) on 1 -th hit Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:366]: qm_malloc(0x82bcfe0, 24) called from <core>: parser/parse_to.c: parse_to_param(283) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:406]: qm_malloc(0x82bcfe0, 24) returns address 0x837e930 frag. 0x837e918 (size=24) on 1 -th hit Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: <core> [parser/parse_to.c:174]: DEBUG: add_param: tag=i76l2xu48i Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: <core> [parser/parse_to.c:803]: end of header reached, state=29 Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: <core> [parser/msg_parser.c:187]: DEBUG: get_hdr_field: <To> [41]; uri=[sip:+4773411459@umb.no] Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: <core> [parser/msg_parser.c:189]: DEBUG: to body [<sip:+4773411459@umb.no>] Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:366]: qm_malloc(0x82bcfe0, 32) called from <core>: parser/msg_parser.c: parse_headers(341) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:406]: qm_malloc(0x82bcfe0, 32) returns address 0x837f780 frag. 0x837f768 (size=32) on 1 -th hit Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:366]: qm_malloc(0x82bcfe0, 32) called from <core>: parser/msg_parser.c: parse_headers(341) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:406]: qm_malloc(0x82bcfe0, 32) returns address 0x837f7d0 frag. 0x837f7b8 (size=32) on 1 -th hit Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:366]: qm_malloc(0x82bcfe0, 24) called from <core>: parser/msg_parser.c: get_hdr_field(149) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:406]: qm_malloc(0x82bcfe0, 24) returns address 0x837eb78 frag. 0x837eb60 (size=24) on 1 -th hit Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: <core> [parser/msg_parser.c:167]: get_hdr_field: cseq <CSeq>: <102> <INVITE> Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 8005 label 369858290 branch 0 Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: tm [t_lookup.c:1018]: DEBUG: t_reply_matching: reply matched (T=0xb59ec41c)! Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: tm [t_lookup.c:1150]: DEBUG: t_check_msg: msg id=19 global id=19 T end=0xb59ec41c Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: tm [t_reply.c:2014]: DEBUG: reply_received: org. status uas=180, uac[0]=180 local=0 is_invite=1) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:366]: qm_malloc(0xb5881000, 1010) called from tm: t_msgbuilder.c: build_local_reparse(368) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:406]: qm_malloc(0xb5881000, 1012) returns address 0xb59ed5a0 frag. 0xb59ed588 (size=1012) on 1 -th hit Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:428]: qm_free(0xb5881000, 0xb59ed5a0), called from tm: t_reply.c: reply_received(2064) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:450]: qm_free: freeing frag. 0xb59ed588 alloc'ed from tm: t_msgbuilder.c: build_local_reparse(368) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: tm [t_reply.c:1129]: ->>>>>>>>> T_code=180, new_code=486 Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:366]: qm_malloc(0x82bcfe0, 43) called from tm: t_reply.c: fake_req(854) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:406]: qm_malloc(0x82bcfe0, 44) returns address 0x837f8c0 frag. 0x837f8a8 (size=44) on 1 -th hit Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: INFO: <script>: FAILUREFROMEXTERNAL: Entered Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: tm [t_lookup.c:1081]: DEBUG: t_check_msg: msg id=16 global id=16 T start=0xb59ec41c Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: tm [t_lookup.c:1153]: DEBUG: t_check_msg: T already found! Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: tm [tm.c:967]: DEBUG:t_check_status: checked status is <486> Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: INFO: <script>: FAILUREFROMEXTERNAL: other detected Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: INFO: <script>: FAILUREFROMEXTERNAL: Exiting Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:428]: qm_free(0x82bcfe0, 0x837f8c0), called from tm: t_reply.c: free_faked_req(885) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:450]: qm_free: freeing frag. 0x837f8a8 alloc'ed from tm: t_reply.c: fake_req(854) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: <core> [msg_translator.c:204]: check_via_address(158.36.141.13, 158.36.141.13, 0) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:366]: qm_malloc(0x82bcfe0, 12) called from <core>: msg_translator.c: rport_builder(390) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:406]: qm_malloc(0x82bcfe0, 12) returns address 0x837f008 frag. 0x837eff0 (size=12) on 1 -th hit Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:366]: qm_malloc(0x82bcfe0, 392) called from <core>: msg_translator.c: build_res_buf_from_sip_req(2032) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:406]: qm_malloc(0x82bcfe0, 392) returns address 0x838007c frag. 0x8380064 (size=564) on 1 -th hit Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:428]: qm_free(0x82bcfe0, 0x837f008), called from <core>: msg_translator.c: build_res_buf_from_sip_req(2192) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:450]: qm_free: freeing frag. 0x837eff0 alloc'ed from <core>: msg_translator.c: rport_builder(390) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:428]: qm_free(0xb5881000, 0xb59ed29c), called from <core>: mem/shm_mem.c: sh_realloc(82) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:450]: qm_free: freeing frag. 0xb59ed284 alloc'ed from <core>: mem/shm_mem.c: sh_realloc(83) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:366]: qm_malloc(0xb5881000, 391) called from <core>: mem/shm_mem.c: sh_realloc(83) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:406]: qm_malloc(0xb5881000, 392) returns address 0xb59ea720 frag. 0xb59ea708 (size=500) on 1 -th hit Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: tm [t_reply.c:1470]: DEBUG: cleanup_uac_timers: RETR/FR timers reset Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: tm [t_reply.c:659]: DEBUG: reply sent out. buf=0x838007c: SIP/2.0 500 I'm terr..., shmem=0xb59ea720: SIP/2.0 500 I'm terr Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:428]: qm_free(0x82bcfe0, 0x838007c), called from tm: t_reply.c: _reply_light(668) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15015]: <core> [mem/q_malloc.c:366]: qm_malloc(0x82bcfe0, 860) called from <core>: receive.c: receive_msg(120) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:450]: qm_free: freeing frag. 0x8380064 alloc'ed from <core>: msg_translator.c: build_res_buf_from_sip_req(2032) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15015]: <core> [mem/q_malloc.c:406]: qm_malloc(0x82bcfe0, 860) returns address 0x837e300 frag. 0x837e2e8 (size=860) on 1 -th hit Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: tm [t_reply.c:669]: DEBUG: _reply_light: finished Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15015]: DEBUG: <core> [parser/msg_parser.c:630]: SIP Request: Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: tm [t_reply.c:1631]: DEBUG: relay_reply: branch=0, save=0, relay=-1 Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15015]: DEBUG: <core> [parser/msg_parser.c:632]: method: <ACK> Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: tm [t_reply.c:1470]: DEBUG: cleanup_uac_timers: RETR/FR timers reset Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15015]: DEBUG: <core> [parser/msg_parser.c:634]: uri: <sip:+4773411459@umb.no> Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: <core> [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15015]: DEBUG: <core> [parser/msg_parser.c:636]: version: <SIP/2.0> Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: <core> [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15015]: <core> [mem/q_malloc.c:366]: qm_malloc(0x82bcfe0, 32) called from <core>: parser/msg_parser.c: parse_headers(341) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: <core> [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15015]: <core> [mem/q_malloc.c:406]: qm_malloc(0x82bcfe0, 32) returns address 0x837ed00 frag. 0x837ece8 (size=32) on 1 -th hit Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: <core> [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15015]: <core> [mem/q_malloc.c:366]: qm_malloc(0x82bcfe0, 116) called from <core>: parser/msg_parser.c: get_hdr_field(130) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: <core> [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15015]: <core> [mem/q_malloc.c:406]: qm_malloc(0x82bcfe0, 116) returns address 0x837e68c frag. 0x837e674 (size=116) on 1 -th hit Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: <core> [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15015]: <core> [mem/q_malloc.c:366]: qm_malloc(0x82bcfe0, 32) called from <core>: parser/parse_via.c: parse_via(2168) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: <core> [receive.c:289]: receive_msg: cleaning up Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15015]: <core> [mem/q_malloc.c:406]: qm_malloc(0x82bcfe0, 32) returns address 0x837ecb0 frag. 0x837ec98 (size=32) on 1 -th hit Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:428]: qm_free(0x82bcfe0, 0x837f2ec), called from <core>: parser/parse_via.c: free_via_param_list(2387) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15015]: DEBUG: <core> [parser/parse_via.c:1287]: Found param type 232, <branch> = <z9hG4bK1f04fb19>; state=6 Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:450]: qm_free: freeing frag. 0x837f2d4 alloc'ed from <core>: parser/parse_via.c: parse_via(2168) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15015]: <core> [mem/q_malloc.c:366]: qm_malloc(0x82bcfe0, 32) called from <core>: parser/parse_via.c: parse_via(2168) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:428]: qm_free(0x82bcfe0, 0x837e68c), called from <core>: parser/parse_via.c: free_via_list(2399) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15015]: <core> [mem/q_malloc.c:406]: qm_malloc(0x82bcfe0, 32) returns address 0x837ec60 frag. 0x837ec48 (size=32) on 1 -th hit Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:450]: qm_free: freeing frag. 0x837e674 alloc'ed from <core>: parser/msg_parser.c: get_hdr_field(130) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15015]: DEBUG: <core> [parser/parse_via.c:1287]: Found param type 235, <rport> = <n/a>; state=17 Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:428]: qm_free(0x82bcfe0, 0x837f29c), called from <core>: parser/hf.c: free_hdr_field_lst(251) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15014]: <core> [mem/q_malloc.c:366]: qm_malloc(0x82bcfe0, 860) called from <core>: receive.c: receive_msg(120) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15015]: DEBUG: <core> [parser/parse_via.c:2300]: end of header reached, state=5 Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:450]: qm_free: freeing frag. 0x837f284 alloc'ed from <core>: parser/msg_parser.c: parse_headers(341) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15014]: <core> [mem/q_malloc.c:406]: qm_malloc(0x82bcfe0, 860) returns address 0x8380078 frag. 0x8380060 (size=860) on 1 -th hit Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15015]: DEBUG: <core> [parser/msg_parser.c:515]: parse_headers: Via found, flags=2 Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:428]: qm_free(0x82bcfe0, 0x837f38c), called from <core>: parser/parse_via.c: free_via_param_list(2387) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15014]: DEBUG: <core> [parser/msg_parser.c:640]: SIP Reply (status): Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15015]: DEBUG: <core> [parser/msg_parser.c:517]: parse_headers: this is the first via Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:450]: qm_free: freeing frag. 0x837f374 alloc'ed from <core>: parser/parse_via.c: parse_via(2168) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15014]: DEBUG: <core> [parser/msg_parser.c:642]: version: <SIP/2.0> Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15015]: DEBUG: <core> [receive.c:145]: After parse_msg... Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:428]: qm_free(0x82bcfe0, 0x837f6e0), called from <core>: parser/parse_via.c: free_via_param_list(2387) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15014]: DEBUG: <core> [parser/msg_parser.c:644]: status: <486> Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15015]: DEBUG: <core> [receive.c:186]: preparing to run routing scripts... Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:450]: qm_free: freeing frag. 0x837f6c8 alloc'ed from <core>: parser/parse_via.c: parse_via(2168) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15014]: DEBUG: <core> [parser/msg_parser.c:646]: reason: <Busy Here> Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15015]: DEBUG: sl [sl_funcs.c:307]: DEBUG : sl_filter_ACK: to late to be a local ACK! / Regards, Miguel Baptista <javascript:void(0);>

2 1

RES: event_route and acc_db_request().
by Alexandre Abreu 22 Mar '11

22 Mar '11

Hello. I have found another important thread "Catching internally generated BYE event & other" regarding this goal. Alex, Do you mind to share how did you catch the internally generated BYE? (by just reading the thread I can't know if you really did) As far as I understand, the event_route could be used for this (accounting) but specially acc_db_request() does not seem to work. Maybe I am missing something basic or there is other approach to reach this target. Thanks for any tip, Alexandre. De: Alexandre Abreu [mailto:alexandre.abreu@redt.com.br] Enviada em: quinta-feira, 17 de março de 2011 13:45 Para: sr-users(a)lists.sip-router.org Assunto: event_route and acc_db_request(). Hello. Why acc_db_request() doesnt work on event_route? Mar 17 13:15:44 devel kamailio[25209]: INFO: <script>: Routing locally generated BYE to <sip:2000000@192.168.200.114:9297> Mar 17 13:15:44 devel kamailio[25209]: ERROR: <core> [db.c:421]: invalid parameter value Mar 17 13:15:44 devel kamailio[25209]: ERROR: acc [acc.c:391]: error in use_table Mar 17 13:15:44 devel kamailio[25209]: INFO: <script>: Routing locally generated BYE to <sip:2000001@192.168.200.149:7335> Mar 17 13:15:44 devel kamailio[25209]: ERROR: <core> [db.c:421]: invalid parameter value Mar 17 13:15:44 devel kamailio[25209]: ERROR: acc [acc.c:391]: error in use_table event_route[tm:local-request] { xlog("L_INFO", "Routing locally generated $rm to <$ru>\n"); if (is_method("BYE")) acc_db_request("rtp-timeout", acc); } If I change acc_db_request() to acc_log_request() everything works fine, but this BYE should go to database for accounting purposes. I am using GIT version from Kamailio branch 3.1. >From 2008 there is a thread that also demonstrate this problem: http://www.mail-archive.com/users@lists.kamailio.org/msg01411.html Unfortunately, in the archives, theres no solution for that. Regards, Alexandre

