- sr-users - kamailio.org

Core db_text module
by Bruno Bresciani 30 Sep '11

30 Sep '11

Hi All, Kamailio 3.1.2 generate a core in db_text module when subscriber table is updated and after this action some SIP message require authentication process... Someone Can Help me to understand why this core is happening? Below is part of kamailio's trace and gdb core too. Kamailio's trace: *Sep 22 16:35:54 sswpst00 /home2/local/kamailio/sbin/kamailio[2327]: DEBUG: auth_db [authorize.c:239]: realm value [192.168.166.199]* Sep 22 16:35:54 sswpst00 /home2/local/kamailio/sbin/kamailio[… [View More]

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Re: [SR-Users] simple question but can't find the answer
by Daniel-Constantin Mierla 30 Sep '11

30 Sep '11

Hello, On 9/30/11 12:11 AM, JJ Zeng wrote: > > Thanks Daniel. I followed your advice and install kamailio 3.1. Now > I get a weird problem: it works 1 out 5 times for the relaying. > try to set like tcp_connection_lifetime=3604 global parameter, or to a value higher than the longer registration time. > > The t_relay is where the the problem is, because I check the response > code and route(LOCATION) always succeeds, which means the entries > exist in "location". … [View More]

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Re: [SR-Users] Placing ds_select_dst in kamailio.cfg, where?
by Daniel-Constantin Mierla 29 Sep '11

29 Sep '11

Hello, please keep the mainling list cc-ed. Calling ds_select_dst() first in the main route block is not really a good place. First you have to handle sanity checks, within dialog requests a.s.o. Of course it is a matter of your purpose, but as general hint, selecting the destination should be done just (or as close as possible) before relaying. If you take default config and you have only load balancing, then you can replace location lookup part with load balancing. Other parts may be … [View More]stripped as well, just giving some direction here. Cheers, Daniel On 9/28/11 11:11 PM, Henrik Aagaard Sørensen wrote: > That did the job. > > I just placed the ds_select_dst-code at the very first thing after route{ > > Thank you for your help! > > 2011/9/28 Daniel-Constantin Mierla <miconda(a)gmail.com > <mailto:miconda@gmail.com>> > > Hello, > > the devel version of dispatcher module README has a different > config example, maybe it helps you more: > > http://kamailio.org/docs/modules/devel/modules_k/dispatcher.html#id2522847 > > I am considering to update it again soon to be a bit more complete > and follow the structure of the default config file shipped with > kamailio. > > Cheers, > Daniel > > > On 9/28/11 6:30 PM, Henrik Aagaard Sørensen wrote: >> Hi Carsten. >> >> The configuration works. It's actually where (which line) in the >> configuration I should put: >> ds_select_dst("1", "0"); >> >> :) I'm not strong in Kamailio and just want it to load balance >> between Freswitch servers. So I have a basic installation of >> Kamailio, without MySQL, NAT, Auth etc. >> >> On Wed, Sep 28, 2011 at 5:09 PM, Carsten Bock >> <carsten(a)ng-voice.com <mailto:carsten@ng-voice.com>> wrote: >> >> Hi, >> >> When you've got problems with placing "ds_select_dst()" in your >> config, you should do the following: >> - Open an Editor of your choice >> - go to the line, where you want to place it >> - type: ds_select_dst("1", "0"); >> - save & exit >> >> Just kidding! You should do the following with your config: >> kamailio -f <path-to-your-config> -c >> >> What error is displayed? If none, then check your syslog for >> errors. >> Could you post your config? It is better to put a full path >> in the >> config to the dispatcher-file... otherwise it might be >> dependent of, >> where you are running the init.d-skript/kamailio.... >> >> I am using dispatcher in Kamailio 1.5, 3.1 and 3.2, >> everythings works >> fine with me. >> >> Kind regards, >> Carsten >> >> 2011/9/28 Henrik Aagaard Sørensen >> <henrikaagaardsorensen(a)gmail.com >> <mailto:henrikaagaardsorensen@gmail.com>>: >> > I've installed 2 Freeswitch on different servers and would >> like to have >> > Kamailio load balance between them. >> > >> > I've installed Kamailio following: >> > >> http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.1.x-from-git >> > >> > I've created dispatcher.list and added: >> > >> > loadmodule "dispatcher.so" >> > >> > modparam("dispatcher", "list_file", "../etc/dispatcher.list") >> > >> > in the kamailio.cfg. >> > >> > But looking at: >> > >> http://kamailio.org/docs/modules/3.1.x/modules_k/dispatcher.html >> > I'm having difficulties to placing: >> > >> > ds_select_dst("1", "0"); >> > >> > in the kamailio.cfg that was created originally. >> > >> > Can anyone help me? >> > >> > _______________________________________________ >> > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users >> mailing list >> > sr-users(a)lists.sip-router.org >> <mailto:sr-users@lists.sip-router.org> >> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> > >> > >> >> >> >> -- >> Carsten Bock >> CEO (Geschäftsführer) >> >> ng-voice GmbH i. Gr. >> Schomburgstr. 80 >> D-22767 Hamburg / Germany >> >> http://www.ng-voice.com >> mailto:carsten@ng-voice.com <mailto:carsten@ng-voice.com> >> >> Mobile +49 179 2021244 <tel:%2B49%20179%202021244> >> Office +49 40 34927219 <tel:%2B49%2040%2034927219> >> Fax +49 40 34927220 <tel:%2B49%2040%2034927220> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users >> mailing list >> sr-users(a)lists.sip-router.org >> <mailto:sr-users@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 <mailto:sr-users@lists.sip-router.org> >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > -- > Daniel-Constantin Mierla --http://www.asipto.com > Kamailio Advanced Training, Oct 10-13, Berlin:http://asipto.com/u/kat > http://linkedin.com/in/miconda -- http://twitter.com/miconda > > -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda [View Less]

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Re: [SR-Users] Logmessage when Dispatcher-Module sets a dispatcher to probing
by Timo Klecker 29 Sep '11

29 Sep '11

Hey again, Btw: I am using Kamailio 3.1.5 and found the event_route in 3.2.x Dispatcher documentation. Is there a possibility in 3.1.x? Greetings Timo Von: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org] Im Auftrag von Timo Klecker Gesendet: Donnerstag, 29. September 2011 11:10 An: 'kamailio users' Betreff: [SR-Users] Logmessage when Dispatcher-Module sets a dispatcher to probing Hey, I'd like to have a log-entry in the kamailio … [View More]

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simple question but can't find the answer
by JJ Zeng 29 Sep '11

29 Sep '11

Hi, I am really new to ser(just installed it today). My situation is here: Both A and B make tcp connection to ser proxy server. If A sends "Invite" to B, ser would look up the tcp connection of B, and relay the sip request. If B is not connected, then 480 is returned to A. When B sends back the response, ser would relay it to A. It looks really simple, but I just can't figure out how to do it. Thanks. JJ

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Logmessage when Dispatcher-Module sets a dispatcher to probing
by Timo Klecker 29 Sep '11

29 Sep '11

Hey, I'd like to have a log-entry in the kamailio logfile when #ds_probind_threshhold OPTIONS pings fail to reach a dispatcher (408 localy generated after #fr_timer ms). Is it possible to do this without changing code? Greetings Timo

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OpenIMS
by Carlos Ruiz Diaz 29 Sep '11

29 Sep '11

Hello list. I'm trying to test/implement http://www.openimscore.org/ to setup an IMS scenario for development purposes. The project seems quite inactive to me and I wonder if there is any other similar project or this one just died and is no longer maintained. The documentation didn't specify if there is any kind of compatibility with Kamailio and based on what was written, only SER core architecture is supported. Any help will be appreciated. Regards. Carlos.

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documentation update for 3.2.0 and wiki system
by Daniel-Constantin Mierla 28 Sep '11

28 Sep '11

Hello, in preparation for release 3.2.0, I was checking our wiki (on kamailio.org), which is quite rich in information, but some tutorials are for older versions. Maybe it is also a good time to do some restructuring along with the update of the useful tutorials to be valid for 3.2.0. For restructuring, the main goal is to provide some ways to let people find the documentation they need (and it is available there) easier. Some ideas, introducing categorization/tags for specific topics (… [View More]

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Placing ds_select_dst in kamailio.cfg, where?
by Henrik Aagaard Sørensen 28 Sep '11

28 Sep '11

I've installed 2 Freeswitch on different servers and would like to have Kamailio load balance between them. I've installed Kamailio following: http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.1.x-from-git I've created dispatcher.list and added: loadmodule "dispatcher.so" modparam("dispatcher", "list_file", "../etc/dispatcher.list") in the kamailio.cfg. But looking at: http://kamailio.org/docs/modules/3.1.x/modules_k/dispatcher.html I'm having difficulties to placing: … [View More]

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Sip-router gentoo ebuild v3.1.5
by caio 28 Sep '11

28 Sep '11

Hello, Just want to announce the availability of an ebuild for version 3.1.5 (kamailio and ser flavours) on Gentoo Portage Overlays. Here the git link: * http://git.overlays.gentoo.org/gitweb/?p=user/caio.git;a=tree;f=net-misc/si… The unofficial overlays portage website: * http://gpo.zugaina.org/net-misc/sip-router Important, this is not an official ebuild, and neither it's not at official gentoo portage tree. You can use *layman* to add this overlay. Also at pkg/ser/gentoo path of sip-… [View More]

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Accounting only the 2nd branch of missed serial forked call
by Ozren Lapcevic 27 Sep '11

27 Sep '11

Hi, I'm having some problems accounting missed serial forked calls to mysql database. I have following setup. Each user can have up to two contacts: telephone number (routed to asterisk) and SIP URI. Users can specify which contact has higher priority - which one should ring first. There is also SEMS voicemail which is forked as 3rd serial call leg if there is no answer at first two contacts. For example, I have two users: oz(a)abc.hr and pero(a)abc.hr. pero(a)abc.hr also has set telephone … [View More]number as alternative number if he is not reachable at sip:pero@abc.hr. Moreover, pero(a)abc.hr has voicemail turned on. When oz(a)abc.hr calls pero(a)abc.hr, first pero(a)abc.hr's SIP client rings, then if there is no answer and after the timeout telephone number rings and finally, if there is no answer at telephone and after the timeout INVITE is forked to SEMS. There are two interesting scenarios accounting-wise which can happened: 1. oz(a)abc.hr calls pero(a)abc.hr, there are no answers and call is forked to voicemail. 2. oz(a)abc.hr calls pero(a)abc.hr, there is no answer at SIP client, but pero answers call at telephone. When scenario 1 happens, I want to have only one log (row) in missed_calls table. When scenario 2 happens, I don't want to have a log in missed_calls table. To accomplish this,* I want to log only the 2nd branch of the forked call. However, there is either a bug in acc module or I'm doing something wrong, and I can't get Kamailio to log only the 2nd branch*. I think that I am setting the FLT_ACCMISSED flag correctly - after the 2nd branch is handled and prior to calling the RELAY route. Logs show that FLT_ACCMISSED flag is set prior to calling t_relay(), and there are no errors in debug log. I am using $ru = "something" to rewrite URI prior to forking request. I can easily set up logging of every call (two missed calls for serially forked call to two locations) by setting FLT_ACCMISSED flag for each INVITE. I can set up logging of every call's 1st branch, by reseting FLT_ACCMISSED flag when handling 2nd branch of the call. Interestingly, logging of only the 2nd branch of the serial forked call works when there is no forking to voicemail! Any ideas how to solve this problem? Bellow are important parts of my config file. I'm running kamailio 3.1.4. Cheers Ozren # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 0) modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) modparam("acc", "log_extra","src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;dst_user=$tU;dst_domain=$td;src_ip=$si") #!endif ... # Main SIP request routing logic # - processing of any incoming SIP request starts with this route route { # per request initial checks route(REQINIT); if (src_ip != ****) { # NAT detection route(NAT); } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); if ( is_method("INVITE") ) { route(DBALIASES); #check for user defined forking priorities and timers route(FORK); } # user location service route(LOCATION); route(RELAY); } #check for user defined forking priorities and timers route[FORK]{ sql_query("con", "select * from usr_pref_custom where uuid='$tu'", "pref"); $avp(uuid)=$dbr(pref=>[0,0]); $avp(email)=$dbr(pref=>[0,1]); $avp(prio1)=$dbr(pref=>[0,2]); $avp(prio2)=$dbr(pref=>[0,3]); $avp(timer1)=$dbr(pref=>[0,5]); $avp(timer2)=$dbr(pref=>[0,6]); if (strlen($avp(prio1))>5) { # user has multiple contacts, do serial forking setflag(FLT_USRPREF); # set counter if (!$avp(prio)) { $avp(prio) = 1; } # overwrite request URI with highest priority contact if ($avp(prio1) =~ "^sip:00") { $ru = $avp(prio1) + "@host"; xlog("L_INFO","PRIO 1 is tel number, RURI set: $ru"); } else { $ru = $avp(prio1); xlog("L_INFO","PRIO 1 is SIP URI, RURI set: $ru"); } } } 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_reply("REPLY_ONE"); t_on_failure("FAIL_ONE"); #if users have priorities set, use FAIL_FORK failure route if ( isflagset(FLT_USRPREF) ) { t_on_failure("FAIL_FORK"); } } if (isflagset(FLT_ACCMISSED)) xlog("L_INFO","RELAY, $rm $ru, ACCMISSED FLAG IS SET"); else xlog("L_INFO","RELAY, $rm $ru, ACCMISSED FLAG IS NOT SET"); if (!t_relay()) { sl_reply_error(); } 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()) { xlog("L_INFO","WITHINDLG, loose_route()"); if (is_method("BYE")) { xlog("L_INFO","WITHINDLG, BYE, DO ACCOUNTING"); 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; } } # USER location service route[LOCATION] { #skip if $ru is telephone number if ($ru =~ "^sip:00") { xlog("L_INFO","SKIP lookup..."); } else { if (!lookup("location")) { switch ($rc) { case -1: case -3: t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } } # when routing via usrloc, log the missed calls also, but only if user doesn't have prios set if ( is_method("INVITE") && !(isflagset(FLT_USRPREF))) { setflag(FLT_ACCMISSED); } } # Failure route for forked calls failure_route[FAIL_FORK] { #!ifdef WITH_NAT if (is_method("INVITE") && (isbflagset(FLB_NATB) || isflagset(FLT_NATS))) { unforce_rtp_proxy(); } #!endif if ($avp(prio) >= 1) { $avp(prio) = $avp(prio) + 1; # handle 2nd branch if ( ($avp(prio) == 2) && ( isflagset(FLT_USRPREF) )) { t_on_failure("FAIL_FORK"); if ($avp(prio2) =~ "^sip:00") { xlog("L_INFO","FAIL FORK, PRIO 2 is tel number"); $ru = $avp(prio2) + "@host"; } else { xlog("L_INFO","FAIL FORK, PRIO 2 is SIP URI"); $ru = $avp(prio2); route(LOCATION); } setflag(FLT_ACCMISSED); } else { $avp(prio) = 0; $ru = $(avp(uuid)); rewritehostport("host:port"); xlog("L_INFO","FAIL FORK, VOICEMAIL email:$avp(email), ru:$ru, br: $br"); append_hf("P-App-Name: voicemail\r\n"); append_hf("P-App-Param: Email-Address=$avp(email)\r\n"); } route(RELAY); } if (t_is_canceled()) { exit; } } [View Less]

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Kamailio with IMS extensions build
by loic.trottin＠orange.com 27 Sep '11

27 Sep '11

Hi all, I try to compile Kamailio with IMS extensions from OpenIMSCore available in carstenbock/ims branch in the GIT repository. I have some problems with the modules "Client_Ro" and "pcscf" that do not compile: CC (gcc) [M Client_Ro.so] ims_ro.o ims_ro.c: In function 'send_ccr_interim': ims_ro.c:594: error: 'AVP_EPC_User_Equipment_Info_Type_MAC' undeclared (first use in this function) ims_ro.c:594: error: (Each undeclared identifier is reported only once ims_ro.c:594: error: … [View More]for each function it appears in.) ims_ro.c: In function 'send_ccr_stop': ims_ro.c:749: error: 'AVP_EPC_User_Equipment_Info_Type_MAC' undeclared (first use in this function) ims_ro.c: In function 'Ro_Send_CCR': ims_ro.c:831: error: 'struct dlg_binds' has no member named 'get_dlg' ims_ro.c:835: error: 'struct dlg_binds' has no member named 'unref_dlg' ims_ro.c:886: error: 'AVP_EPC_User_Equipment_Info_Type_MAC' undeclared (first use in this function) CC (gcc) [M pcscf.so] pcc_avp.o pcc_avp.c:1459: error: 'AVP_IMS_Subscription_Id_Type_E164' undeclared (first use in this function) pcc_avp.c:1459: error: (Each undeclared identifier is reported only once pcc_avp.c:1459: error: for each function it appears in.) pcc_avp.c:1461: error: 'AVP_IMS_Subscription_Id_Type_SIP_URI' undeclared (first use in this function) Does anybody have encountered these problems? Is there a stable version tagged of the branch carstenbock/ims available in the GIT repository ? Best regards, Loïc. ******************************************************************************** IMPORTANT.Les informations contenues dans ce message electronique y compris les fichiers attaches sont strictement confidentielles et peuvent etre protegees par la loi. Ce message electronique est destine exclusivement au(x) destinataire(s) mentionne(s) ci-dessus. Si vous avez recu ce message par erreur ou s il ne vous est pas destine, veuillez immediatement le signaler a l expediteur et effacer ce message et tous les fichiers eventuellement attaches. Toute lecture, exploitation ou transmission des informations contenues dans ce message est interdite. Tout message electronique est susceptible d alteration. A ce titre, le Groupe France Telecom decline toute responsabilite notamment s il a ete altere, deforme ou falsifie. De meme, il appartient au destinataire de s assurer de l absence de tout virus. IMPORTANT.This e-mail message and any attachments are strictly confidential and may be protected by law. This message is intended only for the named recipient(s) above. If you have received this message in error, or are not the named recipient(s), please immediately notify the sender and delete this e-mail message. Any unauthorized view, usage or disclosure ofthis message is prohibited. Since e-mail messages may not be reliable, France Telecom Group shall not be liable for any message if modified, changed or falsified. Additionally the recipient should ensure they are actually virus free. ******************************************************************************** [View Less]

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need little help to setting up pike module
by dirtytoni＠gmx.de 27 Sep '11

27 Sep '11

Hi to all on this list =) jUSt a tiny user question aboUt pike module(at least kamailio doc). are these values for sampling_time_unit seconds or minutes ? it's not in the doc. same 4 remove_latency. thanks for helping me out sorry to bother you best regards Toni -- Empfehlen Sie GMX DSL Ihren Freunden und Bekannten und wir belohnen Sie mit bis zu 50,- Euro! https://freundschaftswerbung.gmx.de

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Account 403 to RADIUS
by Mino Haluz 26 Sep '11

26 Sep '11

Hi, I have this code: if ( is_user_in("From", "blocked") && is_method("INVITE")) { xlog("L_INFO", "XLOG: [number_and_ruri_checks] NOTICE: Account ($fu) to ($ru) is blocked"); sl_send_reply("403", " Account blocked "); exit; } I would like to account also this answer to Radius, now I have it only in application xlog. Thanks, Mino

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Migration from v3.1.x to 3.2.0
by Daniel-Constantin Mierla 26 Sep '11

26 Sep '11

Hello, a kind reminder for everyone to add to the wiki page hints for migration from v3.1.x to v3.2.0. Especially to developers that know what kind of major changes they did to existing modules in 3.1.x. The wiki is open for anyone: * http://sip-router.org/wiki/install/3.1.x-to-3.2.x I just updated the section related to database structure changes. If you spot errors/missing parts, go ahead and fix it. Thanks, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio … [View More]

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Re: [SR-Users] I am getting this error in the log When REGISTER message is received from jitsi
by Daniel-Constantin Mierla 23 Sep '11

23 Sep '11

On 9/22/11 9:04 PM, Krishna Kurapati wrote: > Thanks Daniel, > > Here it is... > > [...] It looked ok for my eye. Can you run kamailio with higher debug level (debug=4 in config file) and send the log messages related to parsing the URI for AoR? Cheers, Daniel > > > On Thu, Sep 22, 2011 at 1:44 PM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: > > Hello, > > can you grab such REGISTER with … [View More]

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Re: [SR-Users] CANCEL is not processed in onsend_route
by Daniel-Constantin Mierla 23 Sep '11

23 Sep '11

Hello, On 9/15/11 4:02 PM, Timo Klecker wrote: > Hi Daniel, > > I managed to get a log with debug 4 from one call. I had to take out all > IP-Addresses and Usernames, hope you can use it. the log confirmed what I expected: Sep 15 15:32:33 vux896 /sbin/kamailio[31594]: DEBUG: siptrace [siptrace.c:954]: no uas msg, local transaction Sep 15 15:32:33 vux896 /sbin/kamailio[31594]: DEBUG: tm [t_fwd.c:1242]: DEBUG: e2e_cancel: e2e cancel proceeding Because a new CANCEL is generated … [View More]

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carrierroute module
by Mino Haluz 23 Sep '11

23 Sep '11

Hi, I have a problem with configuring the carrierroute module, I do not clearly understand how it should work from the documentation. What I would like to do is that I have 2 gateways (A.A.A.A and B.B.B.B), lots of scan_prefixes with different rewrite prefixes. The traffic should be sent to the 1 gateway, and if it fails (5xx,4xx codes) it should go to the another. I have only one SIP domain (domain.com). The question is, how the tables (domain_name, carrierroute, carrierfailroute) should be … [View More]

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XCAP over SIP
by Krishna Kurapati 23 Sep '11

23 Sep '11

Hi, The document http://kb.asipto.com/kamailio:presence:k31-made-simple describes that XCAP can be sent over SIP instead of over HTTP. I tried to do however, the server is unable to update presence rules. I was using PUBLISH method. I was using content-type = text... Has anyone used SIP to send over XCAP? Appreciate if someone could share a link or configuration changes required. Thanks Krish

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I am getting this error in the log When REGISTER message is received from jitsi
by Krishna Kurapati 22 Sep '11

22 Sep '11

Sep 22 09:49:53 siptest /usr/sbin/kamailio[3160]: ERROR: registrar [common.c:75]: failed to parse Address of Record Sep 22 09:49:53 siptest /usr/sbin/kamailio[3160]: ERROR: registrar [save.c:822]: failed to extract Address Of Record Is there a problem when AoR contains "_" for example is AoR krish_kura(a)sip.org is not allowed? Thanks Krish Kura

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perl module compilation on centos
by Daniel-Constantin Mierla 22 Sep '11

22 Sep '11

Hello, while building the v3.1.5 RPMs, the perl module failed to compile on centos 5 and 6, with following error: error: `PL_unitcheckav' undeclared (first use in this function ) so its packaging was disabled. It works fine on opensuse and fedora. Someone on twitter tried with the package perl-ExtUtils-Embed (like in fedora) and worked. However, OpenSUSE Build System fails to find that package. Is anyone here with latest centos 5 or 6 that can try compilation of perl module in v3.1.5 … [View More]

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Re: [SR-Users] Kamailio and FreeSWITCH realtime integration, tutorial?
by Henrik Aagaard Sørensen 22 Sep '11

22 Sep '11

But FreeSwitch don't use MySQL like Asterisk? So adapting isn't that easy or...??? On 21/09/2011, at 16.18, David <david(a)styleflare.com> wrote: You can adapt pretty easily. On 9/21/11 3:46 PM, Henrik Aagaard Sørensen wrote: Does anyone know if there somewhere exists a tutorial about Kamailio and FreeSWITCH realtime integration? I have Googled a lot and found: http://kb.asipto.com/kamailio:index and http://kb.asipto.com/freeswitch:index On the same site there is a tutorial for … [View More]

2 2

Re: [SR-Users] Problem with REGISTER retransmissions
by Daniel-Constantin Mierla 22 Sep '11

22 Sep '11

Hello, I tried to reproduce with a very simple config to handle register requests. So I sent a REGISTER with expires 60 (very much the same as the one in the trace you sent), waited a while, then sent two REGISTERs having the CSeq incremented. All went fine. I tested with master and 3.1 branches (so the last is actually 3.1.5). Can you upgrade to 3.1.5 first and try again? If still fails, then update config to print the return code for save() function, something like: $var(r) = save("… [View More]

2 3

Kamailio 3.1.5 RPMS
by Daniel-Constantin Mierla 22 Sep '11

22 Sep '11

Hello, just to let you know that the RPMs of Kamailio 3.1.5 were generated for CentOS, Fedora, OpenSUSE and RedHat. This time CentOS RPMs are packaged for both versions 5 and 6. Links are available on our wiki at: * http://www.kamailio.org/dokuwiki/doku.php/packages:rpms Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

Kamailio and FreeSWITCH realtime integration, tutorial?
by Henrik Aagaard Sørensen 22 Sep '11

22 Sep '11

Does anyone know if there somewhere exists a tutorial about Kamailio and FreeSWITCH realtime integration? I have Googled a lot and found: http://kb.asipto.com/kamailio:index and http://kb.asipto.com/freeswitch:index On the same site there is a tutorial for Kamailio and Asterisk realtime integration, but not for FreeSWITCH.