1 0

RES: RTPPROXY timeout patch.
by Alexandre Abreu 22 Mar '11

22 Mar '11

Hi Carsten, After more testing, I just realized that the notification works only if I use "-f" parameter in RTPPROXY. If I omit this parameter (to run in background mode), the session timeout but there's no notification sent (BYE). Why's that? Thanks, Alexandre. -----Mensagem original----- De: Alexandre Abreu [mailto:alexandre.abreu@redt.com.br] Enviada em: quarta-feira, 16 de março de 2011 14:41 Para: 'Carsten Bock' Cc: 'sr-users(a)lists.sip-router.org' Assunto: RES: RTPPROXY timeout patch. Carsten, Here it goes: [root@devel rtpproxy-carsten]# ./rtpproxy -T 10 -f -F -i -l 192.168.200.90 -s udp:*:7722 -d DBUG ERR INFO -n tcp:127.0.0.1:7723 INFO:main: rtpproxy started, pid 22495 rtpproxy: >>> Running Timeout-Process DBUG:handle_command: received command "22428_8 Uc0,8,101 080b5d23d1603667 192.168.200.114 6380 0073852a;1" INFO:handle_command: new session 080b5d23d1603667, tag 0073852a;1 requested, type strong INFO:handle_command: new session on a port 48662 created, tag 0073852a;1 INFO:handle_command: pre-filling caller's address with 192.168.200.114:6380 DBUG:doreply: sending reply "22428_8 48662 192.168.200.90" DBUG:handle_command: received command "22436_8 Lc0,8,101 080b5d23d1603667 192.168.200.149 7614 0073852a;1 c758967a;1 xmlrpc:http://127.0.0.1:8000/RPC2" INFO:handle_command: lookup on ports 48662/58604, session timer restarted INFO:handle_command: setting custom timeout handler (xmlrpc:http://127.0.0.1:8000/RPC2) INFO:handle_command: pre-filling callee's address with 192.168.200.149:7614 DBUG:doreply: sending reply "22436_8 58604 192.168.200.90" INFO:process_rtp: session timeout INFO:remove_session: RTP stats: 982 in from callee, 4 in from caller, 986 relayed, 0 dropped INFO:remove_session: RTCP stats: 5 in from callee, 1 in from caller, 6 relayed, 0 dropped INFO:remove_session: session on ports 48662/58604 is cleaned up ERR:do_timeout_notification: Timeout socket is: èîÈÄÐÊ@ÍÈÄÈÄ`ê DBUG:reconnect_timeout_handler: reconnecting timeout socket ERR:reconnect_timeout_handler: can't create timeout socket: Address family not supported by protocol ERR:do_timeout_notification: unable to send timeout notification Very weird chars in the debug of what Timeout socket is... But the custom timeout handler are being printed correctly from the config file. INFO:handle_command: setting custom timeout handler (xmlrpc:http://127.0.0.1:8000/RPC2) Here I am running CentOS 5.2 32-bit. Thanks, Alexandre -----Mensagem original----- De: kaiserbock2(a)googlemail.com [mailto:kaiserbock2@googlemail.com] Em nome de Carsten Bock Enviada em: quarta-feira, 16 de março de 2011 14:18 Para: Alexandre Abreu Cc: sr-users(a)lists.sip-router.org Assunto: Re: RTPPROXY timeout patch. Hi Alexandre, My version of RTP-Proxy is following: bock@bock-tde:~/ims/rtpproxy$ ./rtpproxy -v Basic version: 20040107 Extension 20050322: Support for multiple RTP streams and MOH Extension 20060704: Support for extra parameter in the V command Extension 20071116: Support for RTP re-packetization Extension 20071218: Support for forking (copying) RTP stream Extension 20080403: Support for RTP statistics querying Extension 20081102: Support for setting codecs in the update/lookup command Extension 20081224: Support for session timeout notifications Extension 20090810: Support for automatic bridging Extension 20100819: Support for timeout notifications using XML-RPC towards Kamailio/sip-router.org Please find attached a modified rtpp_notify.c file. I have just added some tiny debug output in order to see some points. Now you should see the following Debug-Outputs: rtpproxy: >>> Running Timeout-Process Then the notifier process is running. That would be good. If not, that's the reason why it is not working. When the request comes in, you should see the following: INFO:handle_command: setting custom timeout handler (xmlrpc:http://localhost:8000/RPC2) Then the Timeout-Socket was properly set, that would be good as well. Now the timeout: INFO:process_rtp: session timeout [...] ERR:do_timeout_notification: Timeout socket is: xmlrpc:http://localhost:8000/RPC2 That would be great, because then the Timeout towards the Kamailio should be triggerd. If these parts are ok, then there must be some issue either in the XML-RPC client library or in the communication between the RTP-Proxy and Kamailio. I hope you did a trace on the XML-RPC Port both on the RTPproxy and on the Kamailio? What distro are you using? My tests were only on Ubuntu and Debian, which are quite similar. Hope we find this issue, Kind regards, Carsten P.S.: I think i removed that check for the port for testing, that's why my version accepted the socket without port... (now i'm using "-n tcp:127.0.0.1:9999") 2011/3/16 Alexandre Abreu <alexandre.abreu(a)redt.com.br>: > Carsten, > > Indeed. Very strange. > > Are we running the same RTPPROXY version? How can you start using '-n > tcp:127.0.0.1' without specifying a port? > > [root@devel rtpproxy-carsten]# ./rtpproxy -T 10 -f -F -i -l > 192.168.200.90 -s udp:*:7722 -d DBUG ERR INFO -n tcp:127.0.0.1 > rtpproxy: can't parse host:port in TCP address > rtpproxy: can't start notification thread > > [root@devel rtpproxy-carsten]# ./rtpproxy -T 10 -f -F -i -l > 192.168.200.90 -s udp:*:7722 -d DBUG ERR INFO -n tcp:127.0.0.1:7723 > INFO:main: rtpproxy started, pid 21169 > > DBUG:handle_command: received command "17828_9 Uc0,8,101 > 4b10ce04de4f8818 > 192.168.200.114 6380 9c4b6265;1" > INFO:handle_command: new session 4b10ce04de4f8818, tag 9c4b6265;1 > requested, type strong > INFO:handle_command: new session on a port 43750 created, tag > 9c4b6265;1 > INFO:handle_command: pre-filling caller's address with > 192.168.200.114:6380 > DBUG:doreply: sending reply "17828_9 43750 192.168.200.90 " > DBUG:handle_command: received command "17847_9 Lc0,8,101 > 4b10ce04de4f8818 > 192.168.200.149 7386 9c4b6265;1 dd69ab1d;1 > xmlrpc:http://localhost:8000/RPC2" > INFO:handle_command: lookup on ports 43750/55796, session timer > restarted > INFO:handle_command: setting custom timeout handler > (xmlrpc:http://localhost:8000/RPC2) > INFO:handle_command: pre-filling callee's address with > 192.168.200.149:7386 > DBUG:doreply: sending reply "17847_9 55796 192.168.200.90 " > INFO:process_rtp: session timeout > INFO:remove_session: RTP stats: 548 in from callee, 5 in from caller, > 553 relayed, 0 dropped > INFO:remove_session: RTCP stats: 3 in from callee, 1 in from caller, 4 > relayed, 0 dropped > INFO:remove_session: session on ports 43750/55796 is cleaned up > DBUG:reconnect_timeout_handler: reconnecting timeout socket > ERR:reconnect_timeout_handler: can't create timeout socket: Address > family not supported by protocol > ERR:do_timeout_notification: unable to send timeout notification > > Above the error message. > > [root@devel ~]# md5sum rtpproxy-carsten.tar.gz > c02b1e2ac57d39562e86bcfc4ee592b8 rtpproxy-carsten.tar.gz > > Thanks, > Alexandre. > > -----Mensagem original----- > De: kaiserbock2(a)googlemail.com [mailto:kaiserbock2@googlemail.com] Em > nome de Carsten Bock Enviada em: quarta-feira, 16 de março de 2011 > 13:03 > Para: Alexandre Abreu > Cc: sr-users(a)lists.sip-router.org > Assunto: Re: RTPPROXY timeout patch. > > Hi Alexandre, > > That is strange: > > I run the RTP-Proxy like this (directly from the TAR-File, i sent you) > and Kamailio with attached config-file. > > bock@bock-tde:~/ims/rtpproxy$ sudo ./rtpproxy -T 10 -f -F -i -l > 127.0.0.1 -s udp:*:22222 -d DBUG -n tcp:127.0.0.1 > rtpproxy: Timer started. > INFO:main: rtpproxy started, pid 4203 > [... Kamailio connects to RTP-Proxy...] > DBUG:handle_command: received command "4259_8 > UAc98,97,99,104,3,0,8,9,96 56f0f83a-5373-46a1-b6f6-9ce2f93e68d5 > 195.71.4.203 4008 5371f039-40d0-4944-aae7-6f75071a2f8c;1" > INFO:handle_command: new session 56f0f83a-5373-46a1-b6f6-9ce2f93e68d5, > tag 5371f039-40d0-4944-aae7-6f75071a2f8c;1 requested, type strong > INFO:handle_command: new session on a port 45508 created, tag > 5371f039-40d0-4944-aae7-6f75071a2f8c;1 > INFO:handle_command: pre-filling caller's address with > 195.71.4.203:4008 > DBUG:doreply: sending reply "4259_8 45508 127.0.0.1 " > DBUG:handle_command: received command "4259_9 LAc98,96 > 56f0f83a-5373-46a1-b6f6-9ce2f93e68d5 195.71.4.203 4000 > 5371f039-40d0-4944-aae7-6f75071a2f8c;1 > 9915df0c-30fc-49c5-aa8a-c86b4242c094;1 > xmlrpc:http://localhost:8000/RPC2" > INFO:handle_command: lookup on ports 45508/45238, session timer > restarted > INFO:handle_command: setting custom timeout handler > (xmlrpc:http://localhost:8000/RPC2) > INFO:handle_command: pre-filling callee's address with > 195.71.4.203:4000 > DBUG:doreply: sending reply "4259_9 45238 127.0.0.1 " > INFO:process_rtp: session timeout > ERR:rtpp_notify_schedule: XMLRPC xmlrpc:http://localhost:8000/RPC2 > INFO:remove_session: RTP stats: 0 in from callee, 0 in from caller, 0 > relayed, 0 dropped > INFO:remove_session: RTCP stats: 0 in from callee, 0 in from caller, 0 > relayed, 0 dropped > INFO:remove_session: session on ports 45508/45238 is cleaned up > ERR:do_timeout_notification: Timeout socket: > xmlrpc:http://localhost:8000/RPC2 > > And it works for me: > > U 2011/03/16 16:50:14.350721 127.0.0.1:5060 -> 127.0.0.1:15061 BYE > sip:2@127.0.0.1:15061;ob SIP/2.0. > Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bK06e9.245e2dd7.0. > To: sip:2@localhost;tag=5371f039-40d0-4944-aae7-6f75071a2f8c. > From: sip:1@localhost;tag=9915df0c-30fc-49c5-aa8a-c86b4242c094. > CSeq: 7905 BYE. > Call-ID: 56f0f83a-5373-46a1-b6f6-9ce2f93e68d5. > Content-Length: 0. > User-Agent: kamailio (3.2.0-dev2 (x86_64/linux)). > Max-Forwards: 70. > . > > U 2011/03/16 16:50:14.350801 127.0.0.1:5060 -> 127.0.0.1:15060 BYE > sip:1@127.0.0.1:15060;transport=UDP;ob SIP/2.0. > Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bK06e9.345e2dd7.0. > To: sip:1@localhost;tag=9915df0c-30fc-49c5-aa8a-c86b4242c094. > From: sip:2@localhost;tag=5371f039-40d0-4944-aae7-6f75071a2f8c. > CSeq: 7905 BYE. > Call-ID: 56f0f83a-5373-46a1-b6f6-9ce2f93e68d5. > Content-Length: 0. > User-Agent: kamailio (3.2.0-dev2 (x86_64/linux)). > Max-Forwards: 70. > . > [...] > > Maybe, you can add some more debug-info from RTP-Proxy... > And can you verify, that the RTP-Proxy is not trying to send the request? > > Kind regards, > Carsten > > 2011/3/16 Alexandre Abreu <alexandre.abreu(a)redt.com.br>: >> Hi Carsten, >> >> Even with your RTPPROXY tarball I was unable to get this working. >> Session remains after RTPPROXY timeout. >> I am using KAMAILIO 3.1 branch from GIT and as I told you, I moved >> the rtpproxy/ from GIT-MASTER to the Kamailio branch (waiting your backport). > Is >> there anything else regarding this feature that should also should be > moved >> (beyond rtpproxy/)? >> >> Thanks, >> Alexandre

2 2

$rU isn't used in t_relay() in failure_route
by Steven Wheeler 21 Mar '11

21 Mar '11

I am attempting to put together code which will allow us to redirect a call to a different tn if the route failed. Example: User dials 11235551234 Kamailio uses LCR to route to 192.168.0.100 (404 error) Change $rU to 11235550000 in failure_route The only way I can get $rU to show up in the SIP message is to call append_branch() which is not a solution because I now end up with two SIP messages one that is correct and one that is incorrect. Here is the failure_route I am using. Any input would be appreciated. failure_route[ECRC] { if (t_is_canceled()) { xlog("L_INFO", "Call has been canceled in 'ECRC'."); exit; } xlog("L_INFO", "SIP Response Code: $T_reply_code"); # We should fail over if we received one of these response codes. if (!t_check_status("[45][0-9][0-9]")) { xlog("L_INFO", "Not a failover error."); exit; } # Translate the dialed number into the ECRC DID. dp_translate("1", "$rU/$rU"); # Load the gateways for normal dialing. xlog("L_INFO", "Trying to find LCR route for $rU"); if (! load_gws("1", "$rU")) { xlog("Unable to load routing"); t_reply("503", "Unable to load routing"); exit; } xlog("L_INFO", "Trying gateway '$avp(i:709)'"); if (!next_gw()) { xlog("L_INFO", "No LCR route for $rU"); t_reply("404", "No available gateways"); exit; } $rd=$dd; $rp=$dp; $du=$ru; xlog("L_INFO", "ru: $ru"); xlog("L_INFO", "du: $du"); #at this point $ru and $du contain the correct uri. append_branch(); # Send out the request. xlog("L_INFO", "Now we're sending the request out."); if (!t_relay()) { xlog("L_INFO" "Relay failed!"); t_reply("502", "Bad Gateway"); exit; } } Now I have two SIP messages one containing the correct phone number: INVITE sip:11235550000@xxx.xxx.xxx.xxx SIP/2.0 Record-Route: <sip:xxx.xxx.xxx.xxx;lr=on> Via: SIP/2.0/UDP xxx.xxx.xxx.xxx;branch=z9hG4bK2e2f.4145be65.2 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK28aa7440;rport=5060 Max-Forwards: 69 From: "Steven Wheeler" <sip:6124243265@xxx.xxx.xxx.xxx>;tag=as30b3d6f5 To: <sip:11235551234@xxx.xxx.xxx.xxx:5060> ... And one containing the wrong phone number: INVITE sip:11235551234@xxx.xxx.xxx.xxx SIP/2.0 Record-Route: <sip:xxx.xxx.xxx.xxx;lr=on> Via: SIP/2.0/UDP xxx.xxx.xxx.xxx;branch=z9hG4bK2e2f.4145be65.3 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK28aa7440;rport=5060 Max-Forwards: 69 From: "Steven Wheeler" <sip:6124243265@xxx.xxx.xxx.xxx>;tag=as30b3d6f5 To: <sip:11235551234@xxx.xxx.xxx.xxx:5060> ... Steven Wheeler swheeler(a)usinternet.com 952-253-3252

3 2

Invitation to connect on LinkedIn
by Andrey Kuprianov via LinkedIn 21 Mar '11

21 Mar '11

LinkedIn ------------Andrey Kuprianov requested to add you as a connection on LinkedIn: ------------------------------------------ Luis, I'd like to add you to my professional network on LinkedIn. - Andrey Accept invitation from Andrey Kuprianov http://www.linkedin.com/e/-y8k3ym-gljdl3wf-1g/XkppUi8mK5TI9VJsZIpnfBovln4Su… View invitation from Andrey Kuprianov http://www.linkedin.com/e/-y8k3ym-gljdl3wf-1g/XkppUi8mK5TI9VJsZIpnfBovln4Su… ------------------------------------------ Why might connecting with Andrey Kuprianov be a good idea? Have a question? Andrey Kuprianov's network will probably have an answer: You can use LinkedIn Answers to distribute your professional questions to Andrey Kuprianov and your extended network. You can get high-quality answers from experienced professionals. http://www.linkedin.com/e/-y8k3ym-gljdl3wf-1g/ash/inv19_ayn/ -- (c) 2011, LinkedIn Corporation

1 0

problem with ser 0.9.6
by Sabatella, Michael 19 Mar '11

19 Mar '11

All, I am a long time lurker but never posted. I am running ser 0.9.6 for many years with no problems. Just recently none of my users can register. Well - they register once then expire out. I noticed that several ser processes are cpu- bound consuming all available cycles. Once they drop register they can never reregister. I restored the ser DB to an older one that was saved when the system was working properly. Any ideas why the ser processes would be CPU bound? Nothing has been changed on the ser proxy in years and the system has been super stable for 2 years. Has anyone seen this behavior in the past? This is from top: PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND 4466 root 25 0 39960 1948 1368 R 35.8 0.0 1:43.97 ser 4477 root 25 0 39960 2028 1448 R 34.5 0.1 0:21.85 ser 4471 root 25 0 39960 2040 1456 R 32.8 0.1 0:47.63 ser 4475 root 25 0 39960 1940 1360 R 32.5 0.0 2:08.90 ser 4470 root 25 0 39960 1916 1336 R 32.2 0.0 1:09.92 ser 4480 root 25 0 39960 1828 1248 R 30.5 0.0 4:45.16 ser Any ideas would be greatly apreciated. Michael See what's next from IPC www.UnigyfromIPC.com<http://www.UnigyfromIPC.com> [cid:unigy-logo-compr64b7.png]<http://www.unigyfromipc.com/> Join the Conversation: LinkedIn<http://www.linkedin.com/groups?mostPopular=&gid=3808474> * Twitter<http://twitter.com/IPC_Systems_Inc> [cid:tree5326.png]Please consider the environment before printing this email. ________________________________ DISCLAIMER: This e-mail may contain information that is confidential, privileged or otherwise protected from disclosure. If you are not an intended recipient of this e-mail, do not duplicate or redistribute it by any means. Please delete it and any attachments and notify the sender that you have received it in error. Unintended recipients are prohibited from taking action on the basis of information in this e-mail.E-mail messages may contain computer viruses or other defects, may not be accurately replicated on other systems, or may be intercepted, deleted or interfered with without the knowledge of the sender or the intended recipient. If you are not comfortable with the risks associated with e-mail messages, you may decide not to use e-mail to communicate with IPC. IPC reserves the right, to the extent and under circumstances permitted by applicable law, to retain, monitor and intercept e-mail messages to and from its systems.

4 7

Re: [SR-Users] $rU isn't used in t_relay() in failure_route
by Steven Wheeler 18 Mar '11

18 Mar '11

Thanks for the quick response, I actually just ended up moving to a newer version of Kamailio (v3.1.2) and that fixed my problem. Steven Wheeler swheeler(a)usinternet.com 952-253-3252 -----Original Message----- From: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of Klaus Darilion Sent: Friday, March 18, 2011 1:00 PM To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - UsersMailing List Cc: Steven Wheeler Subject: Re: [SR-Users] $rU isn't used in t_relay() in failure_route Am 16.03.2011 21:09, schrieb Steven Wheeler: > $rd=$dd; > $rp=$dp; > $du=$ru; This one I do not understand. Also I do not see the code where you change $dU? Anyway, it seems that 2 branches are added: maybe one by lcr module internally via next_gw and one manually? You can try to change $dU before calling next_gw and omit the append_branch() call. regards klaus _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

1 0

problem with AAAA request when dns query for ipv6 is disabled
by CT Radu 18 Mar '11

18 Mar '11

Did any one had any problem with this? I have a test setup where I have the following directives for dns caching: use_dns_cache=yes #enabling DNS caching dns_cache_del_nonexp=yes # deletes entries if the memory is becoming full dns_cache_flags=0 dns_cache_gc_interval=120 dns_cache_init=on dns_cache_max_ttl=3600 dns_try_ipv6=no # yes if you want the lookup to be for IPv6 also dns_try_naptr=yes despite the fact that I have *dns_try_ipv6=no* kamailio sends a AAAA request to the DNS server. This is an extra request in a case where it seems not to be needed. Also, it seems that a NAPTR request is not sent - I need to investigate more, being just a hunch for now. All the best Radu Costin

1 0

SPCE beyond 50 000 subscribers and 2000 concurrent calls
by Deon Vermeulen 18 Mar '11

18 Mar '11

Hi Guys Can the SPCE platform scale beyond 50 000 subscribers and 2000 concurrent calls when adding more hardware, i.e CPU and RAM? rgds Deon

2 2

problem unreferencing dialog in dialog module
by Anton Roman 18 Mar '11

18 Mar '11

Hi all, we are running Kamailio 3.1.2 in a production environment, using the dialog module, and it crashed two hours ago. Here you have the logs we got (addtional log fragments with the acc records involved in this call are appended at the end of the mail): Mar 2 14:43:05 kamailio2 /usr/local/sbin/kamailio[28927]: CRITICAL: dialog [dlg_hash.c:599]: bogus ref -1 with cnt 1 for dlg 0x7f23f472db30 [2490:1070436595] with clid 'e0a20cb844d211e0acd8001422093865@<CLIENT IP>' and tags '1577886432-3759264324-335599788-1698171170' '' Mar 2 14:43:05 kamailio2 /usr/local/sbin/kamailio[28927]: : <core> [mem/q_malloc.c:446]: BUG: qm_free: freeing already freed pointer, first free: dialog: dlg_cb.c: destroy_dlg_callbacks_list(80) - aborting Mar 2 14:43:05 kamailio2 /usr/local/sbin/kamailio[28896]: ALERT: <core> [main.c:741]: child process 28927 exited by a signal 6 Mar 2 14:43:05 kamailio2 /usr/local/sbin/kamailio[28896]: ALERT: <core> [main.c:744]: core was not generated Mar 2 14:43:05 kamailio2 /usr/local/sbin/kamailio[28896]: INFO: <core> [main.c:756]: INFO: terminating due to SIGCHLD Mar 2 14:43:05 kamailio2 /usr/local/sbin/kamailio[28948]: INFO: <core> [main.c:807]: INFO: signal 15 received Mar 2 14:43:05 kamailio2 /usr/local/sbin/kamailio[28942]: INFO: <core> [main.c:807]: INFO: signal 15 received We get the kamailio code from git last week: sercmd> core.info { version: kamailio 3.1.2 id: 4ace86 compiler: gcc 4.3.2 compiled: 09:12:36 Feb 23 2011 flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES } The problem looks like this other one already fixed: http://lists.sip-router.org/pipermail/sr-users/2009-November/027351.html We set the Kamailio to debug level in case it happens again. On the other side, I need to know why the core is not been generated. I have already checked the points mentioned in http://www.kamailio.org/dokuwiki/doku.php/troubleshooting:corefiles 1. disable_core_dump is not set in the config file. 2. From /etc/default/kamailio: ... DUMP_CORE=yes ... 2. From /etc/init.d/kamailio: ... if test "$DUMP_CORE" = "yes" ; then # set proper ulimit ulimit -c unlimited # directory for the core dump files COREDIR=/home/corefiles [ -d $COREDIR ] || mkdir $COREDIR chmod 777 $COREDIR echo "$COREDIR/core.%e.sig%s.%p" > /proc/sys/kernel/core_pattern fi ... 4. Writting permissions of $COREDIR ls -hall /home ... drwxrwxrwx 2 root root 4.0K 2010-12-21 09:15 corefiles ... What else should I check? Thanks in advance, regards Antón *Acc records related to the dialog whose destruction causes the problem:* Mar 2 14:42:44 kamailio2 /usr/local/sbin/kamailio[28902]: NOTICE: acc [acc.c:275]: ACC: transaction answered: timestamp=1299073364;method=INVITE;from_tag=1577886432-3759264324-335599788-1698171170;to_tag=5FFAEA34-6A;call_id=e0a20cb844d211e0acd8001422093865@<client IP>;code=200;reason=OK;src_user=<caller number>;src_domain=<client IP>;dst_ouser=<called number>;dst_user=<called number>;dst_domain=10.90.1.251;src_ip=<client IP> ... Mar 2 14:42:44 kamailio2 /usr/local/sbin/kamailio[28920]: NOTICE: acc [acc.c:275]: ACC: request acknowledged: timestamp=1299073364;method=ACK;from_tag=1577886432-3759264324-335599788-1698171170;to_tag=5FFAEA34-6A;call_id=e0a20cb844d211e0acd8001422093865@<client IP>;code=200;reason=OK;src_user=<caller number>;src_domain=<client IP>;dst_ouser=<called number>;dst_user=<called number>;dst_domain=10.90.1.251;src_ip=<client IP> ... Mar 2 14:43:00 kamailio2 /usr/local/sbin/kamailio[28903]: ERROR: <script>: ACK WITHOUT MATCHING TRANSACTION in e0a20cb844d211e0acd8001422093865@<client IP> call... ignore and discard. ... Mar 2 14:43:00 kamailio2 /usr/local/sbin/kamailio[28904]: NOTICE: acc [acc.c:275]: ACC: transaction answered: timestamp=1299073380;method=BYE;from_tag=1577886432-3759264324-335599788-1698171170;to_tag=5FFAEA34-6A;call_id=e0a20cb844d211e0acd8001422093865@<client IP>;code=200;reason=OK;src_user=<caller number>;src_domain=<client IP>;dst_ouser=<called number>;dst_user=<called number>;dst_domain=10.90.1.251;src_ip=<client IP>

3 8

NAT Traversal
by Spinov Evgeniy 18 Mar '11

18 Mar '11

May be I miss some important details? No suggestions? Thank you. > Hello, all. > Using nathelper + rtpproxy for subj. Kamailio has public and private > network interfaces. Asterisk is only private. RTP Proxy is working in > bridge mode and relaying traffic from UAC to Asterisks. > Everything is working fine, except one configuration. When the client is > behind router ( a specific one, I do not have an access there to > check ), and this UAC is making a call to other public extension, which > is behind router, then RTP Proxy is relaying traffic to the caller, > using another UDP port, then the packets arrive. > For instance: > UAC 1 -> UAC 2 > PUBLIC_IP:10 > KAMAILIO_IP:5555 > KAMAILIO_IP:5678 > PUBLIC_IP:12 > While for the UAC 2 it looks like: > PUBLIC_IP:20 > KAMAILIO_IP:6767 > KAMAILIO_IP:4564 > PUBLIC_IP:20 > The source and destination UDP ports are the same. As result, I can hear > UAC 1 and he cannot hear me. > In case of we have UAC 3, which is behind other router, call is working > fine with same configuration. > "It's routers fault" you can say, but in the same configuration ( I mean > network, not kamailio ) it worked, but when RTPProxy was not in bridge > mode and Kamailio and Asterisks were in public network. Reinvites are > not allowed in both cases. > The question is, why the source and destination UDP ports are different? > Using STUN in first case, cause without it, private IP written in > contacts and as result, traffic relayed from Kamailio is incorrect, > cause heading to private network which is unreachable. > Any ideas where to dig?

2 5

TM module fr_timer
by Geoffrey Mina 17 Mar '11

17 Mar '11

Hello, I am just curious because I have seen a couple of references to the fr_timer being in SECONDS and some look like they are in MILLISECONDS. In 1.5.X, what is this value? Also - will fr_timer control the retransmission of an INVITE if no 100 Trying is received? Right now it appears my configuration is waiting 500ms to retransmit the INVITE when I don't get a 100 trying. This is what my config looks like right now and this is the behavior I'm seeing. modparam("tm","fr_timer",15) modparam("tm","fr_inv_timer",120) modparam("tm","fr_inv_timer_avp", "$avp(i:704)") INITIAL INVITE [No response in 500ms] RETRANS INVITE [NO response in 1000ms] RETRANS INVITE I'm having a hard time correlating that what I have in my config. Thanks, Geoff

2 1

kamailio with module_s/domain modul
by MÉSZÁROS Mihály 17 Mar '11

17 Mar '11

Hi, Is it possible to use kamailio with module_s/domain.so modul? Could be any known problem occur if i want to use kamailio deb packages with module_s/domain.so? Kamailio has an up2date debian packages unlike ser, but i need modul_s/domain.so modul functionality to manage multiple domain. Thanks, Misi

2 1

RES: RTPPROXY timeout patch.
by Alexandre Abreu 17 Mar '11

17 Mar '11

Hello Carsten. Still no lucky. I just moved the whole rtpproxy/ directory into 3.1 branch and recompiled and then changed my script to use rtpproxy_offer()/rtpproxy_answer(). Look: Mar 15 14:04:38 devel rtpproxy[29111]: INFO:handle_command: new session e2361a59b4588d51, tag ed382a18;1 requested, type strong Mar 15 14:04:38 devel rtpproxy[29111]: INFO:handle_command: new session on a port 44206 created, tag ed382a18;1 Mar 15 14:04:38 devel rtpproxy[29111]: INFO:handle_command: pre-filling caller's address with 192.168.200.114:6380 Mar 15 14:04:39 devel kamailio[29544]: NOTICE: acc [acc.c:275]: ACC: transaction answered: timestamp=1300208679;method=INVITE;from_tag=ed382a18;to_tag=19477c42;call_id =e2361a59b4588d51;code=200;reason=OK; Mar 15 14:04:39 devel rtpproxy[29111]: INFO:handle_command: lookup on ports 44206/41680, session timer restarted Mar 15 14:04:39 devel rtpproxy[29111]: INFO:handle_command: setting custom timeout handler (xmlrpc:http://localhost:8000/RPC2) Mar 15 14:04:39 devel rtpproxy[29111]: INFO:handle_command: pre-filling callee's address with 192.168.200.149:9494 Mar 15 14:04:50 devel rtpproxy[29111]: INFO:process_rtp: session timeout Mar 15 14:04:50 devel rtpproxy[29111]: INFO:remove_session: RTP stats: 963 in from callee, 10 in from caller, 973 relayed, 0 dropped Mar 15 14:04:50 devel rtpproxy[29111]: INFO:remove_session: RTCP stats: 5 in from callee, 1 in from caller, 6 relayed, 0 dropped Mar 15 14:04:50 devel rtpproxy[29111]: INFO:remove_session: session on ports 44206/41680 is cleaned up Started with: rtpproxy -T 10 -F -i -l 192.168.200.90 -s udp:127.0.0.1:7722 -n tcp:192.168.200.90:8000 -d DBUG SIP-ROUTER config: # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:localhost:7722") modparam("rtpproxy", "rtpproxy_tout", 3) modparam("rtpproxy", "timeout_socket", "xmlrpc:http://localhost:8000/RPC2") # The Port for incoming XML-RPC requests modparam("mi_xmlrpc", "port", 8000) modparam("mi_xmlrpc", "reply_option", 1) modparam("mi_xmlrpc", "buffer_size", 8192) modparam("mi_xmlrpc", "log_file", "/tmp/openser_xmlrpc") The session keeps active after the RTPPROXY proxy timeout. It seems dlg_terminate_dlg isn't being triggered. Am I missing something else beyond moving rtpproxy/ directory? As this thread is a user-question, I am copying SR-USERS mail list. I am also using xmlrpc-c-1.06.41. Thanks, Alexandre. -----Mensagem original----- De: kaiserbock2(a)googlemail.com [mailto:kaiserbock2@googlemail.com] Em nome de Carsten Bock Enviada em: terça-feira, 15 de março de 2011 05:15 Para: Alexandre Abreu Assunto: Re: RTPPROXY timeout patch. Hi Alexandre, sorry, yes, your correct. My changes are not in the 3.1 branch, but only GIT-Master. Only the GIT-Master currently holds the correct patch and config. I will need to backport the changes from the master... (will do so this week). Carsten 2011/3/14 Alexandre Abreu <alexandre.abreu(a)redt.com.br>: > Hello Carsten, > > Let me ask you a question: > > If "First is important to clarify: from version 3.0.0 on, Kamailio and > SER are identical in terms of source code.", then why your changes > does not reflect in kamailio 3.1 from GIT? > I did see the changes by getting the SIP-ROUTER from GIT. Kamailio 3.1 > from GIT does not have your code. > > Thanks. > Alexandre > > -----Mensagem original----- > De: kaiserbock2(a)googlemail.com [mailto:kaiserbock2@googlemail.com] Em > nome de Carsten Bock Enviada em: segunda-feira, 14 de março de 2011 > 05:31 > Para: Alexandre Abreu > Cc: RTPproxy Development > Assunto: Re: RTPPROXY timeout patch. > > Hello, > > i hope, you had a nice weekend... ;-) > Are you sure, you are using the GIT-Version of the > Kamailio/sip-router.org config? > In the GIT version it is now: > > modparam("rtpproxy", "timeout_socket", > "xmlrpc:http://localhost:8000/RPC2") > > http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blob;f=mod > ules/r > tpproxy/test/kamailio.cfg;h=0d5c73ff2065f33f262981fba2c07e65496e7415;h > b=HEAD > > This should work (as tested by me). > > Carsten > > 2011/3/11 Alexandre Abreu <alexandre.abreu(a)redt.com.br>: >> Hello Carsten, >> >> The patch was applied just fine but I can't get that feature working. >> See >> below: >> >> [root@devel log]# rtpproxy -v >> Basic version: 20040107 >> Extension 20050322: Support for multiple RTP streams and MOH >> Extension >> 20060704: Support for extra parameter in the V command Extension >> 20071116: Support for RTP re-packetization Extension 20071218: >> Support for forking (copying) RTP stream Extension 20080403: Support >> for RTP statistics querying Extension 20081102: Support for setting >> codecs in the update/lookup command Extension 20081224: Support for >> session timeout notifications Extension 20090810: Support for >> automatic bridging Extension 20100819: Support for timeout >> notifications using XML-RPC towards Kamailio/sip-router.org >> >> Using SR 3.1.2, my configuration file was exactly the same as the >> example in test/ subdirectory you commit to rtpproxy module. >> >> # RTP-Proxy >> modparam("rtpproxy", "rtpproxy_sock", "udp:localhost:22222") >> modparam("rtpproxy", "rtpproxy_tout", 3) modparam("rtpproxy", >> "timeout_socket", "http://localhost:8000/RPC2") >> >> # The Port for incoming XML-RPC requests modparam("mi_xmlrpc", >> "port", >> 8000) modparam("mi_xmlrpc", "reply_option", 1) modparam("mi_xmlrpc", >> "buffer_size", 8192) modparam("mi_xmlrpc", "log_file", >> "/tmp/openser_xmlrpc") >> >> Still inside test/ subdir we have the following example: >> >> [root@devel test]# cat exec_rtpproxy.txt ./rtpproxy -T 10 -f -F -i -l >> <your IP> -s udp:*:22222 -d DBUG >> >> You mean that if XML-RPC is used, it has to begin with "xmlrpc:". I >> can't see this string anywhere in these examples. I guess these files >> are not up to date. Am I correct? >> I can see the session timeout in RTPPROXY, but the session does not close. >> If I understand right how this feature works, it should send a >> "dlg_terminate_dlg + call-id" to send the BYE to both sides. >> >> Any tip on what could be wrong? >> >> Thanks, >> Alexandre >> >> -----Mensagem original----- >> De: kaiserbock2(a)googlemail.com [mailto:kaiserbock2@googlemail.com] Em >> nome de Carsten Bock Enviada em: quinta-feira, 10 de fevereiro de >> 2011 >> 20:53 >> Para: Alexandre Abreu >> Cc: RTPproxy Development >> Assunto: Re: RTPPROXY timeout patch. >> >> Hi Alexandre, >> >> i updated the patch in the sip-router-repository (and attached the >> patch to this email). The new patch and the modifications to the >> rtpproxy module need some more testing, i have only done basic > functionality testing so far. >> I have reduced the patch a little, the socket, if the >> XML-RPC-Interface is used, has to begin with "xmlrpc:". This way, the >> protocol of the RTP-Proxy stays intact. >> If you have spare-time, please test it; otherwise, i will do it asap >> (but i am travelling a lot next week). >> >> @Maxim: This new version of the patch is much better, than the old one. >> Maybe you can review this one? Thanks! >> >> Carsten >> >> >> 2011/2/10 Carsten Bock <lists(a)bock.info>: >>> Hi Alexandre, >>> >>> the current GIT-version contains some more modifications regarding >>> timeout sockets. >>> That is the reason why the patch does not work anymore. >>> I will need to review the code and adapt my patch accordingly. >>> I will do so later. >>> >>> Carsten >>> >>> 2011/2/2 Alexandre Abreu <alexandre.abreu(a)redt.com.br>: >>>> Hi Carsten, >>>> >>>> The stable version of RTPProxy is indeed 1.2.1. In this version, >>>> there is no file 'rtpp_notify.c' so I think the patch is really >>>> for git-version. Maybe the original patch was made regarding to an >>>> earlier >> git commit. >>>> >>>> Thanks, >>>> Alexandre >>>> >>>> -----Mensagem original----- >>>> De: kaiserbock2(a)googlemail.com [mailto:kaiserbock2@googlemail.com] >>>> Em nome de Carsten Bock Enviada em: quarta-feira, 2 de fevereiro de >>>> 2011 >>>> 06:02 >>>> Para: Alexandre Abreu >>>> Cc: RTPproxy Development >>>> Assunto: Re: RTPPROXY timeout patch. >>>> >>>> Hi Alexandre, >>>> >>>> the patch was for the stable version of RTP-Proxy (1.2.1?). I will >>>> review the patch for the git-version. >>>> I am a little busy at the moment (business travel), so i can first >>>> do this on friday or next week. >>>> >>>> Kind regards, >>>> Carsten >>>> >>>> P.S.: Please always CC the list. >>>> >>>> 2011/2/1 Alexandre Abreu <alexandre.abreu(a)redt.com.br>: >>>>> Hi Carsten, >>>>> >>>>> >>>>> >>>>> I am testing the new RTPProxy timeout feature in Kamailio (GIT > version). >>>>> >>>>> >>>>> >>>>> Got the rtpproxy from GIT: >>>>> >>>>> git clone git://sippy.git.sourceforge.net/gitroot/sippy/rtpproxy >>>>> >>>>> >>>>> >>>>> I applied the patch: >>>>> >>>>> [root@devel rtpproxy]# patch < rtpproxy.patch >>>>> >>>>> patching file Makefile.am >>>>> >>>>> Hunk #1 FAILED at 4. >>>>> >>>>> 1 out of 1 hunk FAILED -- saving rejects to file Makefile.am.rej >>>>> >>>>> patching file config.h.in >>>>> >>>>> patching file configure.ac >>>>> >>>>> patching file rtpp_command.c >>>>> >>>>> Hunk #1 succeeded at 54 (offset 2 lines). >>>>> >>>>> Hunk #3 succeeded at 272 (offset 2 lines). >>>>> >>>>> Hunk #4 succeeded at 315 with fuzz 2. >>>>> >>>>> Hunk #5 FAILED at 466. >>>>> >>>>> Hunk #6 FAILED at 961. >>>>> >>>>> 2 out of 6 hunks FAILED -- saving rejects to file >>>>> rtpp_command.c.rej >>>>> >>>>> patching file rtpp_notify.c >>>>> >>>>> Hunk #1 FAILED at 38. >>>>> >>>>> Hunk #2 succeeded at 263 (offset 78 lines). >>>>> >>>>> misordered hunks! output would be garbled >>>>> >>>>> Hunk #4 FAILED at 250. >>>>> >>>>> Hunk #5 FAILED at 337. >>>>> >>>>> 3 out of 5 hunks FAILED -- saving rejects to file >>>>> rtpp_notify.c.rej >>>>> >>>>> patching file rtpp_notify.h >>>>> >>>>> Hunk #1 succeeded at 39 with fuzz 2 (offset 8 lines). >>>>> >>>>> patching file rtpp_session.c >>>>> >>>>> patching file rtpp_session.h >>>>> >>>>> >>>>> >>>>> As you see, not so clean but the files were patched. >>>>> >>>>> >>>>> >>>>> On CentOS 5, I cant find the LIBXMLRPC through YUM/RPM, so I >>>>> compiled xmlrpc-c-1.06.41.tgz by hand: >>>>> >>>>> >>>>> >>>>> ./configure prefix=/usr && make && make install >>>>> >>>>> AND ./configure && make && make install >>>>> >>>>> With that, the headers are both on /usr/lib/include and >>>>> /usr/local/lib/include (I did that just to be sure that the config >>>>> wasnt looking at the wrong place). >>>>> >>>>> >>>>> >>>>> xmlrpc_abyss.h >>>>> >>>>> xmlrpc_cgi.h >>>>> >>>>> XmlRpcCpp.h >>>>> >>>>> xmlrpc_server.h >>>>> >>>>> xmlrpc_client.h >>>>> >>>>> xmlrpc.h >>>>> >>>>> xmlrpc_server_w32httpsys.h >>>>> >>>>> >>>>> >>>>> The configure script from patched RTPPROXY doesnt _try_ to find >>>>> the LIBXMLRPC (theres no reference in config.log) and therefore >>>>> my rtpproxy doesnt have the support for the timeout notification. >>>>> >>>>> >>>>> >>>>> Any comment on what could be wrong? >>>>> >>>>> >>>>> >>>>> Alexandre >>>> >>>> >>>> >>>> -- >>>> Carsten Bock >>>> Schomburgstr. 80 >>>> 22767 Hamburg >>>> Germany >>>> >>>> Mobile +49 179 2021244 >>>> Home +49 40 34927217 >>>> Büro (Verl) +49 5246 801427 >>>> Fax +49 40 34927218 >>>> mailto:carsten@bock.info >>>> >>>> >>> >>> >>> >>> -- >>> Carsten Bock >>> Schomburgstr. 80 >>> 22767 Hamburg >>> Germany >>> >>> Mobile +49 179 2021244 >>> Home +49 40 34927217 >>> Büro (Verl) +49 5246 801427 >>> Fax +49 40 34927218 >>> mailto:carsten@bock.info >>> >> >> >> >> -- >> Carsten Bock >> Schomburgstr. 80 >> 22767 Hamburg >> Germany >> >> Mobile +49 179 2021244 >> Home +49 40 34927217 >> Büro (Verl) +49 5246 801427 >> Fax +49 40 34927218 >> mailto:carsten@bock.info >> >> > > > > -- > Carsten Bock > http://www.ng-voice.com > mailto:carsten@ng-voice.com > > -- Carsten Bock http://www.ng-voice.com mailto:carsten@ng-voice.com