4 4

Re: [SR-Users] Dialog module showing calls that have already been terminated
by Timo Reimann 21 Sep '11

21 Sep '11

Hey Phillip, On 20.09.2011 13:48, Phillman25 Kyriacou wrote: > Thanks for your email. > > Yes dlg_manage(); has to now be called on INVITE and BYE/CANCEL messages. > Where would i have to call loose_route()? Only on INVITE? On *all* in-dialog requests, i.e., all requests which contain a To tag. This may include Re-INVITEs too (but not initial INVITEs). > My configuration did not change between 3.1.2 and 3.1.5. > > Call flow example: > ============== > > … [View More]

2 8

Kamailio Development Training, San Francisco Area, October 2011
by Daniel-Constantin Mierla 21 Sep '11

21 Sep '11

Hello, since I am attending GSoC'11 Mentors Summit and come to Bay Area again, it is time to bring the developer training in USA, therefore I am planning to organize a two-day seminar focused on developing Kamailio extensions in C. Among topics to be approached: - internal architecture - SIP parser - memory manager - locking manager - database API - config file language interpreter - RPC interface - pseudo-variables and transformations framework - module interface – write your own … [View More]

1 0

how to remove media description from sdp?
by Juha Heinanen 21 Sep '11

21 Sep '11

trying to hack around the possible fix_nated_sdp("1") issue that i outlined in previous message, i thought to remove video media description from sdp. in sdpops module there is sdp_with_media(type), but for some reason no sdp_remove_media(type) function. is there some other means to get rid of sdp video media description? -- juha

3 3

Digest authentication
by Morten Isaksen 20 Sep '11

20 Sep '11

Hi, Sorry for the slightly off topic question, but I am trying to debug a buggy PBX and want to check if the digest information is correct. I have this perl script. #!/usr/bin/perl use strict; use Digest::MD5 qw(md5 md5_hex md5_base64); my $uri = 'sip:sip1.uni-tel.dk'; my $method = "SIP/2.0"; my $username = "69102208"; my $password = "password"; my $nonce = "4e78d2b0a72196d9123be7c88d093bd74c77f35e"; my $realm = "sip1.uni-tel.dk"; my $a1 = "$username:$realm:$password"; my $ha1 = md5_hex($… [View More]

2 2

wt_timer, delayed replies and sendto error
by Andreas Granig 20 Sep '11

20 Sep '11

Hi, Another funny thing I encountered lately is this: Client sends INVITE via kamailio (via dispatcher) to sems, gets back 100/200, sends another couple of INVITE with same CSeq (looks like retransmissions), gets back more 200, then sends ACKs for each of the 200 it got. Looks like a quite broken client to me, but anyways. The interesting thing is that after >5000ms (which is bigger then tm.wt_timer), another 200 is sent from sems to kamailio to the client. I see the 200 coming to … [View More]

2 6

Re: [SR-Users] Dialog module showing calls that have already been terminated
by Timo Reimann 20 Sep '11

20 Sep '11

Hey Phillip, you probably forgot to keep the mailing list in CC, I'll do that for you. On 19.09.2011 08:43, Phillman25 Kyriacou wrote: > You are right. > > I added the function dlg_manage(); when there is a BYE or CANCEL and now > its working perfectly. Before i had this function defined only on > INVITES. The strange thing is that it was working normally before when > dlg_manage(); was defined only for INVITE messages in the route block. So you are saying that before, … [View More]you called dlg_manage() on INVITEs only but now, you need that function on both INVITE and BYE/CANCEL messages? That'd still qualify as a bug as the trigger to tracking the dialog (i.e., dlg_manage()) is required only once. After that, the dialog module makes sure that all dialog-related messages are properly processed. The only thing that's necessary for sequential requests to be tracked correctly is a call to loose_route() on such messages. If you are sure that your Kamailio configuration didn't change between 3.1.2 and 3.1.5 w.r.t. dialog tracking or anything that could affect it, I'd be willing to take a closer look at this issue (giving you can provide me with more information on your scenario, like call flow examples or similar). Cheers, --Timo > The calls that were suppose to be terminated before where in state 3 to > answer your question. However, i still see calls in state 3 now but they > are valid. > > Regards > Phillip > > On Fri, Sep 16, 2011 at 8:01 PM, Timo Reimann <timo.reimann(a)1und1.de > <mailto:timo.reimann@1und1.de>> wrote: > > Hey Phillip, > > after looking closer at what has been reported to me initially, your > case may be different after all. > > Could you possibly use "kamctl fifo dlg_list" to check what state the > majority of dialogs that should be terminated in your opinion are in? > Being given the reference counter should be useful in debugging this as > well. > > > Thanks, > > --Timo > > > > On 16.09.2011 14:22, Phillman25 Kyriacou wrote: > > On Fri, Sep 16, 2011 at 3:09 PM, Timo Reimann > <timo.reimann(a)1und1.de <mailto:timo.reimann@1und1.de> > > <mailto:timo.reimann@1und1.de <mailto:timo.reimann@1und1.de>>> wrote: > > > > Hey Phillip, > > > > > > On 16.09.2011 13:35, Phillman25 Kyriacou wrote: > > > Hello > > > > > > I have been facing an issue where the dialog module is showing > > calls as > > > being active when in fact the call has already been > completed long ago > > > and this is giving wrong number of concurrent calls on our > SNMP work > > > station (CACTI) when polling the data from Kamailio. I > realized this > > > only started occurring after i upgraded from 3.1.2 to 3.1.5, has > > anyone > > > experienced the same issue? > > > > I was *just* being notified of issues concerning dialogs not being > > deleted. Working on this right now to report back soon. > > > > Thanks for the note! > > > > > > Cheers, > > > > --Timo [View Less]

2 1

Add client certificate checking against a CRL in opensips
by tomsc 20 Sep '11

20 Sep '11

Hi everyone, I'd like to check that a client certificat is revoked or not against a crl. Actually, opensips use context SSL_CTX. How can I do with this context? I do this change to load the crl : load_crl(SSL_CTX * ctx, char *filename) { LM_DBG("entered load crl\n"); X509_STORE *pStore = SSL_CTX_get_cert_store(ctx); X509_LOOKUP* plookup = X509_STORE_add_lookup(pStore, X509_LOOKUP_file()); if (!X509_load_crl_file(plookup, filename, X509_FILETYPE_PEM)){ LM_ERR("unable to load certificate … [View More]

2 1

Lazy evaluation in if expressions?
by Morten Isaksen 20 Sep '11

20 Sep '11

Does Kamailio use lazy evaluation in if expressions? Like if ($var(a) == 2 && $var(b) ==3) { ... } then $var(b) == 3 is only evaluated if $var(a) == 2 is true -- Morten Isaksen

3 2

directed media problem when video is included in sdp
by Juha Heinanen 20 Sep '11

20 Sep '11

when sems receives voice call invite from ua behind nat that has direction:active in its sdp, like this: Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): foo.bar 0 0 IN IP4 192.168.1. 68 Session Name (s): - Connection Information (c): IN IP4 192.168.1.68 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 8500 RTP/AVP 18 4 112 … [View More]100 101 0 8 106 Media Attribute (a): sendrecv Media Attribute (a): rtpmap:18 G729/8000 Media Attribute (a): rtpmap:4 G723/8000 Media Attribute (a): rtpmap:112 AMR/8000 Media Attribute (a): rtpmap:100 EG711U/8000 Media Attribute (a): rtpmap:101 EG711A/8000 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:106 telephone-event/8000 Media Attribute (a): direction:active sems includes direction:passive in 200 OK and starts to correctly send rtp packets to the ip address/port of received rtp packets. however, then sems receives a voice+video call invite like this: Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): foo.bar 0 0 IN IP4 192.168.1.68 Session Name (s): - Connection Information (c): IN IP4 192.168.1.68 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 8500 RTP/AVP 18 4 112 100 101 0 8 106 Media Attribute (a): sendrecv Media Attribute (a): rtpmap:18 G729/8000 Media Attribute (a): rtpmap:4 G723/8000 Media Attribute (a): rtpmap:112 AMR/8000 Media Attribute (a): rtpmap:100 EG711U/8000 Media Attribute (a): rtpmap:101 EG711A/8000 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:106 telephone-event/8000 Media Description, name and address (m): video 8600 RTP/AVP 34 Bandwidth Information (b): AS:300 Media Attribute (a): sendrecv Media Attribute (a): rtpmap:34 H263/90000 Media Attribute (a): direction:active it does not include direction:passive in 200 OK and keeps on sending rtp packets to 192.168.1.68 even after receiving first voice rtp packet from public ip of the ua. is this a bug in sems or is it a bug in sr nathelper_k fix_nated_sdp("1") implementation, i.e., does that function place direction:active line to only one place, when it should place it in two places (in both voice and video media descriptions)? -- juha [View Less]

3 4

Module db_text
by Bruno Bresciani 20 Sep '11

20 Sep '11

Hi All, I'm starting kamailio version 3.1.2 with db_text module but when it tries to load route table, the following error is generated: * "ERROR: <core> [daemonize.c:273]: Main process exited before writing to pipe" * Someone Can Help me to understand why this error is happening? Below is part of kamaio's trace. Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: db_text [dbt_base.c:208]: _tbc->name: route Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG:… [View More] db_text [dbt_base.c:209]: _tbc->nrcols: 9 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: db_text [dbt_base.c:210]: _nc: 8 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: db_text [dbt_base.c:238]: new res with 8 cols Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: db_text [dbt_res.c:69]: new res with 8 cols Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: <core> [db_res.c:118]: allocate 28 bytes for result set at 0x82e23f0 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: <core> [db_res.c:155]: allocate 32 bytes for result names at 0x82e3798 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: <core> [db_res.c:165]: allocate 32 bytes for result types at 0x82e37c0 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: db_text [dbt_api.c:80]: allocate 8 bytes for RES_NAMES[0] at 0x82e2438 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: db_text [dbt_api.c:80]: allocate 8 bytes for RES_NAMES[1] at 0x82e2390 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: db_text [dbt_api.c:80]: allocate 8 bytes for RES_NAMES[2] at 0x82e23d0 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: db_text [dbt_api.c:80]: allocate 8 bytes for RES_NAMES[3] at 0x82e23e0 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: db_text [dbt_api.c:80]: allocate 8 bytes for RES_NAMES[4] at 0x82e37e8 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: db_text [dbt_api.c:80]: allocate 8 bytes for RES_NAMES[5] at 0x82e37f8 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: db_text [dbt_api.c:80]: allocate 8 bytes for RES_NAMES[6] at 0x82e3808 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: db_text [dbt_api.c:80]: allocate 8 bytes for RES_NAMES[7] at 0x82e3818 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: <core> [db_res.c:184]: allocate 8 bytes for rows at 0x82e3828 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: <core> [db_row.c:119]: allocate 160 bytes for row values at 0x82e3838 Sep 19 16:13:01 sswpst00 kamailio: ERROR: <core> [daemonize.c:273]: Main process exited before writing to pipe Best Regards [View Less]

2 1

Re: [SR-Users] rtpproxy VS Turn
by Mikael Williams 19 Sep '11

19 Sep '11

> > ICE is not supported as I select the Media Proxy (which is TURN) manually > in some Nat cases...Even note that I do not have peer to peer calls between > 2 endpoints, just have VOIP to PSTN calls...All my VOIP customers supports > STUN/TURN but without ICE > > One more thing: I would like to ask how rtp proxy help to bypass NAT else > than working as relay server...Does it have any other job like find the NAT > type or check blocked port or something? > > … [View More]