2 13

Release for SPCE version 2.2
by Deon Vermeulen 17 Mar '11

17 Mar '11

Hi Guys When can we expect the release of spce version 2.2? Kind Regards Deon Vermeulen

1 0

Kamailio - Asterisk CDR
by Lucas Alvarez 17 Mar '11

17 Mar '11

Hi, I have an integration kamailio-asterisk. I have many calls that don't pass through asterisk, kamailio route that calls to other media gateways. I would like to log those calls in asterisk CDR, is that possible? Could someone give me any direction? Thanks in advance. Lucas

4 3

Problem on multiple interfaces and conversion between UDP and TCP
by David 16 Mar '11

16 Mar '11

Hello, I have a Kamailio 3.0 server with two interfaces. 192.168.x.x and a public interface. I send PUBLISH out on the private interface to my presence server. Everything else happens on the public address. I did not use the "listen=" option in the config, so Kamailio was automatically detecting my interfaces to bind to. Last night I had a provider sending packets in on TCP and Kamailio was sending out on UDP. Unfortunately, it kept using the private interface despite that the IP was unreachable on this interface. It looks like Kamailio uses the first interface it finds when converting between TCP and UDP, and I believe that on the server, the private interface was being listed first. I finally added listen= in kamailio with all 4 interfaces ( tcp/udp public and tcp/udp private ). I listed the public UDP interface first, restarted and the packets started going out on the public interface. Is this a feature or a bug ? How does Kamailio decide which interface to use ? What if I sent a packet to an IP that is only reachable on one interface? At the very least, I think that a note about this difficulty should be included in the default config so that future users don't have the same issue I did. Thanks, David

2 1

getting fragmented packet when adding record-route header
by Asgaroth 16 Mar '11

16 Mar '11

Hi All, I have a scenario where I have 2 asterisk media servers that, when calling a registered sip account, will forward off the invite to a location server, who looks up the contact information, then forwards the invite off to the proxy that the user registered against (done via the path module). This all appears to work perfectly until the us hangs up the call. The BYE message does not make it all the way back to the originating asterisk media server, it makes it back to the location server that looked up the registration information. To overcome this issue, I add a "Record-Route" header specifying the address of the asterisk media server that the request originated from. However, when I add the custom header, the Invite shows up as fragmented packet my wire-shark trace at the end-user's soft-phone. The call sets up properly and when I hang up it looks like the BYE message makes it all the way back to the media server, but when it is relayed on the the caller, the sip proxy then fails the call with a message too big. If I remove the line that adds the record-route header, then all looks fine, except that I am unable to tear down calls from sip end-points. The specific line I'm using to add the header is as follows: insert_hf("Record-Route: <sip:$si:$sp;lr=on>\r\n","Record-Route"); I do remove this from the route header on in-dialog replies as loose-route does not remove the header on the original location server as it does not see the route header as local, so I manually remove it. Has anyone come across this particular issue? I'm using Kamailio 3.1.2. Am I going about this is the wrong way? I've been reading the module docs for registrar/usrloc/rr/path/textops to see if there is something I've missed but cannot see it. Any tips/suggestions would be greatly appreciated. Thanks

3 7

ACK not sent and rr-enforced
by Dominguez Jover, Ricardo 16 Mar '11

16 Mar '11

Hi everybody. I am using Kamailio 3.1 and RTP proxy for internet calls. I'm using external test accounts to check if the calls are established and the media flow is ok. When I use a sip2sip.info or VoIP Talk accounts, then all is working fine between my internal and these external accounts. But when I use a iptel.org account and this account calls to an internal account (registered with kamailio), then callee sends the 200 OK to the SIP proxy and the SIP proxy to iptel. IPtel.org proxy sends the ACK to my proxy with this lines at the end of the packet: P-hint: rr-enforced\r\n P-hint: rr-enforced\r\n And my SIP proxy never resends the ACK to the callee, so the callee resends OK 200 periodically and after 32 seconds sends a BYE message and the call is finished. I've been reading posts about missing ACKs but I can't find the answer to my problem, that it seems like "t_check_trans" doesn´t recognize the ACK as related to a transaction. But this is only with IPTEL accounts, my proxy SIP is working with other SIP providers, so I don't know if forcing relay of every ACK packet is a good idea. Any help would be appreciated. Thanks, Ricardo

4 10

Installation Problem: : /usr/local/sbin/kamailio: not found
by Derrick Ding 16 Mar '11

16 Mar '11

Hi, All Yesterday I install Kamailio(3.1.x) by myself under a new Ubuntu server 10.04 environment. I got it from http://www.kamailio.org/pub/kamailio/3.1.2/bin/ kamailio-3.1.2_linux_i386.tar.gz <http://www.kamailio.org/pub/kamailio/3.1.2/bin/kamailio-3.1.2_linux_i38 6.tar.gz> I put all stuff under corresponding directory: /usr/local/sbin with bin file /usr/local/etc/kamailio with .cfg file /usr/local/lib/kamailio with lib file Based on http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.1.x-from-gi t I create /etc/init.d/kamailio PATH=/sbin:/bin:/usr/sbin:/usr/bin DAEMON=/usr/local/sbin/kamailio NAME=kamailio DESC=kamailio HOMEDIR=/var/run/kamailio PIDFILE=$HOMEDIR/$NAME.pid DEFAULTS=/etc/default/kamailio RUN_KAMAILIO=no Also I create /etc/default/ kamailio I find a problem after when I run /etc/init.d/kamailio start, it says: root@aastra:/etc/init.d# /etc/init.d/kamailio start Not starting kamailio: invalid configuration file! -e /etc/init.d/kamailio: 168: /usr/local/sbin/kamailio: not found But the file is there: root@aastra:/etc/init.d# ls -l /usr/local/sbin/kamailio -rwxr-xr-x 1 root root 4103257 2010-07-01 12:16 /usr/local/sbin/kamailio root@aastra:/etc/init.d# /usr/local/sbin/kamailio bash: /usr/local/sbin/kamailio: No such file or directory I wonder if this /usr/local/sbin/kamailio is not for Ubuntu server 10.04 and cannot be run Thanks for help Derrick

2 3

major bug in auth module?
by Juha Heinanen 16 Mar '11

16 Mar '11

one sip ua sent this kind of authorization header: Authorization: Digest username="foo",realm="bar",uri="xxx",response="e439fa7438452da7d50690f2268be16f",nonce="4d8060d2000041b2430e64fe7ec681d8fb5709e9c72b8a01",qop=auth,cnonce="0004ccb0",nc=000060a2. and sr auth module didn't like it but emitted error message FIXME:check_nonce: bad nc value. i went and checked the code and found this: if (nc_enabled && (pf & NF_VALID_NC_ID) && auth->digest.nc.s && auth->digest.nc.len){ if (str2int(&auth->digest.nc, &nc)!=0){ /* error, bad nc */ ERR("FIXME:check_nonce: bad nc value %.*s\n", auth->digest.nc.len, auth->digest.nc.s); return 5; /* invalid nc */ } i.e., looks like it is using str2int to convert to nc value to int. the problem is that str2int only handles decimal strings, whereas nc value is a hex string. am i dreaming or what? -- juha

1 1

LCR Routes and memory allocations
by Graham Wooden 15 Mar '11

15 Mar '11

Hi there, I have now approximately 400,000 routes defined in my LCR table and I would like to make sure that I have enough memory allocated for it. I read somewhere (a non official Kamailio site) about PKG_MEM_POOL_SIZE and SHM_MEM_SIZE needing to be altered to handle the increased routes. System is dedicated for the Kamailio 1.5.5 install and has 4GB of RAM. Thoughts to what I need to change these values (or anything else) to accommodate? Thanks, -graham

5 6

Error: compile kamailio with STUN support
by Derrick Ding 15 Mar '11

15 Mar '11

Hi, All I met same problem when compile kamailio with STUN support. http://www.voipuser.org/forum_topic_18741.html /usr/include/bits/string3.h:83: warning: memset used with constant zero length parameter; this could be due to transposed parameters ser_stun.o: In function `add_fingerprint': /root/kamailio-3.0.1/ser_stun.c:890: undefined reference to `SHA1' ser_stun.o: In function `validate_fingerprint': /root/kamailio-3.0.1/ser_stun.c:973: undefined reference to `SHA1' collect2: ld returned 1 exit status make: *** [kamailio] Error 1 I search online that need link libssl -lssl But I don't know where to add this compile flag. The Makefile of this project is too complicated for me to understand. Any one can tell me how to I can fix this? By the way, I have install libssl. It's /usr/lib/libssl.so.0.9.8 Sincerely, Derrick

2 1

Q.850 Reason in CANCEL on Kamailio 3.0
by Jeremy McNamara 15 Mar '11

15 Mar '11

I am attempting to add a Reason header with Q.850 result codes. Unfortunately, this does not work: if (is_method("CANCEL")) { append_hf("Reason: Q.850; cause=16\r\n"); if (t_check_trans()) { t_relay(); } exit; } If I have to use 3.1.x, does it support sending Q.850 codes? Jeremy McNamara

2 4

Kamailio setup
by roman＠dmytriv.com 15 Mar '11

15 Mar '11

Hello everyone!!! What location I use for chrooted directory? Thanks in advance Roman

2 1

Re: [SR-Users] Call subscriber online
by Daniel-Constantin Mierla 15 Mar '11

15 Mar '11

Hello, shouldn't the call go to location service before relaying to subscriber B? Is B at a fix address an port and that is local host port 5060? Are you doing all in your computer for testing purposes, because otherwise an application bound to localhost (like could be the softphone B) cannot really communicate with the inter/intra-network? Cheers, Daniel On 3/11/11 4:50 PM, Stefano Larosa wrote: > > Hi, > > I'm new on Kamailio 3.0 > > This is the scenario I would like to build: > > 1 Subscriber A -> 2 kamailio -> 3 asterisk -> 4 Kamailio -> 5 > Subscriber B > > Everything is working fine until the last step > > This is the code that manage the call from asterisk to kamailio > > /if(is_method("INVITE") && (src_ip==80.169.xx.xx) )/ > > / {/ > > / route(TOPROXYUSER);/ > > / }/ > > And this is the code that should end the call the the subscriber > > route[TOPROXYUSER] { > > xlog("L_NOTICE", "$mi route[$rm][0] $fu > -> $ru START PROCESSING MESSAGE\n"); > > rewritehostport("127.0.0.1:5060"); > > if (is_method("BYE|CANCEL")) { > > route(FAIL_ONE); > > } else if (is_method("INVITE")){ > > route(RELAY); > > }; > > exit; > > } > > Thank you, > > Stivu. > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://www.asipto.com

2 1

Registrar module creates an extra branch
by Efelin Novak 15 Mar '11

15 Mar '11

Hi Folks, I have a problem with registrar module (I suppose). When A calls to B, is some specific situation, B receives two INVITEs. This mostly (not only) happens when one account is registered on two or more UAs. In this situation extra INVITE is sent only to the one that is second in the list (kamctl ul show ACCOUNT). kamctl ul show Alice: Contact:: <sip:orlvuzyd@192.168.28.72:52129 >;q=0;expires=1916;flags=0x0;cflags=0x40;socket=<udp:8.8.8.8:5060 >;methods=0xFFFFFFFF;received=<sip:8.8.8.8:52129>;user_agent=<Blink 0.2.5 (Linux)> Contact:: <sip:Alice@192.168.28.51:5060 >;q=;expires=3061;flags=0x0;cflags=0x40;socket=<udp:8.8.8.8:5060 >;methods=0x129F;received=<sip:8.8.8.8:5060 >;user_agent=<Linksys/PAP2T-3.1.15(LS)> My situation. 1. Alice calls to BOB. 2. kamailio looks up in location table for BOBs registrations 3. in DB there are two rows (two devices(linksys PAP2T and Blink(blink is second one))) 4. kamailio send 3 INVITES ( linksys, blink and blink). Every INVITE has different via.branch (Via: SIP/2.0/UDP 8.8.8.8;branch=z9hG4bKe2d7.009b1df2.1) (Via: SIP/2.0/UDP 8.8.8.8;branch=z9hG4bKe2d7.009b1df2.2) (Via: SIP/2.0/UDP 8.8.8.8;branch=z9hG4bKe2d7.009b1df2.3) All my registrar parameters are set to default. I have no idea whether this is a bug or my missconfiguration. Thanks for replies. Kind regards, Efelin Novak

2 2

usuarios para kamailio y asterisk
by M. ASUNCION MERINO RODRIGUEZ 14 Mar '11

14 Mar '11

Hola en la actualidad estoy montando un servidor kamailio que trabaja con un servidor asterisk. Entiendo que lo mejor seria que tanto kamailio como asterisk trabajaran con los mismos usuarios, por eso el servidor kamailio autentica los usuarios contra un servidor radius. Este servidor radius, acceder a un directorio ldap, para validar a los usuarios cuyo usuario es el login de correo y la contraseña correspondiente. Mi pregunta es, como puedo manejar usuarios como "login" si en asterisk existen usuarios creados como números de telefono? debería hacer que los usuarios de radius fueran numeros de telefono? Cualquier ayuda es bienvenida Gracias

1 0

Paging from asterisk to accounts with Multiple Registrations...
by Derrick Bradbury 14 Mar '11