2 1

Re: [SR-Users] sipp to kamailio doesn't match ACK [SOLVED]
by Uri Shacked 19 Sep '11

19 Sep '11

Hi, I got the same problem. Do you have the XML SIPp file for example? 10x Uri

1 0

Dialog module showing calls that have already been terminated
by Phillman25 Kyriacou 16 Sep '11

16 Sep '11

Hello I have been facing an issue where the dialog module is showing calls as being active when in fact the call has already been completed long ago and this is giving wrong number of concurrent calls on our SNMP work station (CACTI) when polling the data from Kamailio. I realized this only started occurring after i upgraded from 3.1.2 to 3.1.5, has anyone experienced the same issue? my config: ======== loadmodule "dialog.so" . . . . # ----------------- DIALOG MODULE PARAMETERS--------------… [View More]

2 3

kamailio debian/ubuntu support
by Jon Bonilla 16 Sep '11

16 Sep '11

Hi all with the release of kamailio 3.1.5, I've done a small change to the debian repositories: Nightly builds of kamailio_3.0 branch have been dropped and the repository has been erased. There is no development in this branch and it has no sense to keep nightly builds for it. Those (few) people using kamailio_3.0-nightly repositories are advised to move their sources.list to the stable 3.0 repository of upgrade to 3.1 or master branches. kamailio 3.0 will continue to be packaged if any new … [View More]

4 4

rtpproxy VS Turn
by Mikael Williams 16 Sep '11

16 Sep '11

Dear All, I would like to ask please if and in case I'm using STUN/TURN for Nat traversal to drp the rtpproxy from my solution as NAT traversal and relay server as I'm using SYUN for NAT traversal and TURN as relay in case of Symmetric NAT Regards

2 1

Communication from IPv6 to Ipv4
by Loïc Fofana 16 Sep '11

16 Sep '11

My architecture is described below: TDM device----------- Call server[IPv4]---------------[IPv4]Kamailio[IPv6]------------[IPv6]Asterisk---------[IPv6]Linphone * * * * * * * * * * * * Media Server Rtpproxy Media Server and Rtpptoxy carry the media. Call Server carry only … [View More]

2 1

Re: [SR-Users] Kamailio terminating after upgrading from 3.1.2 to 3.1.4
by Phillman25 Kyriacou 16 Sep '11

16 Sep '11

Dear Marius Thank you for your prompt reply. I apologise, As i am a newbie could you please explain how to produce the below output that you require? Thanking you in advance for your help! Regards Phillip > Hello > Hello. You should have a coredump . Can you please send the trace (bt full) as seen with gdb ? Thanks Marius > Im facing a big problem after upgrading from version 3.1.2 to 3.1.4. > The kamailio process keeps on terminating every now and then, this is > what i … [View More]

5 9

(no subject)
by Loïc Fofana 16 Sep '11

16 Sep '11

This is my scenario: Call server -------------(IPV4)Kamailio(IPV6)----------Asterisk(IPV6)------Linphone(IPV6) * * * Media server

2 1

One way communications in Kamailio 3.1.4
by CaptWho 16 Sep '11

16 Sep '11

I'm running 3.1.4 on centos and I'm having some trouble with voice only going one way. Both extensions will ring each other, but after they connect, voice will only travel in one direction. One extension hears fine, but can't talk. I've totally opened up the firewalls (including port 5060) and I'm still having the trouble. I'm not behind NAT. I've tried it using X-Lite, VoIP phones and ATAs in a number of combinations. The problem is across the board. X-Lite sometimes automatically … [View More]

3 3

Open source HLR (Home Location Register).
by Henrik Aagaard Sørensen 16 Sep '11

16 Sep '11

I know this is the mailing list for Kamailio, but hopefully someone can help me out as it is "kinda" related to VoIP etc. Does anyone know any open source / free HLR implementation?

4 3

Siremis site access from Romania
by Dani Popa 16 Sep '11

16 Sep '11

Hi, There is any reasons why any linked Siremis site is not accessible from Romania ? i tried to access http://siremis.asipto.com/ and other sites and they are not accesible. Thanks, Dani Popa

4 6

Re: [SR-Users] planning v3.1.5
by Vitaliy Aleksandrov 16 Sep '11

16 Sep '11

There is a bug in UAC module. uac_req_send doesn't restore avps. I tried to backport your patch to 3.1 but with no success. Could you add this patch to a new stable release ?

2 9

DNS support
by CT Radu 15 Sep '11

15 Sep '11

Dear sirs and madams, I was wondering what is the current support for DNS fallback in the 3.1.x branch . Lets assume we have a new dialog started and the name in the To field needs to be resolved via a NAPTR. Kamailio sends a NAPTR request, gets 3 SRVs , each with a number of A or AAAA records. The best/most suitable SRV is selected, however no response is returned from any of the addresses in the A records. Now, we should go the second SRV and try the A records in there and so on. My … [View More]

2 1

OT: anyone in Bay Area
by Daniel-Constantin Mierla 14 Sep '11

14 Sep '11

Hello, anyone here living/working in San Francisco/Bay Area that has a bit of time to assist me with some stuff? Reply to me for follow up. Thanks, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

Kamailio v3.1.5 Released
by Daniel-Constantin Mierla 14 Sep '11

14 Sep '11

Hello, Kamailio SIP Server v3.1.5 stable release is out. This is a maintenance release of latest stable branch, 3.1, that includes fixes since release of v3.1.4. There is no change to database schema or configuration language structure. Deployments running previous v3.1.x versions are strongly recommended to be upgraded to v3.1.5. For more details about version 3.1.5, visit: http://www.kamailio.org/w/2011/09/kamailio-v3-1-5-released/ Cheers, Daniel -- Daniel-Constantin Mierla -- http:… [View More]

1 0

preparing release of v3.1.5
by Daniel-Constantin Mierla 14 Sep '11

14 Sep '11

Hello, today starting with 14:00 I will be preparing the release of v3.1.5. If any devel has commits to do in branch 3.1, please do it before to avoid conflicts. Thanks, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 1

Invitation to connect on LinkedIn
by Dani Popa 14 Sep '11

14 Sep '11

SIP, I'd like to add you to my professional network on LinkedIn. - Dani Dani Popa Technical Manager at NECC Telecom Romania Confirm that you know Dani Popa: https://www.linkedin.com/e/-szw6fj-gskbz4q0-1q/isd/4204604361/O7PsXc0-/?hs=… -- You are receiving Invitation to Connect emails. Click to unsubscribe: http://www.linkedin.com/e/-szw6fj-gskbz4q0-1q/c6YlT3v6LB4U4BZW-5Wc20tVAQF44… (c) 2011 LinkedIn Corporation. 2029 Stierlin Ct, Mountain View, CA 94043, USA.

1 0

Kamailio terminating after upgrading from 3.1.2 to 3.1.4
by Phillman25 Kyriacou 14 Sep '11

14 Sep '11

Hello Im facing a big problem after upgrading from version 3.1.2 to 3.1.4. The kamailio process keeps on terminating every now and then, this is what i was able to retrieve from my logs. Sep 14 12:52:31 SipProxy1 kernel: [71448.062089] __ratelimit: 15 callbacks suppressed Sep 14 12:52:31 SipProxy1 kernel: [71448.062096] kamailio[13963]: segfault at 14 ip 0816452a sp bfedc940 error 4 in kamailio[8048000+1d9000] Sep 14 12:52:31 SipProxy1 /usr/local/sbin/kamailio[13991]: : <core> [pass_fd.… [View More]

2 1

parse errors in rr-callback of path module
by Andreas Granig 14 Sep '11

14 Sep '11

Hi, path module calls register_rrcb when "use_received" is set to 1 to extract the "received=" param and set its content as dst-uri if present. Now I've noticed that when e.g. a message has a pre-loaded route like "Route: <sip:mydomain.com;lr>", then kamailio (at least 3.1.3) spits the following error message: /usr/sbin/kamailio[23953]: ERROR: <core> [parser/parse_param.c:481]: parse_params(): Invalid character, ; expected /usr/sbin/kamailio[23953]: ERROR: path [path.c:213]: … [View More]

3 8

planning v3.1.5
by Daniel-Constantin Mierla 14 Sep '11

14 Sep '11

Hello, it's being quite some time since we released 3.1.4, so I am thinking of packaging soon v3.1.5 out of stable branch 3.1. My plan is to release it Wednesday, September 14, 2011. Any other opinions? Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

6 6

Does xcap server of kamailio support partial update of xcap document?
by laura testi 13 Sep '11

13 Sep '11

Hi, I am using the XCAP server of kamailio for contact list and presence rules. It seems it only support to update the whole xcap doc of the user but not the partial update. Can anyone tell me if the kamailio's xcap server support the partial update of a user's doc? for example, to remove a contact from the contact list or add a contact to the contact list without PUT or DELETE the whole xcap document of this user? If yes, what's the syntax of the HTTP request? Thank you very much in advanced Best Regards, Laura

3 12

max_contacts
by Rafael Rincon 13 Sep '11

13 Sep '11

Hello Guys, Is there any chance to modify at running time (in a register event) the max_contacts parameter? I am running OpenSER version 1.4 . Regards, Rafael Rincon rrincon(a)ipcomnetwork.com +13054772902 x 111

4 7

cr default route
by caio 13 Sep '11

13 Sep '11

2 3

wildcarded sip contacts
by Antanas Masevicius 13 Sep '11

13 Sep '11

Hello everyone, been looking for a way to do sip wildcarded contact matching where contact could look like <sip:2222..@33.33.33.33:5060>. Current usrloc module does not implement any such matching, but it seems there is a new rfc 5947 issued regarding this problem. Maybe someone is already working on it? What options do i have in sip-router / kamailio besides LCR based ones ? I want to do local contact matching as fast as possible and without a need to reload lcr table. Antanas Masevicius NTT

1 0

Re: [SR-Users] Fwd: configuration
by Daniel-Constantin Mierla 13 Sep '11

13 Sep '11

Hello, On 9/12/11 4:14 PM, neili ma wrote: > Hello, > > Thanks a lot for your advice, I found and solved the problem thanks to > the debugger, actually the problem was with > > modparam("auth_db", > "db_url","mysql://openser:openserrw@localhost/openser") > > I just had to change localhost to my locationserver IP and the problem > was solved :), thanks again. good that you find it. Don't forget to cc the mailing list always so people reading the thread (… [View More]archive later) see the conclusion -- it can help many others running on same situation. Cheers, Daniel > > Sincerely, > Neili > > > 2011/9/12 Daniel-Constantin Mierla <miconda(a)gmail.com > <mailto:miconda@gmail.com>> > > Hello, > > you have another group of actions that can send 401 Unauthorized: > > > if (!check_to()) { > sl_send_reply("401", "Unauthorized"); > exit; > } > > You can either run with debug=4 and watch the syslog to see some > hints in messages where it fires this 401, or use debugger module > with cfgtrace option set and you see the execution path of the > config in the logs. > > If you send here the ngrep of such REGISTER and the reply we can > spot what could be the situation for 401. > > Cheers, > Daniel > > > On 9/11/11 9:18 PM, neili ma wrote: >> Hello sr-users, >> >> I'm new using kamailio open source, and I would like to ask for help. >> >> My version is 3.1.4 and I am trying to configure 3 servers, one >> with proxy, one with registrar and one with location server. >> >> My problem is when I try to send a register message it returns an >> error "401 Unauthorized", I know the error is in the fuction >> www_authorize("10.254.239.7", "subscriber") but I don't know how >> to solve it, for me it seems correct the parameters. >> >> Below I have attatched the proxy and registrar relevant route >> code for my problem, and the IPs for proxy an registrar are: >> 10.254.239.7-----------10.254.239.8 >> >> If someone can help me, I will be gratefull. Thanks in advance. >> >> ####REGISTRAR CODE >> >> route{ >> >> # initial sanity checks -- too long messages >> if (msg:len >= 2048 ) { >> sl_send_reply("513", "Message too big"); >> exit; >> }; >> >> if (!uri==myself) { >> sl_send_reply("400", "Bad destination, only >> register messages de$ >> exit; >> }; >> >> # the only valid message for the registrar is the register >> message >> # other messages are answered with an error message >> if (uri==myself) { >> >> if (method=="REGISTER") { >> >> sl_send_reply("100", "Trying"); >> >> if (!www_authorize("10.254.239.7", "subscriber")) { >> www_challenge("10.254.239.7", "1"); >> exit; >> }; >> >> if (!check_to()) { >> sl_send_reply("401", "Unauthorized"); >> exit; >> }; >> >> consume_credentials(); >> if(!save("location")) { >> sl_reply_error(); >> }; >> exit; >> } else { >> sl_send_reply("403", "Forbidden"); >> exit; >> }; >> }; >> } >> >> #########PROXY CODE >> >> if (method=="ACK") { >> route(1); >> exit; >> } if (method=="INVITE") { >> route(3); >> exit; >> } else if (method=="REGISTER") { >> route(2); >> exit; >> }; >> >> lookup("aliases"); >> if (!uri==myself) { >> route(1); >> }; >> >> route[2] { >> #rewrinting the host allows the registrar to know that the >> message is de$ >> >> rewritehost("10.254.239.8"); >> if(!t_relay()) { >> sl_reply_error(); >> }; >> exit; >> } >> >> >> >> >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org> >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > -- > Daniel-Constantin Mierla --http://www.asipto.com > Kamailio Advanced Training, Oct 10-13, Berlin:http://asipto.com/u/kat > http://linkedin.com/in/miconda -- http://twitter.com/miconda > > -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda [View Less]

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CANCEL is not processed in onsend_route
by Timo Klecker 13 Sep '11

13 Sep '11

Hello all, I want to log the incoming and outgoing CANCEL on my Kamailio. I tried with a call of sip_trace in the default route setflag(22); if (method == "CANCEL") { sip_trace(); if (t_check_trans()) { t_relay(); exit; } } This works great, I get the incoming CANCEL in the Database. But the outgoing CANCEL is not logged with flag set. So I tried to use onsend_route … [View More]

3 4

question bout UA and BOX in same NAT
by MingHon 12 Sep '11

12 Sep '11

Hello List, Can i have UA in the same nat with the box and also UA in different nat at the same time? My box is currrently working behind nat with ip 192.168.2.3 and advertised_address is set to sip.mydomain.com is it possible to append the via header replace the sip.mydomain.com to 192.168.2.3 ? Thanks, -- Regards, MingHon

2 4

no publish from pua/pua_mi to outbound proxy
by Juha Heinanen 12 Sep '11

12 Sep '11

i started to test sr presence and for some reason sr does not send out any publish request when i issue one via pua_mi. i have in route [xmlrpc_requests]: xlog("L_INFO", "Handling XMLRPC $rm from <$si> with body <$(rb{re.subst,/[\n\r\t]//g})>\n"); set_reply_no_connect(); set_reply_close(); dispatch_rpc(); syslog shows that sr receives xmlrpc request: Aug 29 19:42:13 sip /usr/sbin/pres-serv[2118]: INFO: Handling XMLRPC POST from <127.0.0.1> with body <… [View More]

2 13

Nat problems
by David Zambrano 12 Sep '11

12 Sep '11

Hi Ive been trying to configure kamailio as a loadbalancer between two transcoders that afterwards send all calls to a softswitch but I currently have two problems. My first and most important one is a NAT problem or so I think. All calls in UDP work perfectly but when dealing with calls in TCP only outgoing calls work, no incoming calls work. After doing some traces on the network I realized that the transcoder is trying to reach the router in front of the softphone and skipping the Kamailio … [View More]

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Fwd: configuration
by neili ma 12 Sep '11

12 Sep '11

Hello sr-users, I'm new using kamailio open source, and I would like to ask for help. My version is 3.1.4 and I am trying to configure 3 servers, one with proxy, one with registrar and one with location server. My problem is when I try to send a register message it returns an error "401 Unauthorized", I know the error is in the fuction www_authorize("10.254.239.7", "subscriber") but I don't know how to solve it, for me it seems correct the parameters. Below I have attatched the proxy and … [View More]registrar relevant route code for my problem, and the IPs for proxy an registrar are: 10.254.239.7-----------10.254.239.8 If someone can help me, I will be gratefull. Thanks in advance. ####REGISTRAR CODE route{ # initial sanity checks -- too long messages if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; if (!uri==myself) { sl_send_reply("400", "Bad destination, only register messages de$ exit; }; # the only valid message for the registrar is the register message # other messages are answered with an error message if (uri==myself) { if (method=="REGISTER") { sl_send_reply("100", "Trying"); if (!www_authorize("10.254.239.7", "subscriber")) { www_challenge("10.254.239.7", "1"); exit; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); exit; }; consume_credentials(); if(!save("location")) { sl_reply_error(); }; exit; } else { sl_send_reply("403", "Forbidden"); exit; }; }; } #########PROXY CODE if (method=="ACK") { route(1); exit; } if (method=="INVITE") { route(3); exit; } else if (method=="REGISTER") { route(2); exit; }; lookup("aliases"); if (!uri==myself) { route(1); }; route[2] { #rewrinting the host allows the registrar to know that the message is de$ rewritehost("10.254.239.8"); if(!t_relay()) { sl_reply_error(); }; exit; } [View Less]

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Re: [SR-Users] Kamailio 3.0.4 is crashed
by Daniel-Constantin Mierla 12 Sep '11

12 Sep '11

Hello, try to keep the mailing list cc-ed -- others may be able to help you faster or this discussion will be helpful for different people. The username_spec and password_spec were removed indeed, the first was useless since the username was taken from auth header and the second became parameter to pv_www_authenticate, see: http://kamailio.org/docs/modules/3.1.x/modules/auth.html#pv_www_authenticate You simply have to use something like: if (!pv_www_authenticate("$td", "$var(password)",… [View More] "0")) { www_challenge("$td", "0"); exit; } Cheers, Daniel On 9/10/11 12:51 PM, Khoa Phuong Viet Ngo wrote: > Hi Daniel, > > Thanks Daniel for your answer. > > It's very helpful for me :). > > Actually i tried with using kamailio 3.1.4 (lastest kamailio) but i > got the problem with auth module and parameter with auth module. > Because some parameters at auth module have changed. > http://kamailio.org/docs/modules/stable/modules/auth.html [3.1.4] > > But in 3.0.4 kamailio, i am using auth module with these parameter and > it had error with new kamailio (3.1.4) . :( > > kamailio.cfg > > ######################################################## > loadmodule auth.so > > # ----- auth params ----- > modparam("auth", "username_spec", "$var(username)") > modparam("auth", "password_spec", "$var(password)") > modparam("auth", "calculate_ha1", 1) > > route[AUTH] { > $var(utype) = 0; > #!ifdef WITH_AUTH > if (is_method("REGISTER")) > { > if(is_present_hf("Authorization")) > { > # authenticate the REGISTER requests (uncomment to enable > auth) > if(!route(GETPASSWD)) > { > www_challenge("", "0"); > exit; > } > if(!pv_www_authorize("")) > { > www_challenge(""/*realm*/,"0"/*qop*/); > exit; > } > } else { > www_challenge("", "0"); > exit; > } > } else { > $var(srcgrp) = allow_source_address_group(); > if ($var(srcgrp) > 0) { > # src ip allowed > $var(utype) = 1; > if($var(srcgrp)!=100) > { > $rd = $Ri; > $fs = "udp:" + $Ri + ":5062"; > } > return 1; > } > # authenticate if from local subscriber (uncomment to enable auth) > if (from_uri==myself) > { > if(is_present_hf("Proxy-Authorization")) > { > # authenticate the non-REGISTER requests (uncomment to > enable auth) > if(!route(GETPASSWD)) > { > proxy_challenge("", "0"); > exit; > } > if(!pv_proxy_authorize("")) > { > proxy_challenge(""/*realm*/,"0"/*qop*/); > exit; > } > } else { > proxy_challenge("", "0"); > exit; > } > $var(utype) = 2; > consume_credentials(); > # caller authenticated > } > } > #!endif > return; > } > > ##################################################### > > Error logs: > > Sep 6 19:37:01 CorePBX kamailio: ERROR: <core> [modparam.c:150]: > set_mod_param_regex: parameter <username_spec> not found in module <auth> > Sep 6 19:37:01 CorePBX kamailio: : <core> [cfg.y:3419]: parse error > in config file /usr/local/etc/kamailio/kamailio.cfg, line 260, column > 51: Can't set module parameter > Sep 6 19:37:01 CorePBX kamailio: ERROR: <core> [modparam.c:150]: > set_mod_param_regex: parameter <password_spec> not found in module <auth> > Sep 6 19:37:01 CorePBX kamailio: : <core> [cfg.y:3419]: parse error > in config file /usr/local/etc/kamailio/kamailio.cfg, line 261, column > 51: Can't set module parameter > Sep 6 19:37:01 CorePBX kamailio: ERROR: <core> [modparam.c:150]: > set_mod_param_regex: parameter <calculate_ha1> not found in module <auth> > Sep 6 19:37:01 CorePBX kamailio: : <core> [cfg.y:3419]: parse error > in config file /usr/local/etc/kamailio/kamailio.cfg, line 262, column > 36: Can't set module parameter > Sep 6 19:37:01 CorePBX kamailio: : <core> [cfg.y:3419]: parse error > in config file /usr/local/etc/kamailio/kamailio.cfg, line 562, column > 27: unknown command, missing loadmodule? > Sep 6 19:37:01 CorePBX kamailio: : <core> [cfg.y:3419]: parse error > in config file /usr/local/etc/kamailio/kamailio.cfg, line 594, column > 30: unknown command, missing loadmodule? > Sep 6 19:37:39 CorePBX kamailio: ERROR: <core> [modparam.c:150]: > set_mod_param_regex: parameter <username_spec> not found in module <auth> > Sep 6 19:37:39 CorePBX kamailio: : <core> [cfg.y:3419]: parse error > in config file /usr/local/etc/kamailio/kamailio.cfg, line 260, column > 51: Can't set module parameter > Sep 6 19:37:39 CorePBX kamailio: ERROR: <core> [modparam.c:150]: > set_mod_param_regex: parameter <password_spec> not found in module <auth> > Sep 6 19:37:39 CorePBX kamailio: : <core> [cfg.y:3419]: parse error > in config file /usr/local/etc/kamailio/kamailio.cfg, line 261, column > 51: Can't set module parameter > Sep 6 19:37:39 CorePBX kamailio: ERROR: <core> [modparam.c:150]: > set_mod_param_regex: parameter <calculate_ha1> not found in module <auth> > > ######################################################## > > I appreciate all your help on this case. > > Thanks Daniel, > KhoaNVP > > > > > On Sat, Sep 10, 2011 at 1:58 PM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: > > Hello, > > it seems like a buffer overflow happened previously, since the > crash is in malloc. Do you get any error messages is syslog? Can > you recompile with QM_DBG_MALLOC like explained in the next link > and try again watching the syslog messages? > http://www.kamailio.org/dokuwiki/doku.php/troubleshooting:memory > > On another hand, 3.0.4 is quite old, why are not using 3.1.4, it > is far more updated than 3.0 series -- we are just to release > first in 3.2 series. Upgrading from 3.0 to 3.1 is rather trivial. > > Cheers, > Daniel > > > On 9/9/11 8:38 PM, KhoaNVP wrote: > > Hi everyone, > > I hope you can help me with this issues. > I'm working with kamailio 3.0.4 and when i implement the call > via kamailio > after 1-->3 mins , kamailio is crashed. > > It's my debug from core log. > > Core was generated by `kamailio -E -ddd'. > Program terminated with signal 11, Segmentation fault. > #0 fm_malloc (qm=0x88c860, size=<value optimized out>) at > mem/f_malloc.c:382 > 382 *f=frag->u.nxt_free; > (gdb) qt > Undefined command: "qt". Try "help". > (gdb) bt > #0 fm_malloc (qm=0x88c860, size=<value optimized out>) at > mem/f_malloc.c:382 > #1 0x000000000041fd27 in dup_lump_list_r (l=0x9275f8, > dir=LD_AFTER, > error=0x7fff23bd661c) at data_lump.c:453 > #2 0x000000000041ffa9 in dup_lump_list_r (l=0x927640, > dir=LD_AFTER, > error=0x7fff23bd666c) at data_lump.c:476 > #3 0x000000000041ff6d in dup_lump_list_r (l=0x926db0, > dir=<value optimized > out>, error=0x7fff23bd66b4) at data_lump.c:465 > #4 0x0000000000420058 in dup_lump_list (l=0x88c860) at > data_lump.c:508 > #5 0x00002b8b2efb858d in prepare_new_uac (t=0x2b8b3218ffa8, > i_req=0x926ef0, > branch=0, uri=0x927150, path=0x9274d0, next_hop=<value > optimized out>, > fsocket=0x0, snd_flags=0 '\000', fproto=0, flags=0) at > t_fwd.c:242 > #6 0x00002b8b2efb94c7 in add_uac (t=0x2b8b3218ffa8, > request=0x926ef0, > uri=0x927150, next_hop=0x7, path=0x9274d0, proxy=0x0, fsocket=0x0, > snd_flags=0 '\000', proto=0, flags=0) at t_fwd.c:679 > #7 0x00002b8b2efbce7a in t_forward_nonack (t=0x2b8b3218ffa8, > p_msg=0x926ef0, proxy=0x0, proto=0) at t_fwd.c:1340 > #8 0x00002b8b2efb4996 in t_relay_to (p_msg=0x926ef0, > proxy=0x0, proto=0, > replicate=0) at t_funcs.c:358 > #9 0x0000000000413489 in do_action (h=0x7fff23bd71e0, a=0x8be788, > msg=0x926ef0) at action.c:860 > #10 0x0000000000415e03 in run_actions (h=0x7fff23bd71e0, > a=0x8be788, > msg=0x926ef0) at action.c:1315 > #11 0x00000000004161e7 in run_actions_safe (h=0x7fff23bd8bb0, > a=0x38, > msg=0x88ca08) at action.c:1370 > #12 0x000000000049696b in rval_get_int (h=0x7fff23bd8bb0, > msg=0x88ca08, > i=0x7fff23bd75dc, rv=0x7, cache=0x7) at rvalue.c:897 > #13 0x000000000049b7fc in rval_expr_eval_int (h=0x7fff23bd8bb0, > msg=0x926ef0, res=0x7fff23bd75dc, rve=0x8be878) at rvalue.c:1841 > #14 0x000000000049b8ad in rval_expr_eval_int (h=0x7fff23bd8bb0, > msg=0x926ef0, res=0x7fff23bd79d8, rve=0x8bef48) at rvalue.c:1848 > #15 0x0000000000411d49 in do_action (h=0x7fff23bd8bb0, a=0x8bf798, > msg=0x926ef0) at action.c:834 > #16 0x0000000000415e03 in run_actions (h=0x7fff23bd8bb0, > a=0x8be608, > msg=0x926ef0) at action.c:1315 > #17 0x0000000000413084 in do_action (h=0x7fff23bd8bb0, a=0x8f6060, > msg=0x926ef0) at action.c:479 > #18 0x0000000000415e03 in run_actions (h=0x7fff23bd8bb0, > a=0x8f6060, > msg=0x926ef0) at action.c:1315 > #19 0x0000000000411dab in do_action (h=0x7fff23bd8bb0, a=0x8f62d0, > msg=0x926ef0) at action.c:853 > #20 0x0000000000415e03 in run_actions (h=0x7fff23bd8bb0, > a=0x8f62d0, > msg=0x926ef0) at action.c:1315 > #21 0x0000000000413084 in do_action (h=0x7fff23bd8bb0, a=0x8bd698, > msg=0x926ef0) at action.c:479 > #22 0x0000000000415e03 in run_actions (h=0x7fff23bd8bb0, > a=0x8bd398, > msg=0x926ef0) at action.c:1315 > #23 0x0000000000411dab in do_action (h=0x7fff23bd8bb0, a=0x8bd788, > msg=0x926ef0) at action.c:853 > #24 0x0000000000415e03 in run_actions (h=0x7fff23bd8bb0, > a=0x8a1340, > msg=0x926ef0) at action.c:1315 > #25 0x0000000000416174 in run_top_route (a=0x8a1340, > msg=0x926ef0, c=<value > optimized out>) at action.c:1391 > #26 0x000000000048009c in receive_msg (buf=0x57e779 "DEBUG", > len=<value > optimized out>, rcv_info=0x7fff23bd8e20) at receive.c:196 > #27 0x000000000050698b in udp_rcv_loop () at udp_server.c:520 > #28 0x0000000000455e2f in main_loop () at main.c:1447 > #29 0x0000000000456f32 in main (argc=<value optimized out>, > argv=0x7fff23bd90e8) at main.c:2251 > (gdb) > > > Thanks with all the help, > Khoa > > > -- > Daniel-Constantin Mierla -- http://www.asipto.com > Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat > http://linkedin.com/in/miconda -- http://twitter.com/miconda > > -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda [View Less]