14 Mar '11

Hi all... interesting problem... I am using kamailio as the registration server for a cluster of asterisk servers. I am allowing multiple registrations on the same account to allow for parallel forking on a normal call... The problem is set paging... when I send the page to the account that has multiple registrations, the parallel fork works properly and the first phone to respond gets "control" of the call. Unfortunately there is an auto answer field in the invite. So simple case: one account, two phones.. invite gets sent to both phones, both phones answer... BUT one phone grabs "control" of the call, and the other phone no longer receives any more commands... so it never hangs up as it doesn't receive a "BYE". Is there any way to turn off the parallel fork behavior, and keep sending all packets to both (or more) phones? Any thoughts are appreciated! Thanks, Derrick

2 1

Re: [SR-Users] decimal fraction problem
by Daniel-Constantin Mierla 14 Mar '11

14 Mar '11

Hello, cc-ing to the mailing list is very important because even it is an email to show the previous answer was good, that will help other people with similar problem that search on web and read the mailing list archive to know the proposed solution worked and they can use it without asking again on mailing list if it was ok or not. Thanks, Daniel On 3/14/11 12:07 PM, 侯旭光 wrote: > got it > > thanks > > cheers > > 在 2011年3月14日下午5:22，Daniel-Constantin Mierla <miconda(a)gmail.com> 写道： >> Hello, >> >> are you using kamailio 3.1.x? If not, you have to upgrade, the >> $branch(...) variable was updated to be writable starting with this version. >> >> I played last week with it in a need of combining serial forking with >> parallel forking and all is ok with assigning values to $branch(...). >> >> Cheers, >> Daniel >> >> On 3/14/11 5:25 AM, 侯旭光 wrote: >>> sorry to bother again >>> >>> >>> $(branch(q)[-1]) = $var(q); >>> >>> this script line doesn't work and the pv $branch() aren't >>> writable,just readable . index -1 is not accessable either. >>> >>> if append_branch() function doesn't take the q value parameter,the >>> $branch(q) just return NULL (which I think is the default value >>> Q_UNSPECFIED=-1) >>> >>> I find a function set_ruri_q() in dset.c but I don't know how to call >>> it in the configure file. >>> >>> >>> 2011/2/23 Daniel-Constantin Mierla <miconda(a)gmail.com>: >>>> Hello, >>>> >>>> On 2/21/11 10:28 AM, 侯旭光 wrote: >>>> >>>> Hello >>>> I need to add q value while using function append_branch(),but the function >>>> only takes decimal fraction as the parameter. >>>> What if I want to use pv to add q value? >>>> The $var and $avp just have string and integer type. >>>> Thanks a lot! >>>> >>>> do: >>>> >>>> km_append_branch("$var(branchuri)"); >>>> $(branch(q)[-1]) = $var(q); >>>> >>>> $var(q) has to hold an integer value that represents the decimal fraction >>>> value multiplied with 100 (so if q should be 0.5, then $var(q) = 50). >>>> >>>> Cheers, >>>> Daniel >>>> >>>> -- >>>> Daniel-Constantin Mierla >>>> http://www.asipto.com >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>> sr-users(a)lists.sip-router.org >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> -- >> Daniel-Constantin Mierla >> http://www.asipto.com >> >> -- Daniel-Constantin Mierla http://www.asipto.com

1 0

change in list setup - default is now reply to list
by Henning Westerholt 14 Mar '11

14 Mar '11

Hello, for your information - i just did a small change in the mailling list configuration for sr-dev and sr-users, now all replies goes per default to the list instead of the sender. This should help to keep the discussion on the list, especially if new people join. :-) In the past a lot of "Please always include the list in your reply" reminders were necessary.. If you choose "reply to all" then the mail still goes to the list and the sender, so no change in this case. Regards, Henning

2 1

Siremis - Auto update dialplan on save?
by Russell Feeser 14 Mar '11

14 Mar '11

To Siremis Admin, I'm a bit of a php rookie, and even more of a noob with openbiz cubi, but I've burnt a day and a half trying to implement a new button on the SER Menu > Server Services > Dialplan List page, and curious if you could give some words of advice concerning my plight. Ideally, I would like dialplan to auto update the Kamailio mySQL database with the dp_reload feature; I am simply sick of bouncing Kamailio with a "kamctl restart" from the command line each time I implement a change to the dialplan in Siremis. I have successfully enabled this feature on the drop-down menu in SER Menu > Command Services > MI Commands, however, I don't want to have to navigate to the MI Commands page each time I wish to refresh dialplan. In siremisMICommands.xml, I was able to activate dp_reload with the following code: (taken from siremisMICommands.xml) <?xml version="1.0" standalone="no"?> <PluginService Name="siremisMICommands" Package="asipto" Class="siremisMICommands"> <MIConfig name="MIConfig" type="udp" mode="rich"> <Local name="local" address="127.0.0.1" port="8044" timeout="3.0"/> <Remote name="remote" address="127.0.0.1" port="8033"/> <MICommands> <cmd name="ps" title="List Processes" command="ps"/> <cmd name="uptime" title="Show Uptime" command="uptime"/> <cmd name="getstatsall" title="Get All Statistics" command="get_statistics all"/> <cmd name="which" title="Get All MI Commands" command="which"/> <cmd name="dp_reload" title="Reload Dailplan" command="dp_reload"/> <------ Code I added, WORKS! </MICommands> </MIConfig> </PluginService> Here's where I'm stuck: I can't seem to get a new button to implement dp_reload in the DialplanListForm.xml. I'm not sure if I need to write a function to call dp_reload, or if I can simply activate dp_reload from the DialplanListForm.xml. The code I have written creates a button on the SER Menu > Server Services > Dialplan List page, but thats as far as I've gotten. My code currently looks like this: (taken from DialplanListForm.xml) <Element Name="btn_refresh" Class="Button" Text="Activate" CssClass="button_gray_m" Description="Refresh dialplan in Kamailio without restarting Kamailio"> <EventHandler Name="dp_reload" Event="onclick" EventLogMsg="" Function="" ContextMenu="Activate"/> </Element> Anything you can say would be most helpful, as this is a feature I'm sure many using Siremis would appreciate having. Best, -- R. Z. Feeser 717.460.8095 www.RZFeeser.com www.NaturaGardens.com

2 1

pseudo variables available in on_reply route
by Asgaroth 14 Mar '11

14 Mar '11

Hi All, I have a requirement to perform some processing based on the source and destination addresses on a message in on_reply route. I can get source ip address using $si pseudo variable, but I cant seem to access the destination ($dd). Is there any way I can access destination ip/domain of message in on_reply route? Thanks for any tips.

2 1

Register a CISCO IP phone
by Pang, Gary (Liguang) 14 Mar '11

14 Mar '11

Dear Sir, I have a difficulty to register a CISCO SPA 303 IP phone. I can register a Soft phone to the SER server by setting the Hold IP address, Proxy, account number.. But it is not working with the CISCO phone. Can you advice? Thanks, Gary

3 2

decimal fraction problem
by 侯旭光 14 Mar '11

14 Mar '11

Hello I need to add q value while using function append_branch(),but the function only takes decimal fraction as the parameter. What if I want to use pv to add q value? The $var and $avp just have string and integer type. Thanks a lot!

2 3

OPENSER MIB
by Stefan Tiedje 14 Mar '11

14 Mar '11

Hi, In the Kamailio OPENSER-MIB there is the counter "openserTotalNumFailedDialogSetups". This is a Counter32. The description is: "The total number of calls that failed with an error. The following codes define a failed call:" Question: * I'm looking for the corresponding counter to "openserTotalNumFailedDialogSetups" who counts successful Dialog setups of Counter32 type. Does it exist? * If not, does it exist a work around? * Where in the code can the new suggested counter be added? * Something else???? Suggestion for the new counter is a name like: "openserTotalNumSucceededDialogSetups". It has a counter32. Description: "The total number of calls that succeeded" I know that there are the counters openserCurNumDialogs, openserCurNumDialogsInProgress and openserCurNumDialogsInSetup but these are of Gauge type who only reflects the current situation. These Gauge counters can't be used together with a Counter32 counter. That don't mix. The calculation done for the counter "openserCurNumDialogsInProgress" should be used where every new dialog setup is added to the new suggested counter. A counter of 32 should cover a great deal of connections. These counters are usually read, if used, every 15 minutes or 1 hour. Rationale: The reason for the new counter is that a calculation between succeeded and failed dialog setups can be done and be used for SLA agreements. Without this, its hard to make any customer versus provider agreements. /Stefan PS. Ask if anything is unclear and I need an answer rapidly.

2 1

acc failed_transaction_flag and t_newtran()
by Juha Heinanen 12 Mar '11

12 Mar '11

in order to get invite requests accounted, it seems that accounting flags need to be set before t_newtran() is called. is there any way to unset the accounting flags after calling t_newtran()? for example, i would not like to account invites that result to 407 proxy authentication required. -- juha

2 4

pseudo variables available in on_reply route
by Asgaroth 11 Mar '11

11 Mar '11

Hi All, I have a requirement to perform some processing based on the source and destination addresses on a message in on_reply route. I can get source ip address using $si pseudo variable, but I cant seem to access the destination ($dd). Is there any way I can access destination ip/domain of message in on_reply route? Thanks for any tips.

5 8

Call subscriber online
by Stefano Larosa 11 Mar '11

11 Mar '11

Hi, I'm new on Kamailio 3.0 This is the scenario I would like to build: 1 Subscriber A -> 2 kamailio -> 3 asterisk -> 4 Kamailio -> 5 Subscriber B Everything is working fine until the last step This is the code that manage the call from asterisk to kamailio if(is_method("INVITE") && (src_ip==80.169.xx.xx) ) { route(TOPROXYUSER); } And this is the code that should end the call the the subscriber route[TOPROXYUSER] { xlog("L_NOTICE", "$mi route[$rm][0] $fu -> $ru START PROCESSING MESSAGE\n"); rewritehostport("127.0.0.1:5060"); if (is_method("BYE|CANCEL")) { route(FAIL_ONE); } else if (is_method("INVITE")){ route(RELAY); }; exit; } Thank you, Stivu.

1 0

kamailio setup
by roman＠dmytriv.com 10 Mar '11

10 Mar '11

Hello there! I'm in the process of setting my first kamailio server and run into dead end. I was following your instructions very closely at http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.1.x-from-git I'm using Centos 5.5 i386. Everything vent pretty much fine until step 7 were I had to create MySQL database. When I run /usr/local/kamailio-3.1/etc/kamailio/kamctlrc it gives me "Permission denied" error. I was able to access kamctlrc with a text editor and uncommented DBENGINE=MYSQL. Now when I run /usr/local/kamailio-3.1/sbin/kamdbctl create I have ERROR 2002. Please take a look at the attachment. Please help. Roman

4 3

Question about LCR
by Geoffrey Mina 10 Mar '11

10 Mar '11

Hello, I have a question about LCR which I have been unable to solve. I have 4 upstream carrier gateways owned by 2 carriers. Each carrier provides a primary and secondary gateway for load balancing purposes. On a 5XX error I am trying to send the same call to the other carrier. If both carriers reject the call with 5XX, I allow the response to go downstream to my asterisk server. The issue I am running into is that in a scenario where both my carriers respond with a 5XX, I end up presenting the same call to all 4 gateways. I would like to present the call to one gateway on each carrier and not try the same carriers second gateway for the same call. Here is what is happening now: ASTERISK --> INVITE --> KAMAILIO INVITE --> CARRIER A/GATEWAY 1 <-- 5XX Error INVITE --> CARRIER A/GATEWAY 2 <-- 5XX Error INVITE --> CARRIER B/GATEWAY 1 <-- 5XX Error INVITE --> CARRIER B/GATEWAY 2 <-- 5XX Error KAMAILIO --> 5XX Error --> ASTERISK OR any combination of the above... i.e. ASTERISK --> INVITE --> KAMAILIO INVITE --> CARRIER A/GATEWAY 1 <-- 5XX Error INVITE --> CARRIER B/GATEWAY 2 <-- 5XX Error INVITE --> CARRIER B/GATEWAY 1 <-- 5XX Error INVITE --> CARRIER A/GATEWAY 2 <-- 5XX Error KAMAILIO --> 5XX Error --> ASTERISK What I want to happen is: ASTERISK --> INVITE --> KAMAILIO INVITE --> CARRIER A/GATEWAY 1 or 2 <-- 5XX Error INVITE --> CARRIER B/GATEWAY 1 or 2 <-- 5XX Error KAMAILIO --> 5XX Error --> ASTERISK I tried dealing with the issue using some flags on the gateway, but i couldn't get the logic to work properly. Here is the path I was heading down, but my plan fell apart after some testing. CARRIER A has a gateway flag of "1" CARRIER B has a gateway flag of "2" failure_route[1]{ $var(gw_flag) = $avp(i:712); while(next_gw()){ if($var(gw_flag) != $avp(i:712)){ xlog("L_INFO","Found an LCR destination which is different than current, routing. ($ci)"); t_on_reply("1"); t_on_failure("1"); t_relay(); exit; }else{ xlog("L_INFO","The next destination in LCR has the same AVP flag, skipping. ($ci)"); } } # let the reply go upstram - it is the default action xlog("L_ERR", "No Next Gateway - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); exit; } Any help would be greatly appreciated. Thanks, Geoff

4 6

kamailio setup
by roman＠dmytriv.com 09 Mar '11

09 Mar '11

Hello there! I'm in the process of setting my first kamailio server and run into dead end. I was following your instructions very closely at http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.1.x-from-git I'm using Centos 5.5 i386. Everything vent pretty much fine until step 7 were I had to create MySQL database. When I run /usr/local/kamailio-3.1/etc/kamailio/kamctlrc it gives me "Permission denied" error. I was able to access kamctlrc with a text editor and uncommented DBENGINE=MYSQL. Now when I run /usr/local/kamailio-3.1/sbin/kamdbctl create I have ERROR 2002. Please take a look at the attachment. Please help. Roman

1 0

Does Kamailio support RFC5626 as Edge Proxy(or outbound proxy)
by Derrick Ding 09 Mar '11

09 Mar '11

Dear All, I wonder if Kamailio supports RFC5626 as edge proxy or outbound proxy. That is, Edge Proxy adds a Path header with a flow token and includes the "ob" parameter when Processing Register Requests. Eg: Path: <sip:VskztcQ/S8p4WPbOnHbuyh5iJvJIW3ib@ep1.example.com;lr;ob> If does, how can I config it in kamailio.cfg. At present, Kamailio can only forward register msg to sip server and reply 200 OK to SIP UA with "require: outbound" when register accepted. No "Path" and "ob" added in the msg forwarded to SIP server. You answer is appreciated!! Sincerely, Derrick

3 3

#!define and #!subst problem
by Klaus Darilion 09 Mar '11

09 Mar '11

Hi! I tried #!subst "/IPADDRESS_VIRTUAL/83.136.32.161/" listen=udp:IPADDRESS_VIRTUAL:5060 and #!define IPADDRESS_VIRTUAL 83.136.32.161 listen=udp:IPADDRESS_VIRTUAL:5060 Both do not work - am I doing something wrong or is this a known limitation with "listen" statements? Thanks Klaus

2 2

dispatcher confusion
by Klaus Darilion 07 Mar '11

07 Mar '11

Hi! Every time I use the dispatcher(k) module I am confused again. Sometimes it seems that "probing" is a dedicated state, sometimes it seems that "probing" is done also in "active" state, but never in "inactive" state. IMO "probing" should only be a flag which indicates if OPTIONS should sent or not. If probing is successful, then the state should be "active". If probing is unsuccessful, then the state should be "inactive". Current behavior is very strange (ds_probing_mode(0)): --> startup state: A --> no probing kamctl fifo ds_set_state i 1 sip:.... --> state: I --> no probing kamctl fifo ds_set_state a 1 sip:.... --> state: A --> no probing calling 3 times ds_mark_dst("p") --> state: P --> probing (and dst is still loaded as last value into the dst_avp, why?) kamctl fifo ds_set_state i 1 sip:.... --> state: I --> still probing kamctl fifo ds_set_state a 1 sip:.... --> state: P (???) --> probing Thus, "ds_set_state a ..." does not set active, but probing if it was probing before. Strange. And as "inactive" does not stop probing when dst was in probing mode, the destination becomes automatically "active" if the probing succeeds. And why is a destination in probing mode loaded into the dst_avp? Very weird. Is there a reason for this behavior? Thanks Klaus

3 4

Kamailio 3.0.4: dispatcher module request
by Klaus Feichtinger 07 Mar '11

07 Mar '11

Hello list, I have a question to the dispatcher module in Kamailio version 3.0.4. In my integration I use this module for distributing calls to a set of gateways (with the round robin algorithm). Sometimes a gateway does not react or send back a negative response. In that case I use the function 'ds_mark_dst("p")' in the failure_route and mark the current destination with the flag 'probing'. I have set the threshhold value for probing mode to '10'. This means that a gateway is effectively set to probing mode only when it was marked with the flag 'probing' for 10 times. However, sometimes a gateway is sending back a negative response for e.g. 5 times and is afterwards working fine - without any problems. A few hours later (e.g.) it is sending a negative reply again for one time. This means: the counter for the probing_flags is increasing every time when the function ds_mark_dst("p") is executed (in case that the process was not restarted in the meantime!). So the problem is, that the counter is reaching the threshhold value at any time and at that moment the gateway is set to probing mode. Even if the gateway has sent a negative reply for only one time (at that moment) - but according the history the proging_flag was set several times. What I miss in this module is a function to _reset_ that flag counter from a REPLY_ROUTE. Note: I have to reset the counter, because the gateway does not yet SIP OPTIONS requests and therefore it is set to 'probing' all time until it is manually set back to active via MI command or after a process restart. According the README file (and practical experience) the function ds_mark_dst() is executable only within the failure_route. However, when I want to set the flag ("a") for the current destination I will do it within a reply_route, but not in the failure_route (because I have received a positive response and not a negative one). From my point of view it does not make sense setting the 'active' flag for a destination from within a FAILURE route. Does anybody have an idea, how the 'flag-counter' could be reset (from within the script)? I do not want to use a MI command..... configuration excerpt: [...] modparam("dispatcher", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)") modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)") modparam("dispatcher", "ds_ping_from", "sip:proxy@192.168.37.87") modparam("dispatcher", "ds_probing_mode", 0) modparam("dispatcher", "ds_probing_threshhold", 10) modparam("dispatcher", "ds_ping_interval", 30) modparam("dispatcher", "flags", 2) modparam("dispatcher", "force_dst", 1) [...] if (!ds_select_dst("1", "4")) { sl_send_reply("404", "No destination (disp)"); exit; } t_on_failure("failureroute"); t_on_reply("replyroute"); [...] failure_route[failureroute] { [...] if (t_check_status("[45][08]0") || (t_branch_timeout() && !t_branch_replied())) { ds_mark_dst("p"); if (!ds_next_dst()) { t_reply("404", "No destination (disp)"); exit; } t_on_failure("failureroute"); t_on_reply("replyroute"); } [...] } onreply_route [replyroute] { if (t_check_status("180|200") { # nice to have...... # ds_mark_dst("a"); } } Thanks in advance! regards, Klaus

2 4

Siremis Installation - Hangs at Database Configuration Page
by Tim Nelson 07 Mar '11

07 Mar '11

Greetings- I'm attempting an installation of Siremis on a CentOS 5.5 64-bit system running Kamailio 3.1.0. Following the instructions on the wiki [1], I find that I cannot pass step 3 for the database configuration. I've triple checked all of my details are correct and verified the permissions are set appropriately on the Siremis source dir. Also, I've successfully run 'make apache-config' and 'make prepare'. After clicking 'next', I simply get the 'Loading... please wait awhile' graphic. On three different fresh installs, I've waited for 3 hours each. By now, I have to assume it shouldn't be taking this long and there is an error somewhere. The Apache logs don't indicate anything of interest other than a missing favicon. Do you have any ideas? Thank you! --Tim [1] http://kb.asipto.com/siremis:install20:main

1 0

Limit call duration
by Hans Goossens 07 Mar '11

07 Mar '11

Hi, I'm wondering if there is a easy way to limit the call duration in Kamailio ? I would like to automaticaly end the call when the max. duration (for example 12 hours) is reached. In the feature overview of Kamailio 1.5 I found: "option to send BYE when dialog expires" but i'm can't figure out how to get this working (if it's still implemented at all in the current version). Thanks in advance, Kind regards Hans

2 2

about syn_branch
by Juha Heinanen 07 Mar '11

07 Mar '11

we noticed that if sr 3.1 config does not contain syn_branch=0 then acks to 200 oks that are t_relayed by sr, contain branch=0 param in topmost via. in kamailio 1.5, via branch param has proper value even when tm param syn_branch is not set. a couple of questions: 1) according to core cookbook, syn_branch core param only affects stateless forwarding. in my config, all acks are t_relayed, i.e., syn_branch param should not have any effect, but is does. how is that possible? is there a bug in core cookbook? 2) should t_relaying of acks to 200 oks use proper (i.e.. non 0) via branch param even when syn_branch is not set to any value in config? if not why? if yes, will branch=0 be used only when ack to 200 ok does not match any invite transaction? -- juha

2 3

How to access additional xavp's
by Alex Hermann 07 Mar '11

07 Mar '11

Hello, I'm currently toying with xavp's and have some trouble accessing the values. I want to have access to the xavp that isn't the last added one. From the wiki page on http://sip-router.org/wiki/devel/xavp I got the impression that indices are supported, but that doesn't seem to work. In the following fragment i want access to the values 1A & 1B, how to do that? $xavp(test=>a) = "1A"; $xavp(test[0]=>b) = "1B"; $xavp(test=>a) = "2A"; $xavp(test[0]=>b) = "2B"; xlog("$xavp(test[0]=>a)"); xlog("$xavp(test[0]=>b)"); xlog("$xavp(test[1]=>a)"); xlog("$xavp(test[1]=>b)"); pv_xavp_print(); Which gives the following result: Dec 21 14:30:35 veyron wsproxy1[10398]: ERROR: <script>: 2A Dec 21 14:30:35 veyron wsproxy1[10398]: ERROR: <script>: 2B Dec 21 14:30:35 veyron wsproxy1[10398]: ERROR: <script>: <null> Dec 21 14:30:35 veyron wsproxy1[10398]: ERROR: <script>: <null> Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:470]: +++++ XAVP list: 0xb3a38770 Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:473]: *** XAVP name: test Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:474]: XAVP id: 2063405720 Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:475]: XAVP value type: 6 Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:496]: XAVP value: <xavp:0xb3a387c4> Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:470]: +++++ XAVP list: 0xb3a387c4 Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:473]: *** XAVP name: b Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:474]: XAVP id: 110 Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:475]: XAVP value type: 2 Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:484]: XAVP value: 2B Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:473]: *** XAVP name: a Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:474]: XAVP id: 109 Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:475]: XAVP value type: 2 Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:484]: XAVP value: 2A Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:505]: ----- XAVP list Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:473]: *** XAVP name: test Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:474]: XAVP id: 2063405720 Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:475]: XAVP value type: 6 Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:496]: XAVP value: <xavp:0xb3a386c8> Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:470]: +++++ XAVP list: 0xb3a386c8 Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:473]: *** XAVP name: b Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:474]: XAVP id: 110 Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:475]: XAVP value type: 2 Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:484]: XAVP value: 1B Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:473]: *** XAVP name: a Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:474]: XAVP id: 109 Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:475]: XAVP value type: 2 Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:484]: XAVP value: 1A Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:505]: ----- XAVP list Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:505]: ----- XAVP list -- Greetings, Alex Hermann

2 4

Balancing Asterisk
by Andy Lippitt 07 Mar '11

07 Mar '11

Hello all, I've read as many of the asterisk balancing threads as I can find. Either my situation is unusual or I simply haven't understood anything I've read. In short, I'm building an web/phone mashup which uses Asterisk's AGI to get its work done. My only users are on the PSTN connected to Asterisk through a SIP trunk provider. So presently, in and out through the same trunk, apps live on the single Asterisk box. My goal is scaling and failover. I don't have any need for cross talk or transfers between the asterisk instances, and the algo's in dispatcher seem fine. It seems to me that I should be setting the sip-router up a replacement for the existing peer in Asterisk. What leaves me scratching my head is how I then register the sip-router with the upstream provider. Alternatively, if I use the sip-router as an outboundproxy from asterisk (which seems like it's going to take some hacking to make this work in 1.4), doesn't this now mean I have multiple UAC's trying to register for the same name? Can someone set me on the right track? Thanks, Andy Lippitt

2 1

Newly acquired SIP fails authorization from softphone
by Larry Baumbach 06 Mar '11

06 Mar '11

I am a newbie to the world of VIOP. I am attempting to set up an ATA with SIP. I created a SIP at account iptel.com and received an email confirmation stating "We are reserving the following SIP address for you: sip:larry.baumbach@iptel.org". I tried to testing this address in Xlite but got messages saying: "Account failed to enable. Account Iptel could not be enabled. Verify your user ID, password and authorization name. When I set up the SIP account in Xlite I used: UserID: larry.baumbach(a)iptel.org ( I also tried "larry.baumbach" & "sip:larry.baumbach@iptel.org") Domain: sip.iptel.org Password: my password Authorization name: (I left blank as I did not receive any) I can log into SERweb with the same UserID and Password and access my account. What am I doing wrong? Or is there some kind of wait time before my SIP address is activated? I have spent too much time trying to get this to work on my own. Thanks very much for any help you can provide. -- Larry Baumbach 920.358.8695 larry.baumbach(a)gmail.com

2 1

Re: [SR-Users] [sr-dev] planning release of v3.1.2
by Bernhard Suttner 06 Mar '11

06 Mar '11

Hi, whats about: http://sip-router.org/tracker/index.php?do=details&task_id=108 I think that some users are trying to register to an asterisk system. Or does somebody have a solution for that? A new release would be nice!!! Thanks in advance. Best regards, Bernhard ----- Original Message ----- From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] To: kamailio [mailto:sr-users@lists.sip-router.org], sr-dev [mailto:sr-dev@lists.sip-router.org] Sent: Fri, 28 Jan 2011 20:15:50 +0100 Subject: [sr-dev] planning release of v3.1.2 > Hello, > > I think it is time to release v3.1.2, first date that comes in my mind > is next Thursday if everyone feels it is enough time to take care of > backporting any fix he/she did and it is not yet there. That will > provide us a fresh release for the FOSDEM event. If not, then maybe the > other week, Tuesday, so the participants at the Kamailio Devel training > in Barcelona can practice on it. > > Soon after we should plan also a release for previous stable, branch 3.0. > > Anyone having other options? > > Thanks, > Daniel > > -- > Daniel-Constantin Mierla > http://www.asipto.com > > > _______________________________________________ > sr-dev mailing list > sr-dev(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev >

2 1

Kamailio auth_radius: duplicate User-Name attribute
by Kosilov Fedor 06 Mar '11

06 Mar '11

Hello List! I'm trying to set up authorization with our billing proprietary radius server, using Freeradius as a proxy. Currently I'm experiencing the following problem: The Access-Request packet, sent by Kamailio, contains two User-Name attribute records Here is a log from the Freeradius server: rad_recv: Access-Request packet from host 127.0.0.1 port 59294, id=112, length=298 User-Name = "2219001(a)example.com" Digest-Attributes = 0x0a0932323139303031 Digest-Attributes = 0x01106c696e6b2d726567696f6e2e7275 Digest-Attributes = 0x022254584452634531773045524b7368796f30684a70544f4f6a69424d386b32534a Digest-Attributes = 0x04147369703a6c696e6b2d726567696f6e2e7275 Digest-Attributes = 0x030a5245474953544552 Digest-Attributes = 0x050661757468 Digest-Attributes = 0x090a3030303030303031 Digest-Attributes = 0x080c32383034636535373032 Digest-Response = "e79b47955c02401fe52d05f7956609aa" Service-Type = Sip-Session Sip-Uri-User = "2219001" * User-Name = "call-id=domcmqmnychbwlp@koffe-work"* NAS-Identifier = "kamserv.example.com" NAS-Port = 5060 NAS-IP-Address = 127.0.0.1 # Executing section authorize from file /etc/freeradius/sites-enabled/default +- entering group authorize {...} ++[preprocess] returns ok ++[chap] returns noop ++[mschap] returns noop [digest] Checking for correctly formatted Digest-Attributes [digest] Digest-Attributes look OK. Converting them to something more usful. Digest-User-Name = "2219001" Digest-Realm = "example.com" Digest-Nonce = "TXDRcE1w0ERKshyo0hJpTOOjiBM8k2SJ" Digest-URI = "sip:example.com" Digest-Method = "REGISTER" Digest-QOP = "auth" Digest-Nonce-Count = "00000001" Digest-CNonce = "2804ce5702" [digest] Adding Auth-Type = DIGEST ++[digest] returns ok [suffix] Looking up realm "example.com" for User-Name = "2219001(a)example.com " [suffix] Found realm "example.com" [suffix] Adding Realm = "example.com" [suffix] Proxying request from user 2219001 to realm example.com [suffix] Preparing to proxy authentication request to realm "example.com" ++[suffix] returns updated [eap] No EAP-Message, not doing EAP ++[eap] returns noop ++[files] returns noop ++[expiration] returns noop ++[logintime] returns noop ++[pap] returns noop Sending Access-Request of id 250 to 127.0.0.1 port 1822 User-Name = "2219001(a)example.com" Digest-Attributes = 0x0a0932323139303031 Digest-Attributes = 0x01106c696e6b2d726567696f6e2e7275 Digest-Attributes = 0x022254584452634531773045524b7368796f30684a70544f4f6a69424d386b32534a Digest-Attributes = 0x04147369703a6c696e6b2d726567696f6e2e7275 Digest-Attributes = 0x030a5245474953544552 Digest-Attributes = 0x050661757468 Digest-Attributes = 0x090a3030303030303031 Digest-Attributes = 0x080c32383034636535373032 Digest-Response = "e79b47955c02401fe52d05f7956609aa" Service-Type = Sip-Session Sip-Uri-User = "2219001" * User-Name = "call-id=domcmqmnychbwlp@koffe-work"* NAS-Identifier = "kamserv.example.com" NAS-Port = 5060 NAS-IP-Address = 127.0.0.1 Proxy-State = 0x313132 Proxying request 1 to home server 127.0.0.1 port 1822 As I understand, this second User-Name attribute has to be a call-id attribute.

3 5

Calls are killed during kamailio restart
by Efelin Novak 04 Mar '11

04 Mar '11

Hi, I'd like to ask whether my situation is normal. During kamailio restart calls are dropped from mediaproxy. Those two programs are connected using kamailio.sock. Why RTP strem is dropped when SIP proxy is restarted? I would expect just undelivered BYE or something. Thanks for answer, Efelin

4 3

WARNING: <core> [db_id.c:281]: identical DB URLs, but different DB connection pid [30123/30107]
by Klaus Darilion 04 Mar '11

04 Mar '11

Any idea what does it mean? Where does it come from? Do I have to change my config? How? WARNING: <core> [db_id.c:281]: identical DB URLs, but different DB connection pid [30123/30107] Thanks Klaus

5 7

re-SUBSCRIBE and Record-Route
by marius zbihlei 04 Mar '11

04 Mar '11

Hello list, An UAC sends a resubscribe in the form SUBSCRIBE sip:Alice@example.com SIP/2.0 Route: <sip:sip.example.com;lr=on> Route:<sip:sip.next-hop.example.com;lr=on> .... The request has a to tag, from tag... Basic resubscribe(R-URI is the same as original SUBSCRIBE) The proxy that handles it is a Kamailio (1.5 but this refers to 3.1 as well). The problem is that loose_route() treats this request as coming from a strict routing router, not from the UAC, so it applies the strict-to-loose rules and rewrites the request as: SUBSCRIBE sip:sip.next-hop.example.com;lr=on SIP/2.0 Route: <sip:sip.example.com;lr=on> .... The segment that handles this is pretty standard: if (loose_route()) { # sequential request withing a dialog should take the path # determined by record-routing if(has_totag()) { forward(); exit; } It sees that message has Route set, it finds the to tag (so the routes were not a preset), and then , the important thing, sees that the domain from the RURI matches itself (kamailio has aliases configured for example.com), thus considering the request as coming from a strict router. Then it takes the bottom Route header and overwrites the RURI. What is the best way to ensure that loose routing is applied to the resubscribe as described above? Cheers, Marius

3 5

Re: [SR-Users] set_dlg_profile() in failure_route
by Timo Reimann 04 Mar '11

04 Mar '11

Hey 侯旭光, please keep SR-Users in CC so others can share in some assistance and benefit from a solution. On 04.03.2011 08:29, 侯旭光 wrote: > I did trace the dialog in the initial REQUEST > > my script: > ... > modparam("dialog","dlg_flag","4") > ... > route{ > > if (is_method("INVITE")){ > dlg_setflag("2"); > dlg_manage(); > ... > if(avp_db_load("$ruri/username","$avp(s:busyforward)")) > { > setflag(26); > t_on_failure(FAIL_ONE); > } > ... > } > failure_route[FAIL_ONE]{ > ... > if(isflagset(26)&&t_check_status("486")) > { > $ru=$avp(s:busyforward); > append_branch(); > dlg_setflag("2"); > avp_delete("$avp(s:busyforward)"); > resetflag("26"); > lookup("location"); > route(RELAY); > } > ... > } > route[RELAY]{ > if(!t_relay()){ > sl_reply_error(); > } > } > #----------------------------------------------------------------------# > > Is anything wrong? You set flag 2 but showed flag 4 to be modparam'ed to the dialog flag. Is that intentional? Anyways, since you are doing dlg_manage() in the INVITE transaction, you should still be fine. Just for the sake of debugging, can you set flag 4 in your 'is_method("INVITE")' block and see if it makes a difference? I have no in-depth experience with on demand dialog tracking as we use the dialog flag exclusively. By the way, I see no calls to set_dlg_profile() in your route logic. Is this really an excerpt of the configuration you are having trouble with? Cheers, --Timo > > 2011/3/3 Timo Reimann <timo.reimann(a)1und1.de>: >> Hi, >> >> >> On 03.03.2011 14:47, 侯旭光 wrote: >>> set_dlg_profile() function carshed in failure_route when I need do the >>> call-forwarding. >>> >>> When in busy-call-forward or no-answer-forward mode ,it'll run into >>> the failure_route.But at the time , the route_type of the dialog is >>> FAILURE_ROUTE and the get_current_dialog() return NULL, >>> >>> then the set_dlg_profile() function won't work. >>> >>> How can I set dialog profile in the busy or no-answer call-forwarding ? >>> >>> CRITIAL dialog [dlg_profile.c]: BUG - dialog not found in non > REQUEST route >>> ERROR dialog [dialog.c] : failed to set profile >> >> I think you did not track the dialog in the first place, i.e., when the >> initial request associated to the dialog you seek to profile was > processed. >> >> Are you using dlg_flag, and if so, have you enabled it in REQUEST_ROUTE? >> If you want to track dialogs on demand instead, try calling dlg_manage() >> prior to calling set_dlg_profile(). I'm not sure if this is the most >> elegant approach for your use case but it might at least work. >> >> >> Cheers, >> >> --Timo