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Kamailio 3.0.4 is crashed
by KhoaNVP 10 Sep '11

10 Sep '11

Hi everyone, I hope you can help me with this issues. I'm working with kamailio 3.0.4 and when i implement the call via kamailio after 1-->3 mins , kamailio is crashed. It's my debug from core log. Core was generated by `kamailio -E -ddd'. Program terminated with signal 11, Segmentation fault. #0 fm_malloc (qm=0x88c860, size=<value optimized out>) at mem/f_malloc.c:382 382 *f=frag->u.nxt_free; (gdb) qt Undefined command: "qt". Try "help". (gdb) bt #0 fm_malloc (qm=… [View More]0x88c860, size=<value optimized out>) at mem/f_malloc.c:382 #1 0x000000000041fd27 in dup_lump_list_r (l=0x9275f8, dir=LD_AFTER, error=0x7fff23bd661c) at data_lump.c:453 #2 0x000000000041ffa9 in dup_lump_list_r (l=0x927640, dir=LD_AFTER, error=0x7fff23bd666c) at data_lump.c:476 #3 0x000000000041ff6d in dup_lump_list_r (l=0x926db0, dir=<value optimized out>, error=0x7fff23bd66b4) at data_lump.c:465 #4 0x0000000000420058 in dup_lump_list (l=0x88c860) at data_lump.c:508 #5 0x00002b8b2efb858d in prepare_new_uac (t=0x2b8b3218ffa8, i_req=0x926ef0, branch=0, uri=0x927150, path=0x9274d0, next_hop=<value optimized out>, fsocket=0x0, snd_flags=0 '\000', fproto=0, flags=0) at t_fwd.c:242 #6 0x00002b8b2efb94c7 in add_uac (t=0x2b8b3218ffa8, request=0x926ef0, uri=0x927150, next_hop=0x7, path=0x9274d0, proxy=0x0, fsocket=0x0, snd_flags=0 '\000', proto=0, flags=0) at t_fwd.c:679 #7 0x00002b8b2efbce7a in t_forward_nonack (t=0x2b8b3218ffa8, p_msg=0x926ef0, proxy=0x0, proto=0) at t_fwd.c:1340 #8 0x00002b8b2efb4996 in t_relay_to (p_msg=0x926ef0, proxy=0x0, proto=0, replicate=0) at t_funcs.c:358 #9 0x0000000000413489 in do_action (h=0x7fff23bd71e0, a=0x8be788, msg=0x926ef0) at action.c:860 #10 0x0000000000415e03 in run_actions (h=0x7fff23bd71e0, a=0x8be788, msg=0x926ef0) at action.c:1315 #11 0x00000000004161e7 in run_actions_safe (h=0x7fff23bd8bb0, a=0x38, msg=0x88ca08) at action.c:1370 #12 0x000000000049696b in rval_get_int (h=0x7fff23bd8bb0, msg=0x88ca08, i=0x7fff23bd75dc, rv=0x7, cache=0x7) at rvalue.c:897 #13 0x000000000049b7fc in rval_expr_eval_int (h=0x7fff23bd8bb0, msg=0x926ef0, res=0x7fff23bd75dc, rve=0x8be878) at rvalue.c:1841 #14 0x000000000049b8ad in rval_expr_eval_int (h=0x7fff23bd8bb0, msg=0x926ef0, res=0x7fff23bd79d8, rve=0x8bef48) at rvalue.c:1848 #15 0x0000000000411d49 in do_action (h=0x7fff23bd8bb0, a=0x8bf798, msg=0x926ef0) at action.c:834 #16 0x0000000000415e03 in run_actions (h=0x7fff23bd8bb0, a=0x8be608, msg=0x926ef0) at action.c:1315 #17 0x0000000000413084 in do_action (h=0x7fff23bd8bb0, a=0x8f6060, msg=0x926ef0) at action.c:479 #18 0x0000000000415e03 in run_actions (h=0x7fff23bd8bb0, a=0x8f6060, msg=0x926ef0) at action.c:1315 #19 0x0000000000411dab in do_action (h=0x7fff23bd8bb0, a=0x8f62d0, msg=0x926ef0) at action.c:853 #20 0x0000000000415e03 in run_actions (h=0x7fff23bd8bb0, a=0x8f62d0, msg=0x926ef0) at action.c:1315 #21 0x0000000000413084 in do_action (h=0x7fff23bd8bb0, a=0x8bd698, msg=0x926ef0) at action.c:479 #22 0x0000000000415e03 in run_actions (h=0x7fff23bd8bb0, a=0x8bd398, msg=0x926ef0) at action.c:1315 #23 0x0000000000411dab in do_action (h=0x7fff23bd8bb0, a=0x8bd788, msg=0x926ef0) at action.c:853 #24 0x0000000000415e03 in run_actions (h=0x7fff23bd8bb0, a=0x8a1340, msg=0x926ef0) at action.c:1315 #25 0x0000000000416174 in run_top_route (a=0x8a1340, msg=0x926ef0, c=<value optimized out>) at action.c:1391 #26 0x000000000048009c in receive_msg (buf=0x57e779 "DEBUG", len=<value optimized out>, rcv_info=0x7fff23bd8e20) at receive.c:196 #27 0x000000000050698b in udp_rcv_loop () at udp_server.c:520 #28 0x0000000000455e2f in main_loop () at main.c:1447 #29 0x0000000000456f32 in main (argc=<value optimized out>, argv=0x7fff23bd90e8) at main.c:2251 (gdb) Thanks with all the help, Khoa -- View this message in context: http://old.nabble.com/Kamailio-3.0.4-is-crashed-tp32433892p32433892.html Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com. [View Less]

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Error reading AVP
by Carlos Ruiz Diaz 09 Sep '11

09 Sep '11

Hello. I'm currently developing an OpenSER module which does some business logic for the company I work for. For compatibility reasons we are working with openser-1.2.2-tls and we haven't been able to upgrade to newer version yet. In one block of the code I have to read an avp which is written inside the openser.cfg script using: OPENSER: avp_copy("sip:123@123.123.123:5060", "$avp(s:selectedroute)/g"); C CODE: if (is_response && is_invite_seq) { struct usr_avp* destavp = NULL; … [View More]

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10 Years SER Conference - Presentations
by Daniel-Constantin Mierla 08 Sep '11

08 Sep '11

Hello, the presentations from 10 Years SER Conference I received so far were uploaded at: * http://www.kamailio.org/events/2011-10yearsSER/ Expect the rest during the next days. The recording of the talk about Gemeinschaft 4 by Stefan Wintermeyer is available on youtube: * http://www.youtube.com/watch?v=mbGn5N_y4rw Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/… [View More]

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Re: [SR-Users] Kamailio + rtpproxy talking to multiple carrier gateways - Solution
by Sarat C. Vemuri 07 Sep '11