1 1

set_dlg_profile() in failure_route
by 侯旭光 04 Mar '11

04 Mar '11

Hello set_dlg_profile() function carshed in failure_route when I need do the call-forwarding. When in busy-call-forward or no-answer-forward mode ,it'll run into the failure_route.But at the time , the route_type of the dialog is FAILURE_ROUTE and the get_current_dialog() return NULL, then the set_dlg_profile() function won't work. How can I set dialog profile in the busy or no-answer call-forwarding ? LOG file CRITIAL dialog [dlg_profile.c]: BUG - dialog not found in non REQUEST route ERROR dialog [dialog.c] : failed to set profile thanks

2 2

Re: [SR-Users] About RPM of Kamailio
by Ovidiu Sas 02 Mar '11

02 Mar '11

Please post your questions to the mailing list. If you want to build an rpm, the location of the spec file is available in the source tree: ./pkg/kamailio/rpm/kamailio.spec.CenOS If you prefer a link: http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blob;f=pkg/kama… Regards, Ovidiu Sas On Wed, Mar 2, 2011 at 5:56 AM, Suresh Bhandari <bring2sb(a)yahoo.com> wrote: > Hello Ovidiu, > > I searched through the kamailio.org/pub but can't find what you said. > > Can you verify the exact location of package that you've downloaded. > > Thanks. > Suresh > >

1 0

how to combine alias_db_lookup() with lookup()
by x-kamailio＠sidell.org 02 Mar '11

02 Mar '11

I'm trying to use the db_alias module as a way to define "generic" addresses that map to a set of actual phones. For example, I'd like the alias "home(a)foo.bar" to map to "kitchen(a)foo.bar" and "office(a)foo.bar", so that both phones ring when a call comes in to "home". I have set the append_branches param to 1: modparam("alias_db", "append_branches", 1) I also modified the dbaliases database table so that key "alias_idx" isn't unique, thereby allow me to add multiple rows for the same alias. The relevant script section is taken verbatim from 3.1 kamailio.cfg: # USER location service route[LOCATION] { #!ifdef WITH_ALIASDB # search in DB-based aliases alias_db_lookup("dbaliases"); #!endif if (!lookup("location")) { switch ($rc) { case -1: case -3: xlog( "L_WARN", "XXX $ru $fu\n"); t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } } When I place a call to an alias, the kamailio debug log shows that alias_db_lookup() is correctly setting the ruri to the first entry found in the table, and using append_branch() to add the others. But only the first matching phone gets an INVITE, not the others. I suspect that the lookup() call is blowing away the branches set up by alias_db_lookup() and replacing them with the single phone that matches the ruri for the first alias entry. Is there a way to get alias_db_lookup() and lookup() to play together, so that the first function can set up a list of branches, and the second function can resolve all of the branches to the actual device locations? -- Mark Sidell Partner Forte, Inc. 919-942-7068 fax 919-969-2844 www.forteinc.com

2 1

Radius authentication
by Pablo Ros 02 Mar '11

02 Mar '11

I am trying to authenticate through radius (info in LDAP database). I am using kamailio 3.1 First of all I would like to clear up an issue: As shown the way to do the authentication is done with: *Code:* if (! radius_www_authorize("uu.net")) { www_challenge("uu.net", "0"); return; } Ok, when I installed openser I did it with kamailio "flavour" so it was using the auth_radius module belonging to it. Does anyone know which would be the correct way to do the challenge to the user cause is was not working at all. The radius client does not even send it. I got stuck some time till I just decided to load the module from ser modules folder and then freeradius server started to get correctly my requests. Actually I can even see it authenticates them correctly. However Openser/Kamailio doesn't seem to see the same and it doesn't saves location. *Code:* route[AUTH] { #!ifdef WITH_AUTH # Primeramente comprobamos si pertenece a nuestro dominio; si no ya no hace falta comprobar credenciales. if (uri==myself) { if (is_method("REGISTER")) { xlog("L_NOTICE","KAM-INFO: r[AUTH] - REGISTER - User info: ($fu):($si)>\n"); if (!radius_www_authorize("i2cat.net")){ route(RADIUS); #www_challenge("i2cat.net","0"); exit; } } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { #!ifdef WITH_IPAUTH if(allow_source_address()) { # source IP allowed return; } #!endif # authenticate if from local subscriber if (from_uri==myself) { if (!proxy_authorize("$fd", "subscriber")) { proxy_challenge("$fd", "0"); exit; } if (is_method("PUBLISH")) { if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { if ($au!=$fU) { sl_send_reply("403","Forbidden auth ID"); exit; } } consume_credentials(); # caller authenticated } else { # caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (!uri==myself) { sl_send_reply("403","Not relaying"); exit; } } } #!endif return; } Before doing the challenge then it just goes throught: *Code:* route[RADIUS] { sl_send_reply("100", "Trying"); append_to_reply("Expires: 600\r\n"); append_to_reply("Min-Expires: 240\r\n"); xlog("L_NOTICE","KAM-INFO: RADIUS AUTHENTICATION - AUTHORIZING USER $fU - <$fu>:<$si>\n"); xlog("L_NOTICE","KAM-INFO: CHALLENGING. - RETCODE-> $rc \n"); www_challenge("i2cat.net", "0"); switch($rc){ case -5: xlog("L_INFO", "-> 500: internal server error"); sl_send_reply("500", "Internal Server Error"); case -4: xlog("L_INFO", "-> 404: credentials not found"); sl_send_reply("404", "Credentials Not Found"); case -3: xlog("L_INFO", "-> 400: bad request - stale nonce"); sl_send_reply("400", "Bad Request"); case -2: xlog("L_INFO", "-> 401: invalid password"); sl_send_reply("401", "Invalid Password"); case -1: xlog("L_INFO", "-> 401: invalid user"); sl_send_reply("401", "Invalid User"); default: xlog("L_INFO", "-> 401: unauthorized"); sl_send_reply("401", "Unauthorized"); } } Buuuuuuuuuuuuut... I got that in the debug of Kamailio: *Code:* 4(31099) DEBUG: auth [api.c:95]: auth: digest-algo: MD5 parsed value: 1 4(31099) DEBUG: auth_radius [sterman.c:271]: radius_authorize_sterman(): Success 4(31099) WARNING: auth_radius [authorize.c:89]: RADIUS server did not send SER-UID attribute in digest authentication reply 4(31099) DEBUG: auth [challenge.c:102]: build_challenge_hf: realm=' i2cat.net' 4(31099) DEBUG: auth [challenge.c:113]: build_challenge_hf: qop='auth' 4(31099) DEBUG: auth [challenge.c:236]: auth: 'WWW-Authenticate: Digest realm="i2cat.net", nonce="TWZJLk1mSAKFVzL0b+dVPzkuyyAnZHQs", qop="auth" I guess it has something to do with this SER-UID attribute and thus something about the dictonary? It is weird seeing that the radius server says 'ok' but then openser is not authenticating it. I need some clues! Thank you!. -- Pablo Ros

2 1

rtpproxy and connection information field
by Dominguez Jover, Ricardo 01 Mar '11

01 Mar '11

Hi all, I'm using rtpproxy 1.2.1 and kamailio. RTPproxy is working fine in almost every case. However I have a problem with some call. When softphone A using sip2sip.info account calls softphone B using my Kamailio server account, the Kamilio receives SIP packets from IP1 (81.23.228.129). The invite packet has IP2 (81.23.228.150) in the conecction information field (c=IN IP4 81.23.228.150): INVITE sip:12345@xxxxxxxx.xxx;transport=udp SIP/2.0 Record-Route: <sip:81.23.228.129;lr;ftag=d9e99adf;did=e81.3538c317> Via: SIP/2.0/UDP 81.23.228.129;branch=z9hG4bKd14b.12802794.0 Via: SIP/2.0/UDP 192.168.xx.xx:7964;received=88.xx.xx.xx;branch=z9hG4bK-d8754z-8ba1b4b47e37bb89-1---d8754z-;rport=7964 Max-Forwards: 69 Contact: <sip:54321@88.xxx.xxx.xxx:7964;transport=udp> To: "2205"<sip:12345@xxxxxxxx.xxx> From: "R sip2sip"<sip:54321@sip2sip.info>;tag=d9e99adf Call-ID: OGE5MTFiMzljZjE0NTYxN2M0N2VkZjgzNDRhMzE2ZTA. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces User-Agent:xxxxxxxxxxxxxxxxxxxxxxxxx Content-Length: 279 v=0 o=- 12943454020854250 1 IN IP4 192.168.xxx.xx s=xxxxxxxxxxxxxxxxxxxx c=IN IP4 81.23.228.150 t=0 0 m=audio 52854 RTP/AVP 107 0 8 18 101 a=rtpmap:107 BV32/16000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv Then Kamilio translates "c=" IP address to my RTPproxy IPaddress and sends the INVITE to softphone B. After the OK softphone B sends RTP packets to the RTPproxy as specified in the "c=" field, however the RTPproxy sends RTP packets to IP1 instead of send packets to IP2. How can I tell the RTPproxy to send packets to IP2 (the one in the c= field?) Thanks, Ricardo Dominguez

1 1

Segfault in Kamailio 1-3-4 to 1.5.5
by Andrew O. Zhukov 01 Mar '11

01 Mar '11

In continue of letters: Kamailio 1.5.5 No TLS Segmentation Fault After upgrade from openser 1.3.4 to kamailio 1.5.5 the same crash set Can someone from developers provide me commercial support to fix this bug in malloc module. If so, contact me directly.

6 12

modulo operator
by Klaus Darilion 01 Mar '11

01 Mar '11

Hi! Using kamailio 3.1.1, I failed to use '%' as described in the core cookbook. Using 'mod' instead seems to work. regards Klaus

2 3

Kamailio Installation
by Suresh Bhandari 01 Mar '11

01 Mar '11

Hello Community, I am new to Kamailio, and this list as well. I am trying to install Kamailio 3.1.2, but I am getting too many errors. I have fixed some but still not getting the way. I am using Red Hat Enterprise Linux (RHEL) 5, and /usr/local directory. When I ran the following command: make group_include="standard standard-dep mysql" include_modules="carrierroute peering" install it prompted not found error for the file docbookx.dtd, I found it (modules/auth/auth.xml, and modules_s/acc_syslog/acc_syslog.xml) and fixed it as it was errorenous URL location. For reference earlier it was http://www.oasis-open.org/docbookid/id/g/4.5/docbookx.dtd, which I changed to http://www.oasis-open.org/docbook/xml/4.5/docbookx.dtd. Now when I run the previous command again, I am getting the follwing errors: *nsgmls:<URL>http://www.oasis-open.org/docbook/xml/4.5/docbookx.dtd:116:17:E: "X20AC" is not a function name* If I ignore this, and continue, I am not able to find the "sip-router" service in /etc/init.d. The entire errors is attached here. Please help me solve the issue. TIA Suresh

2 1

no private memory left after mi mt_reload command
by Javier Gallart 28 Feb '11

28 Feb '11

Hi all we are running kamailio 3.1.1. Basically for every call it has to query a mtree that is reloaded every 15 minutes. The mtree table is about 4M records and is stored in a Pgostgres database. For the reload we use the command mi mt_reload embedded in a xmlrpc request. The code for this is: route[XMLRPC]{ if (method == "POST" || method == "GET") { if (search("^User-Agent:")) { xlog("xmlrpc request\n"); set_reply_close(); set_reply_no_connect(); # optional dispatch_rpc(); } exit; } } This server has been running with no issued for some weeks now, but at some point during the weekend the mi command started to fail. This is what the logs show: ERROR: <core> [db_res.c:181]: no private memory left ERROR: db_postgres [km_res.c:225]: could not allocate rows ERROR: db_postgres [km_dbase.c:280]: failed to convert rows ERROR: mtree [mtree_mod.c:627]: Error while fetching result ERROR: db_postgres [km_dbase.c:318]: invalid parameter value ERROR: mtree [mtree_mod.c:738]: cannot re-load info from database I've restarted kamailio and now it's working fine. Should we increase the private memory? On the other hand, we'd like to detect this failure condition on the script and act accordingly (for instance sending an email with an alarm message), is there anyway to do this? Thanks in advance Regards Javier

2 7

Re: [SR-Users] Can only tranfer (refer) a call one time
by Youngjin Park 28 Feb '11

28 Feb '11

> Hi, > > Can you tell me what advantage on INVITE has against REFER? > > REFER is a kind of blind transfer and INVITE in sipXbridge is 3PCC? > > Thanks in advance. > > Youngjin > > > On Sun, Feb 27, 2011 at 6:40 AM, Grzegorz Stanislawski < > stangrze(a)netitel.pl> wrote: > >> Hi. >> We have proverb for this, i don't know english version but it goes like >> this: >> "When it isn't known what it's all about, it's about money" >> >> Your ITSP had troubles with proper handling second transfer for billing >> purposes so decided to disable it. >> Proxy doesnt participate in call transfer, but ITSP it must charge users >> properly: Alice should pay just for call to Bob, Bob for "his" call to >> Charlie and so on. >> If You are using sipX You should use sipXbridge, it replaces REFER with >> INVITE and bridges calls. >> >> Grzegorz Stanislawski >> >> >> >> W dniu 2011-02-25 11:40, niklas rehnberg pisze: >> >>> Hi, >>> Thank for the quick response. >>> The issue occur only when the Alice is a PSTN client. >>> My ITSP says that they only supporting one call transfer (very strange). >>> They can not explain why etc... >>> PSTN client: Alice >>> MGW/MGC(ITSP): Cisco/SER >>> Our sip server: SIPX >>> >>> BR Niklas >>> 2011/2/25 Iñaki Baz Castillo <ibc(a)aliax.net <mailto:ibc@aliax.net>> >>> >>> 2011/2/25 niklas rehnberg <niklas.rehnberg(a)gmail.com >>> <mailto:niklas.rehnberg@gmail.com>>: >>> > Hi, >>> > Have following issue: >>> > >>> > Alice calling Bob. >>> > Bob make call transfer to Charlie (works fine) >>> > Charlie transfer Alice to David. (the call break) >>> > >>> > Why is not possible to transfer the call more than one time? >>> > Is it any parameters? >>> > >>> > My ITSP use SER together with Cisco MGW. >>> >>> Niklas, nothing in SIP protocol neither in SER/Kamailio makes your >>> scenario to fail. It must be a problem in your custom setup. Try >>> identifying the problem capturing SIP traces. >>> Also take into account that a proxy doesn't participate at all in the >>> process of a "call transfer". It's just a transparent mechanism for a >>> proxy. >>> >>> -- >>> Iñaki Baz Castillo >>> <ibc(a)aliax.net <mailto:ibc@aliax.net>> >>> >>> >>> >>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>> sr-users(a)lists.sip-router.org >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>> >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users(a)lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> > >

2 1

Re: [SR-Users] kamctl db exec
by Henning Westerholt 28 Feb '11

28 Feb '11

On Saturday 26 February 2011, 侯旭光 wrote: > But I need to use the feature, is it aviliable or not? Hi 侯旭光, ok.. Have you tried if its work with an simple example? kamctl exec "show databases" Cheers, Henning

1 0

UC Expo 2011, London, next week
by Daniel-Constantin Mierla 28 Feb '11

28 Feb '11

Hello, I will be visiting the first day of UC Expo in London (http://www.ucexpo.co.uk/), Tuesday, March 8. If you are going to be there and want to meet for a chat about Kamailio & VoIP, drop me an email. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

Re: [SR-Users] Unable to start Siremis
by jack＠4voice.net 28 Feb '11

28 Feb '11

1 0

Re: [SR-Users] Unable to start Siremis
by hb 28 Feb '11

28 Feb '11

Hi jack： I got the same problem with you. I know little about apache and HTML, would you tell me how to configuration apache specific. Thanks a lot. haibo huang

1 0

how to route to new URI on failure
by x-kamailio＠sidell.org 27 Feb '11

27 Feb '11

I'm trying to route failed calls to a voicemail URI. The failure route couldn't be simpler: failure_route[FAIL_ONE] { if (t_is_canceled()) { exit; } if( t_check_status("486|408") { append_branch( "sip:foo@2.2.2.2"); t_relay(); } } But, it doesn't work. For example, let's say the initial INVITE resolves to a local device "me(a)1.1.1.1". This works fine, and the phone rings. After a timeout, the failure_route executes. The branch "foo(a)2.2.2.2" gets appended, and kamailio sends a new INVITE, but instead of determining the correct proxy for the new address, it sends the INVITE, with the new URI, to the device that original received the INVITE, "me(a)1.1.1.1". Obviously, this doesn't work. I've been able to force kamailio to route the call correctly by modifying failure_route[ to use t_relay_to_udp() as follows: failure_route[FAIL_ONE] { if (t_is_canceled()) { exit; } if( t_check_status("486|408") { append_branch( "sip:foo@2.2.2.2"); t_relay_to_udp( "2.2.2.2", "5060"); } } But, it seems like kamailio should figure what where to send the new INVITE itself. What am I doing wrong? -- Mark Sidell Partner Forte, Inc. 919-942-7068 fax 919-969-2844 www.forteinc.com