07 Sep '11

I think I have this working. For anyone else trying to do similar thing (or placing Kamailio behind FW/NAT), here is my cookbook. We are trying to setup the following: [UAS] <------> (PubIP)[FireWall/One-to-OneNAT]<-------->(PrivIP)[Kamailio/RTPProxy](PrivIP)<---------->(PrivIP2)[UAC] UAC/UAS cannot reach each other directly. For simplicity I am running RTPProxy on same machines as Kamailio. But this can be changed. Versions/Patches/fixes you need if you are running 3.1.… [View More]2 : (Trunk/3.2 has these): a. Fixed forward.c for mhome to work properly (included in 3.1.4). b. rr module from trunk to be able to use 2 strings for record_route_preset c. Patched rtpproxy module to actually use the forced ip supplied to force_rtp_proxy. Patch at the end of this message d. rtpproxy version 1.2.1 What to do: 1. Setup FW/NAT rules to map public IP to PrivateIP and allow communications to/from UAS 2. Setup RTPProxy in "bridged" mode (e.g -F -l 10.10.5.90/10.10.5.90 -s udp:10.10.5.90:9005) listening on PrivateIP. 3. Setup the config file to: a. Setup modparams to make the above rtpproxy available. Use set_rtp_proxy_set if you have more than one instance. b. Do the following where you are processing Outbound *initial* invite (i.e. check totag) (Call from UAC to UAS) set_advertised_address("YourPublicIP "); if(!has_totag()) { record_route_preset("YourPublicIP:5060;r2=on;nat=yes","YourPrivateIp:5060;r2=on;nat=yes"); } force_rtp_proxy("ocfaei"," YourPublicIP "); c. Do the following where you are processing Inbound invite (Call from UAS to UAC) if(!has_totag()) { record_route_preset("YourPrivateIP:5060;r2=on;nat=yes", "YourPublicIp:5060;r2=on;nat=yes"); } force_rtp_proxy("ocfaei"); d. Do the following where you are processing replies (180/183/2XX) For Inbound Calls force_rtp_proxy("ocfa","YourPublicIP"); For Outbound Calls force_rtp_proxy("ocfa"); e. Unforce RTPProxy on BYE or Failures. Do not use record_route in the above scenario. If you have it now, remove it or make sure it doesn't get called for this specific case. Explanation: In #b, you are fixing the Via headers to use public IP, setting Record-Route headers so that ACK/RE-INVITE can find way back and are force fixing SDP with public IP for outbound calls. In #c, You are doing same thing except no need to fix Via and the route set is reversed. No need to force fix SDP either since it is figured out automatically. In #d, You are force fixing the SDP to use the public IP for inbound or usual for outbound so that the audio stream from UAS can be relayed. The r2=on parameter is required (as well as enabling double rr, which is default) so that the proxy can deal with the route sets properly Hope the above helps others. SV. Rtpproxy Path 2226,2233c2226 < LM_DBG("Str2 is %s\n", str2); < if(str2) { < newip.s = str2; < LM_DBG("NewIp is %s\n", newip.s); < } < else { < newip.s = (argc < 2) ? str2 : argv[1]; < } --- > newip.s = (argc < 2) ? str2 : argv[1]; [View Less]

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Re: [SR-Users] Kamailio + rtpproxy talking to multiple carrier gateways
by Sarat C. Vemuri 07 Sep '11

07 Sep '11

Ovidiu, Thanks for your time. The "fixes" I pulled in is the latest rr module only. I didn't see anything in that diff to make this work. Could you elaborate a little? However, I was able to get "Outbound" working properly by adding ";r2=on" to the record_route_preset IPs. In going through the source code, I noticed that this is what regular record_route uses to enable double rr and it would automatically take both headers out if it sees that. That seems to have worked. Any pitfals … [View More]with this? I am still working on "Inbound". For some reason my carrier GW keeps resending invites even after receiving ACK. I need to see if it is an issue with the carrier. Thanks SV. -----Original Message----- From: Ovidiu Sas [mailto:osas@voipembedded.com] Sent: Tuesday, September 06, 2011 9:50 AM To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List Subject: Re: [SR-Users] Kamailio + rtpproxy talking to multiple carrier gateways It seems that something is miss configured on your server. The fixes that I made in the trunk (and you pulled in your local 3.1 repo) were designed to handle the scenario that you are trying to implement. The ACK should be handled properly and routed to the upstream carrier (following the same path as the initial INVITE). Regards, Ovidiu Sas On Mon, Sep 5, 2011 at 5:07 PM, Sarat C. Vemuri <sarat.vemuri(a)fthco.com> wrote: > Again, I apologize for this clumsy way of replying. > > Ovidiu, > > Thanks for the pointer on set_advertised_address. I had to patch rtpproxy module (and rr module for the two parameters to request_route_preset) since I am running 3.1. > > However, I still have a problem with ACKs after following what you suggested. > > INVITE from Internal to Carrier routes properly (two Request-Route headers, one internal IP and other public IP). On 200 OK, the carrier GW properly copies the route set in to Route header. Now the route contains two entries, the public IP and the private IP of Kamailio. The Internal UAC then sends the ACK back to Kamailio. Everything is fine till this point. Now, Kamailio removes the top entry, which is the private IP and then promptly sends the ACK to the public IP of itself!. Of course, that doesn't go anywhere. > > How do I remove the "public IP" entry from the route set before forwarding the reply to Internal UAC? Is there another way to deal with this? I've tried to set an alias= core parameter with the public IP, but doesn't seem to have any effect. The public IP is not reachable from internal network. > > Thanks for your help > > SV. > ------------------------------ > > Message: 3 > Date: Sat, 3 Sep 2011 16:44:22 -0700 > From: Ovidiu Sas <osas(a)voipembedded.com> > Subject: Re: [SR-Users] Kamailio + rtpproxy talking to multiple > carrier gateways - some via Firewall/NAT > To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> > Message-ID: > > <CAND0Lkt_dpcTm2WKMywMhX6rdsX1ia0r=LYrzB1WfX3oN32Wrg(a)mail.gmail.com> > Content-Type: text/plain; charset=windows-1252 > > It is feasable to what you want: kamailio behind NAT proxying traffic > from/to public internet to/from private network. > You will need to properly craft the INVITE and use proper record route headers. > Use set_advertised_address when needed: > http://kamailio.org/dokuwiki/doku.php/core-cookbook:3.1.x#set_advertis > ed_address Also, use record_route_preset (note that there are two > parameters): > http://kamailio.org/docs/modules/devel/modules_k/rr.html#id2547566 > > That should do it. You don't need any patches for rtpproxy. > Just use force_rtpp_proxy (and force the IP address): > http://kamailio.org/docs/modules/devel/modules/rtpproxy.html#id2546034 > > > Note: Make sure that you understand how in-dialog requests are routed > by a proxy in order to know how to properly handle the initial INVITE. > > > Regards, > Ovidiu Sas > > On Sat, Sep 3, 2011 at 2:53 PM, Sarat C. Vemuri <sarat.vemuri(a)fthco.com> wrote: >> We are trying to configure Kamailio ?(3.1.x) as a ?boarder proxy? >> where it acts as the front for various carrier gateways so that >> internal UACs and UASs are unaware of the carrier gateways. >> >> >> >> Let me try to present a clear picture of our setup. >> >> 1.?????? Kamailio has several NICs (physical or vlan).? Each on a >> different subnet. One subnet is internal/has routes for internal.? >> Other subnets are private connections to carriers or a ?route to public Internet. >> >> 2.?????? All of these subnets are non-routable from Internet. In >> addition , the carrier private connections are not routable internally. >> >> 3.?????? Connection to public internet is via a FW/NAT (one-to-one >> NAT), which maps to one of the interfaces. >> >> 4.?????? All internal? UAC/UAS connect on the internal subnet. >> >> 5.?????? We are using RTPProxy? (at least one instance per carrier) >> to relay media between internal and carrier subnets >> >> >> >> We are able to make this setup up work great except for one >> scenario.? One of the carriers is only reachable via public >> Internet.? Due to security requirements, our Kamailio cannot have a >> public IP address and must use FW/NAT. I guess this scenario is >> ??Proxy behind NAT? and not really encouraged. But I would like to >> see if there is a way to make this work.? We cannot use the >> ?advertised_address? since it changes the IP for every ?route?. >> >> >> >> We were able to get this mostly ?working by doing the following >> >> 1.?????? mhomed=1 >> >> 2.?????? Small patch in the rtpproxy module so that ?force_rtp_proxy >> actually uses the IP address passed (public IP). >> >> 3.?????? Using request_route_preset(?publicIP?) >> >> >> >> The above ?mostly? works.? By that I mean, the INVITE transaction is >> properly passed between internal UAS and carrier SBC and the call is setup. >> However, further transactions (BYE/re-INVITE) etc do not work >> properly. So, far-end hangups are not ?working etc. >> >> >> >> I?ve searched various archives of this and other SER lists looking to >> see if anyone was able to get this scenario working, but couldn?t >> find a definitive answer.? Most of them point to using ?advertised_address?. >> >> >> >> So, and ideas on how to make ?Proxy behind NAT? work without using >> advertised_address?? Am I wasting my time? >> >> >> >> Thanks in advance for any help you can offer. >> >> >> >> SV. >> >> >> >> _______________________________________________ >> 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 > [View Less]

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permanent usrloc entries in Kamailio
by Klaus Darilion 07 Sep '11

07 Sep '11

Hi! Recently I had a problem with permanent usrloc entries when upgrading a server from SER 0.9 to Kamailio 3.1. Obviously the problem was that the permanent entries had "flag=128" and now expires=0 is used, see old thread: http://old.nabble.com/usrloc-permanent-records-and-strptime-td17668417.html The strange thing is, that even setting the expiry to 2099-0...... did not worked and Kamailio deleted the "expired" entries. Only setting it to "1970-01-01 01:00:00" worked. Any ideas what … [View More]

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Siremis User Role permissions
by Camila Troncoso 07 Sep '11

07 Sep '11

Hi, I have a question about the assignment of Role permissions to limit some Users privileges in Siremis. I defined a user , and then a Role denying some actions, but I couldn’t deny the action from de SER menu . The Role permissions only show the modules on Application Menu from Administration Tab . What if I want to deny access to some items inside SER Menu , like modifying the ACL services or /and Rounting services? Regards, *Camila Troncoso **|* Ingeniero de Desarrollo VoissNet … [View More]

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Check disconnected users
by Anto 06 Sep '11

06 Sep '11

Hello I want to store in a file access and disconnections to the system. To access (registrar), I thought scheduling a perl script and executed with the perl module, the problem I find with disconnections. Since the user can be disconnected for various reasons, had planned to do a script that runs an SQL table location and according to the expires field, see if disconnection has occurred or not. Would have to run this script in cron every X minutes. Is there a simpler way to check this or have … [View More]

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Re: [SR-Users] Kamailio + rtpproxy talking to multiple carrier gateways
by Sarat C. Vemuri 06 Sep '11

06 Sep '11

Again, I apologize for this clumsy way of replying. Ovidiu, Thanks for the pointer on set_advertised_address. I had to patch rtpproxy module (and rr module for the two parameters to request_route_preset) since I am running 3.1. However, I still have a problem with ACKs after following what you suggested. INVITE from Internal to Carrier routes properly (two Request-Route headers, one internal IP and other public IP). On 200 OK, the carrier GW properly copies the route set in to Route … [View More]header. Now the route contains two entries, the public IP and the private IP of Kamailio. The Internal UAC then sends the ACK back to Kamailio. Everything is fine till this point. Now, Kamailio removes the top entry, which is the private IP and then promptly sends the ACK to the public IP of itself!. Of course, that doesn't go anywhere. How do I remove the "public IP" entry from the route set before forwarding the reply to Internal UAC? Is there another way to deal with this? I've tried to set an alias= core parameter with the public IP, but doesn't seem to have any effect. The public IP is not reachable from internal network. Thanks for your help SV. ------------------------------ Message: 3 Date: Sat, 3 Sep 2011 16:44:22 -0700 From: Ovidiu Sas <osas(a)voipembedded.com> Subject: Re: [SR-Users] Kamailio + rtpproxy talking to multiple carrier gateways - some via Firewall/NAT To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Message-ID: <CAND0Lkt_dpcTm2WKMywMhX6rdsX1ia0r=LYrzB1WfX3oN32Wrg(a)mail.gmail.com> Content-Type: text/plain; charset=windows-1252 It is feasable to what you want: kamailio behind NAT proxying traffic from/to public internet to/from private network. You will need to properly craft the INVITE and use proper record route headers. Use set_advertised_address when needed: http://kamailio.org/dokuwiki/doku.php/core-cookbook:3.1.x#set_advertised_ad… Also, use record_route_preset (note that there are two parameters): http://kamailio.org/docs/modules/devel/modules_k/rr.html#id2547566 That should do it. You don't need any patches for rtpproxy. Just use force_rtpp_proxy (and force the IP address): http://kamailio.org/docs/modules/devel/modules/rtpproxy.html#id2546034 Note: Make sure that you understand how in-dialog requests are routed by a proxy in order to know how to properly handle the initial INVITE. Regards, Ovidiu Sas On Sat, Sep 3, 2011 at 2:53 PM, Sarat C. Vemuri <sarat.vemuri(a)fthco.com> wrote: > We are trying to configure Kamailio ?(3.1.x) as a ?boarder proxy? where it > acts as the front for various carrier gateways so that internal UACs and > UASs are unaware of the carrier gateways. > > > > Let me try to present a clear picture of our setup. > > 1.?????? Kamailio has several NICs (physical or vlan).? Each on a different > subnet. One subnet is internal/has routes for internal.? Other subnets are > private connections to carriers or a ?route to public Internet. > > 2.?????? All of these subnets are non-routable from Internet. In addition , > the carrier private connections are not routable internally. > > 3.?????? Connection to public internet is via a FW/NAT (one-to-one NAT), > which maps to one of the interfaces. > > 4.?????? All internal? UAC/UAS connect on the internal subnet. > > 5.?????? We are using RTPProxy? (at least one instance per carrier) to relay > media between internal and carrier subnets > > > > We are able to make this setup up work great except for one scenario.? One > of the carriers is only reachable via public Internet.? Due to security > requirements, our Kamailio cannot have a public IP address and must use > FW/NAT. I guess this scenario is ??Proxy behind NAT? and not really > encouraged. But I would like to see if there is a way to make this work.? We > cannot use the ?advertised_address? since it changes the IP for every > ?route?. > > > > We were able to get this mostly ?working by doing the following > > 1.?????? mhomed=1 > > 2.?????? Small patch in the rtpproxy module so that ?force_rtp_proxy > actually uses the IP address passed (public IP). > > 3.?????? Using request_route_preset(?publicIP?) > > > > The above ?mostly? works.? By that I mean, the INVITE transaction is > properly passed between internal UAS and carrier SBC and the call is setup. > However, further transactions (BYE/re-INVITE) etc do not work properly. So, > far-end hangups are not ?working etc. > > > > I?ve searched various archives of this and other SER lists looking to see if > anyone was able to get this scenario working, but couldn?t find a definitive > answer.? Most of them point to using ?advertised_address?. > > > > So, and ideas on how to make ?Proxy behind NAT? work without using > advertised_address?? Am I wasting my time? > > > > Thanks in advance for any help you can offer. > > > > SV. > > > > _______________________________________________ > 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 > > [View Less]