2 2

Can only tranfer (refer) a call one time
by niklas rehnberg 27 Feb '11

27 Feb '11

Hi, Have following issue: Alice calling Bob. Bob make call transfer to Charlie (works fine) Charlie transfer Alice to David. (the call break) Why is not possible to transfer the call more than one time? Is it any parameters? My ITSP use SER together with Cisco MGW. BR Niklas

4 4

Lcr Command Disabled
by Amit Nepal 25 Feb '11

25 Feb '11

Lcr is working fine but Why would i get LCR commands disabled ? kamctl lcr reload ERROR: command disabled-- Thank You Amit Nepal Systems Administrator Phoenix Internet Phone: 602-385-0731 602-234-0917#112 http://www.phoenixinternet.net

3 8

Rtpproxy newbie question
by Marc 25 Feb '11

25 Feb '11

Hi, I installed a kamailio from the debian repository, i use the default configuration. The only enable the tcp and tls transport modes, and configured mysql login. Everyting seems to be working fine. I would need now to configure rtprpoxy, but i just cant make it work. My goal is to force rtpproxy for all calls. Any help is highly appreciated!! Thank you Marc Rtpproxy is installed and seems to be running: This is my rtpproxy configuration: ------------ # Defaults for rtpproxy # The control socket. #CONTROL_SOCK="unix:/var/run/rtpproxy/rtpproxy.sock" # To listen on an UDP socket, uncomment this line: CONTROL_SOCK=udp:127.0.0.1:22222 LISTEN_ADDR=[external_ip] # Additional options that are passed to the daemon. EXTRA_OPTS="-l ${LISTEN_ADDR}" /usr/sbin# ------------- It seems to be running: ----------- /etc/init.d# ./rtpproxy start Starting RTP relay: rtpproxy. /etc/init.d# netstat -an|grep 22222 udp 0 0 127.0.0.1:22222 0.0.0.0:* /etc/init.d# ----------------- Here is my kamailio configuration, default as is installed: ------------------------------------------- #!KAMAILIO # # Kamailio (OpenSER) SIP Server v3.1 - default configuration script # - web: http://www.kamailio.org # - git: http://sip-router.org # # Direct your questions about this file to: <sr-users(a)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 # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!define ""mysql:[my_mysql_db_config]"" "mysql://openser:openserrw@localhost/openser" #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=2 log_stderror=no #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_DAEMON fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ disable_tcp=no /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ #auto_aliases=no /* add local domain aliases */ /* 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:[external_ip]:5060 listen=tcp:[external_ip]:5062 listen=tls:[external_ip]:5061 /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=5060 port=5062 tls_port_no=5061 enable_tls=yes #!ifdef WITH_TLS enable_tls=yes #!endif ####### Custom Parameters ######### # These parameters can be modified runtime via RPC interface # - see the documentation of 'cfg_rpc' module. # # Format: group.id = value 'desc' description # Access: $sel(cfg_get.group.id) or @cfg_get.group.id # #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting #pstn.gw_ip = "" desc "PSTN GW Address" #!endif ####### Modules Section ######## # set paths to location of modules #!ifdef LOCAL_TEST_RUN mpath="modules_k:modules" #!else mpath="/usr/lib/kamailio/modules_k/:/usr/lib/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif 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 "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "mi_rpc.so" loadmodule "acc.so" loadmodule "tls.so" modparam("tls", "config", "/etc/kamailio/tls.cfg") #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- 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) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", ""[my_mysql_db_config]"") modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", "[my_mysql_db_config]") modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", ""[my_mysql_db_config]"") modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", "[my_mysql_db_config]") modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", "[my_mysql_db_config]") modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", "[my_mysql_db_config]") # use caching modparam("domain", "db_mode", 1) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", "[my_mysql_db_config]") # ----- presence_xml params ----- modparam("presence_xml", "db_url", "[my_mysql_db_config]") modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:22222") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@[mydomain]") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") modparam("tls", "tls_debug", 10) #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route route { # per request initial checks route(REQINIT); # NAT detection route(NAT); # 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(); # authentication route(AUTH); # 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(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # 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 to PSTN route(PSTN); # user location service route(LOCATION); route(RELAY); } route[RELAY] { #!ifdef WITH_NAT if (check_route_param("nat=yes")) { setbflag(FLB_NATB); } if (isflagset(FLT_NATS) || isbflagset(FLB_NATB)) { route(RTPPROXY); } #!endif /* example how to enable some additional event routes */ if (is_method("INVITE")) { #t_on_branch("BRANCH_ONE"); t_on_reply("REPLY_ONE"); t_on_failure("FAIL_ONE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); 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(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... 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() ) { # no loose-route, but stateful ACK; # must be an 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")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); exit; } } # USER location service route[LOCATION] { #!ifdef WITH_ALIASDB # search in DB-based aliases alias_db_lookup("dbaliases"); #!endif if (!lookup("location")) { switch ($rc) { case -1: case -3: t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("$td", "subscriber")) { www_challenge("$td", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { #!ifdef WITH_IPAUTH if(allow_source_address()) { # source IP allowed return; } #!endif # authenticate if from local subscriber if (from_uri==myself) { if (!proxy_authorize("$fd", "subscriber")) { proxy_challenge("$fd", "0"); exit; } if (is_method("PUBLISH")) { if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { if ($au!=$fU) { sl_send_reply("403","Forbidden auth ID"); exit; } } consume_credentials(); # caller authenticated } else { # caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (!uri==myself) { sl_send_reply("403","Not relaying"); exit; } } } #!endif return; } # Caller NAT detection route route[NAT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[RTPPROXY] { #!ifdef WITH_NAT if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ force_rtp_proxy(); } if (!has_totag()) add_rr_param(";nat=yes"); #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # Sample branch router branch_route[BRANCH_ONE] { xdbg("new branch at $ru\n"); } # Sample onreply route onreply_route[REPLY_ONE] { xdbg("incoming reply\n"); #!ifdef WITH_NAT if ((isflagset(FLT_NATS) || isbflagset(FLB_NATB)) && status=~"(183)|(2[0-9][0-9])") { force_rtp_proxy(); } if (isbflagset("6")) { fix_nated_contact(); } #!endif } # Sample failure route failure_route[FAIL_ONE] { #!ifdef WITH_NAT if (is_method("INVITE") && (isbflagset(FLB_NATB) || isflagset(FLT_NATS))) { unforce_rtp_proxy(); } #!endif if (t_is_canceled()) { 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; ##} # uncomment the following lines if you want to redirect the failed # calls to a different new destination ##if (t_check_status("486|408")) { ## sethostport("192.168.2.100:5060"); ## append_branch(); ## # do not set the missed call flag again ## t_relay(); ##} }

2 2

postgres in Kamailio 3.1.2
by Bruno Bresciani 24 Feb '11

24 Feb '11

Hi, I configure the Kamailio 3.1.2 with postgres but I cann't start. In the log file is generated the following error: ** ERROR: load_module: could not open module </home2/local/kamailio/lib/kamailio/modules/db_ postgres.so>: /home2/local/kamailio/lib/kamailio/modules/db_postgres.so: undefined symbol: PQdescribePrepared Why this error is generated? Best Regards

2 1

tcp_persistent_flag
by Bruno Bresciani 24 Feb '11

24 Feb '11

Hi, there is in kamailio 3.1.2 some exported parameter that work as tcp_persistent_flag paramenter? I want to keep alive the connections TCP/TLS as long as the register contact is valid. Best Regards

2 1

dialplan regex confusion
by thrillerbee 24 Feb '11

24 Feb '11

I'm using the following in dialplan: id | dpid | pr | match_op | match_exp | match_len | subst_exp | repl_exp | attrs ----+------+----+----------+----------------------------------------+-----------+----------------------------------------------+----------+---------------------------------- 3 | 3 | 0 | 1 | ^30\d{3}#\+?1?[2-9]\d{2}[2-9]\d{6}$ | 0 | ^(30\d{3}#)(\+?1?)([2-9]\d{2}[2-9]\d{6}$) | \11\3 | normalize prefixed domestic A R-URI goes in as 30001#12345678910 It comes out as 30001#*\1*12345678910 Can someone help me understand why? I'd expect to be exactly what it was originally but for some reason '\1' is thrown in the middle... Thanks, Ryan

2 2

pipelimit db schema
by thrillerbee 24 Feb '11

24 Feb '11

Can anyone point me to the db schema for the new pipelimit module? Thanks, Ryan

3 6

kamctl db exec
by 侯旭光 24 Feb '11

24 Feb '11

Hello How to use "kamctl db exec <query> " function? My SQL_ENGINE is MySQL How to write the command line when I want to do the query "select * from acc" with kamctl db tool? thanks

2 1

NAT Traversal
by Spinov Evgeniy 24 Feb '11

24 Feb '11

Hello, all. Using nathelper + rtpproxy for subj. Kamailio has public and private network interfaces. Asterisk is only private. RTP Proxy is working in bridge mode and relaying traffic from UAC to Asterisks. Everything is working fine, except one configuration. When the client is behind router ( a specific one, I do not have an access there to check ), and this UAC is making a call to other public extension, which is behind router, then RTP Proxy is relaying traffic to the caller, using another UDP port, then the packets arrive. For instance: UAC 1 -> UAC 2 PUBLIC_IP:10 > KAMAILIO_IP:5555 KAMAILIO_IP:5678 > PUBLIC_IP:12 While for the UAC 2 it looks like: PUBLIC_IP:20 > KAMAILIO_IP:6767 KAMAILIO_IP:4564 > PUBLIC_IP:20 The source and destination UDP ports are the same. As result, I can hear UAC 1 and he cannot hear me. In case of we have UAC 3, which is behind other router, call is working fine with same configuration. "It's routers fault" you can say, but in the same configuration ( I mean network, not kamailio ) it worked, but when RTPProxy was not in bridge mode and Kamailio and Asterisks were in public network. Reinvites are not allowed in both cases. The question is, why the source and destination UDP ports are different? Using STUN in first case, cause without it, private IP written in contacts and as result, traffic relayed from Kamailio is incorrect, cause heading to private network which is unreachable. Any ideas where to dig?

1 0

Rtpproxy newbie question
by Marc 24 Feb '11

24 Feb '11

Hi, I installed a kamailio from the debian repository, i use the default configuration. The only enable the tcp and tls transport modes, and configured mysql login. Everyting seems to be working fine. I would need now to configure rtprpoxy, but i just cant make it work. My goal is to force rtpproxy for all calls. Any help is highly appreciated!! Thank you Marc Rtpproxy is installed and seems to be running: This is my rtpproxy configuration: ------------ # Defaults for rtpproxy # The control socket. #CONTROL_SOCK="unix:/var/run/rtpproxy/rtpproxy.sock" # To listen on an UDP socket, uncomment this line: CONTROL_SOCK=udp:127.0.0.1:22222 LISTEN_ADDR=[external_ip] # Additional options that are passed to the daemon. EXTRA_OPTS="-l ${LISTEN_ADDR}" /usr/sbin# ------------- It seems to be running: ----------- /etc/init.d# ./rtpproxy start Starting RTP relay: rtpproxy. /etc/init.d# netstat -an|grep 22222 udp 0 0 127.0.0.1:22222 0.0.0.0:* /etc/init.d# ----------------- Here is my kamailio configuration, default as is installed: ------------------------------------------- #!KAMAILIO # # Kamailio (OpenSER) SIP Server v3.1 - default configuration script # - web: http://www.kamailio.org # - git: http://sip-router.org # # Direct your questions about this file to: <sr-users(a)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 # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!define ""mysql:[my_mysql_db_config]"" "mysql://openser:openserrw@localhost/openser" #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=2 log_stderror=no #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_DAEMON fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ disable_tcp=no /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ #auto_aliases=no /* add local domain aliases */ /* 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:[external_ip]:5060 listen=tcp:[external_ip]:5062 listen=tls:[external_ip]:5061 /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=5060 port=5062 tls_port_no=5061 enable_tls=yes #!ifdef WITH_TLS enable_tls=yes #!endif ####### Custom Parameters ######### # These parameters can be modified runtime via RPC interface # - see the documentation of 'cfg_rpc' module. # # Format: group.id = value 'desc' description # Access: $sel(cfg_get.group.id) or @cfg_get.group.id # #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting #pstn.gw_ip = "" desc "PSTN GW Address" #!endif ####### Modules Section ######## # set paths to location of modules #!ifdef LOCAL_TEST_RUN mpath="modules_k:modules" #!else mpath="/usr/lib/kamailio/modules_k/:/usr/lib/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif 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 "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "mi_rpc.so" loadmodule "acc.so" loadmodule "tls.so" modparam("tls", "config", "/etc/kamailio/tls.cfg") #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- 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) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", ""[my_mysql_db_config]"") modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", "[my_mysql_db_config]") modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", ""[my_mysql_db_config]"") modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", "[my_mysql_db_config]") modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", "[my_mysql_db_config]") modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", "[my_mysql_db_config]") # use caching modparam("domain", "db_mode", 1) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", "[my_mysql_db_config]") # ----- presence_xml params ----- modparam("presence_xml", "db_url", "[my_mysql_db_config]") modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:22222") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@[mydomain]") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") modparam("tls", "tls_debug", 10) #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route route { # per request initial checks route(REQINIT); # NAT detection route(NAT); # 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(); # authentication route(AUTH); # 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(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # 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 to PSTN route(PSTN); # user location service route(LOCATION); route(RELAY); } route[RELAY] { #!ifdef WITH_NAT if (check_route_param("nat=yes")) { setbflag(FLB_NATB); } if (isflagset(FLT_NATS) || isbflagset(FLB_NATB)) { route(RTPPROXY); } #!endif /* example how to enable some additional event routes */ if (is_method("INVITE")) { #t_on_branch("BRANCH_ONE"); t_on_reply("REPLY_ONE"); t_on_failure("FAIL_ONE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); 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(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... 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() ) { # no loose-route, but stateful ACK; # must be an 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")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); exit; } } # USER location service route[LOCATION] { #!ifdef WITH_ALIASDB # search in DB-based aliases alias_db_lookup("dbaliases"); #!endif if (!lookup("location")) { switch ($rc) { case -1: case -3: t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("$td", "subscriber")) { www_challenge("$td", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { #!ifdef WITH_IPAUTH if(allow_source_address()) { # source IP allowed return; } #!endif # authenticate if from local subscriber if (from_uri==myself) { if (!proxy_authorize("$fd", "subscriber")) { proxy_challenge("$fd", "0"); exit; } if (is_method("PUBLISH")) { if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { if ($au!=$fU) { sl_send_reply("403","Forbidden auth ID"); exit; } } consume_credentials(); # caller authenticated } else { # caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (!uri==myself) { sl_send_reply("403","Not relaying"); exit; } } } #!endif return; } # Caller NAT detection route route[NAT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[RTPPROXY] { #!ifdef WITH_NAT if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ force_rtp_proxy(); } if (!has_totag()) add_rr_param(";nat=yes"); #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # Sample branch router branch_route[BRANCH_ONE] { xdbg("new branch at $ru\n"); } # Sample onreply route onreply_route[REPLY_ONE] { xdbg("incoming reply\n"); #!ifdef WITH_NAT if ((isflagset(FLT_NATS) || isbflagset(FLB_NATB)) && status=~"(183)|(2[0-9][0-9])") { force_rtp_proxy(); } if (isbflagset("6")) { fix_nated_contact(); } #!endif } # Sample failure route failure_route[FAIL_ONE] { #!ifdef WITH_NAT if (is_method("INVITE") && (isbflagset(FLB_NATB) || isflagset(FLT_NATS))) { unforce_rtp_proxy(); } #!endif if (t_is_canceled()) { 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; ##} # uncomment the following lines if you want to redirect the failed # calls to a different new destination ##if (t_check_status("486|408")) { ## sethostport("192.168.2.100:5060"); ## append_branch(); ## # do not set the missed call flag again ## t_relay(); ##} }

1 0

403 "Not Relaying" problem
by yan wang 24 Feb '11

24 Feb '11

Dear Friends, With Kamailio 3.1.2, after successfully registering, the caller sends INVITE to the proxy, and is always responded with 403 not relaying. I found that in the route[AUTH] subroutine, like below. Regarding the red comparing, it failed. I also check the code, and found Kamailio will use the host name in the From header to compare all the socket's IP address, such as host = mysipserver.com, but socket's ip = 127.0.0.1 or 10.1.1.5. As you can see, this comparing will never be equal.Could anyone show me how to handle this problem? Thanks a lot! # authenticate if from local subscriber if (from_uri==myself) { if (!proxy_authorize("$fd", "subscriber")) { proxy_challenge("$fd", "0"); exit; } if (is_method("PUBLISH")) { if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { if ($au!=$fU) { sl_send_reply("403","Forbidden auth ID"); exit; } } consume_credentials(); # caller authenticated } else { # caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (!uri==myself) { sl_send_reply("403","Not relaying"); exit; } } Best Regards,Spencer

1 0

how to forward on busy
by x-kamailio＠sidell.org 23 Feb '11

23 Feb '11

Is there an example somewhere that shows how to do forward-on-busy, starting with the kamailio.cfg from version 3.1? Ultimately, I want to use a per-user AVP to obtain the forwarding URI, but just an example that shows how to forward to a fixed URI from the failure route would be great. -- Mark Sidell Partner Forte, Inc. 919-942-7068 fax 919-969-2844 www.forteinc.com

1 0

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