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KAMAILIO RTP PROXY
by Phillman25 Kyriacou 06 Sep '11

06 Sep '11

Hi I want to use Kamailio to handle both signalling and media always. My scenario is the following PSTN==>KAMAILIO==>ASTERISK. I have a dialplan that points to ASTERISK when a call comes into Kamailio. Signalling is fine, however i cant seem to get the Media to work, i have installed the RTPPROXY as suggested by RTPPROXY module and have implemented it into the routing logic, however i have one way audio. The PSTN side cannot hear anything. I dont use the Kamailio for sip registrations … [View More]

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db_mysql "Commands out of sync; you can't run this command now "
by MÉSZÁROS Mihály 06 Sep '11

06 Sep '11

Hi, I am using the git version of kamailio, and i am experiencing problem This message is repeated in log with all kind of mysql query (insert/delete/select) Aug 27 22:04:05 hal /usr/sbin/kamailio[25681]: ERROR: db_mysql [km_dbase.c:124]: driver error on query: Commands out of sync; you can't run this command now Aug 27 22:04:05 hal /usr/sbin/kamailio[25681]: ERROR: <core> [db_query.c:103]: error while submitting query Aug 27 22:04:05 hal /usr/sbin/kamailio[25681]: ERROR: usrloc [… [View More]

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Re: [SR-Users] Kamailio + rtpproxy talking to multiple carrier gateways
by Sarat C. Vemuri 05 Sep '11

05 Sep '11

Alex, Sorry about this clumsy way of replying, but I am stuck with Outlook 2010 and Exchange. You stated the problem correctly. Carrier GWs (UAS) and internal clients (UAC) cannot reach each other directly (TCP/UDP). They have to go through Kamailio and rtpproxy. I am goint to reply to Ovidiu as well with some more details. Thanks SV. ------------------------------ Message: 2 Date: Sat, 03 Sep 2011 19:12:38 -0400 From: Alex Balashov <abalashov(a)evaristesys.com> Subject: Re: [SR-… [View More]

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SIP Analysis & Testing tools
by Vladimir Broz 05 Sep '11

05 Sep '11

Hi, for sake of Vaclav's courage, let me introduce the "SIP Analysis & Testing tools". This is a set of very powerful utilities that may help you to do SIP call flows analysis. It can do SIP dialogs and transactions matching and analysis in large pcaps, create pretty call flow diagrams, check and verify SIP message content (headers, parameters,..) and many more.... It's available on: https://github.com/vkubart/sipat For more details, please ask Vaclav Kubart (vaclav.kubart(a)iptel.… [View More]

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question
by Allen D. Ford 05 Sep '11

05 Sep '11

Where are the instructions for this gui Im trying to setup a sip load balancer and rtp proxy for nating "Success is not measured by the positions you reach in life but by the obstacles you overcome -B.T.W" [Description: Description: Description: Description: Description: cyford]<http://www.cyfordtechnologies.com/>Director Of Technology Allen D. Ford 4290-106 Bells Ferry Road, Suite 20 , Kennesaw , GA 30144 Tel: 404-424-8270: Fax: 404-424-8271 Email: allen(a)cyfordtechnologies.com&… [View More]

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10 Years SER - Awards
by Daniel-Constantin Mierla 05 Sep '11

05 Sep '11

Hello, a surprise part of 10 Years SER Conference was granting *special* awards to the people attending the event that had a major contribution in the management and development of SER/Kamailio to date. Most of them are well known in the community, but not all of them. Hope you like them, it proved to be a very entertaining moment. Here is the pdf with the awards: * http://asipto.com/u/48 The plan is to extend it with the other people in our community as ideas with award titles pop … [View More]

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10 Years SER - Follow UP
by Daniel-Constantin Mierla 05 Sep '11

05 Sep '11

Hello, first I want to thank to all participants at 10 Years SER event in Berlin last Friday. We spent great time, at least I did! We accepted 50 visiting participants to match the maximum capacity of the conference room, being full booked about 2 weeks before. Just few people didn't show up and didn't inform us at all in advance, which was a pity over all since several persons that registered later were not accepted, thus some of them could have attended. Folks at Fokus working with … [View More]

2 1

tls + tcp problems
by MÉSZÁROS Mihály 05 Sep '11

05 Sep '11

Hi, I am experiencing two problem: First: I am experiencing that tls read buffer overflow, if i use default value of tcp_rd_buf_size If i doubled the size of read buffer and i use tcp_rd_buf_size=8192 then the problem seems to be solved. Second: I am experiencing that TLS connection is dropped by sip proxy very frequently! And i think the tcp connection shouldn't dropped! So my guess is that TLS communication is not restarting the timeout counters! So in every 2 minutes the sip-… [View More]

2 1

Carrier module for kamailio
by Mino Haluz 05 Sep '11

05 Sep '11

Hi, I need a module which could allow me to send traffic to various carriers and it has to support some important features. So some basic ones: - possibility to re-route the call in case the original route fails - peak/offpeak conditions (time-based) - route traffic according to prefix I found LCR and carrierroute module, but it does not have peak/offpeak feature. Correct me if I am wrong..

2 1

new developer: Sven Knoblich
by Henning Westerholt 05 Sep '11

05 Sep '11

Hi all, I'm happy to announce a new developer for the sip-router project: Sven Knoblich. Sven is an experienced C/C++ developer in our offices in Karlsruhe and since about a year part of the team that develops our VoIP backend services. Past contributions of him were the CDR based accounting extensions in the acc module and several other bugfixes and extensions for the dialog module. He'll maintain in the future this new acc and dialog functionality and also other code contributed from … [View More]

3 2

Pass all rtp to RTPProxy
by Stas Bakulin 05 Sep '11

05 Sep '11

Hello! What is it right way to pass all rtp traffic through RTPProxy? How this can be configured in Kamailio? Stas Bakulin ---------------------------------------------------------------------- ogion(a)kvant-telecom.ru | www.kvant-telecom.ru Tel: +7 473 233 0330 (128)

2 1

Kamailio + rtpproxy talking to multiple carrier gateways - some via Firewall/NAT
by Sarat C. Vemuri 03 Sep '11

03 Sep '11

We are trying to configure Kamailio (3.1.x) as a "boarder proxy" where it acts as the front for various carrier gateways so that internal UACs and UASs are unaware of the carrier gateways. Let me try to present a clear picture of our setup. 1. Kamailio has several NICs (physical or vlan). Each on a different subnet. One subnet is internal/has routes for internal. Other subnets are private connections to carriers or a route to public Internet. 2. All of these subnets are non-… [View More]

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10 years of SER - statistics
by Daniel-Constantin Mierla 31 Aug '11

31 Aug '11

Hello, it is now one month till the 10 years SER event in Berlin. I am thinking to collect few statistics about usage of SER-based code at this moment (SER, OpenSER, Kamailio, OpenIMSCore ...) and present them during the conference and publish on the web. Everything is going to be like a combined report, not individual listing. Therefore, if you want to participate, your name or company won't be mentioned. Here are the stats I thought of: - type of usage: production, evaluation (… [View More]

2 3

Resend INVITE in case of failure
by Mino Haluz 31 Aug '11

31 Aug '11

Hi, I would like to resend the INVITE to another gateway if the original fails (503, 500, etc.). Is there any module or mechanism which could do this ? Mino

4 3

Celebrating 10 years SER SIP Router in Vienna: 8th September 2011
by Klaus Darilion 31 Aug '11

31 Aug '11

Hi! If you haven't noticed yet, SER (the mother of the SIP proxy projects Openser, Kamailio, sip-router, opensips, ....) is celebrating their 10th year. There will be a main event happening in Berlin (http://sip-router.org/10-years-ser/). For those who can not travel to this event, there will be some smaller events around the world. In Vienna, this will happen on the 8th September at 7pm in the evening, organized by Andreas Granig and myself. We do not plan to have a big event with speaker, … [View More]

1 1

Kamailio setup help - needed in Hamburg / Germany
by Rene Bormann 30 Aug '11

30 Aug '11

Hello SR List, our background is classic telephony with E1 interfaces, active telephony-cards etc. We now got our telephone-plattform transformed for SIP, but all behind a firewall. Actually we receive calls from some carriers throu the firewall on our Asterisk, which has E1 links to the IVR -Plattform. Now we want to eliminate Asterisk as kinda internal PSTN gateway and direcly route the calls throu Kamailio to our IVR services. So we thought, hey Kamailio can't be that complicated to … [View More]

2 1

Kamailio 3.1.1 snmpstat help!!
by Stephen Dodge (Bistech) 30 Aug '11

30 Aug '11

Hello, I am having difficulty running the snmpstats module. We are running CentOS 5.6 and Kamailio version 3.1.1 (i386/linux). I've tried a fresh install of Kamailio, installing net-snmp from source but I can't seem to get it running - I've seen a couple of posts regarding the issue but nothing seems to work for us!! The issue I get when I start Kamailio is: 0(27523) ERROR: <core> [sr_module.c:523]: ERROR: load_module: could not open module </usr/local/lib/kamailio/modules_k/… [View More]

2 4

registered("location") not working
by Skyler 30 Aug '11

30 Aug '11

Hi all, I am attempting to only reply to NOTIFY when the request is originating from a user/device that is authenticated on kamailio already. There are 2 ways I've tried (below) but neither will work. Any idea on what I am doing wrong or suggestions on how to do it right? TIA, Skyler if(is_method("NOTIFY") && is_registered("location")) { route(ROUTE_NOTIFY); } OR .... if(is_method("NOTIFY")) { # We only reply OK if coming from authenticated … [View More]

2 3

SEMS 1.4.2 released
by Stefan Sayer 29 Aug '11

29 Aug '11

(apologies if you receive multiple copies of this mail) Hello, the SIP Express Media Server version 1.4.2 is available. Users of SEMS 1.4 versions and anyone else is recommended to upgrade to that version. Please find below the relevant changelog, the source at ftp://ftp.iptel.org/pub/sems/sems-1.4.2.tar.gz and Debian/Ubuntu packages as always at OBS https://build.opensuse.org/project/show?project=home%3Ateam-sems Many thanks to everyone who contributed with testing, bug fixes, … [View More]

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Re: [SR-Users] Kamailio as simple SIP proxy between client and SIP server.
by Edson - Lists 29 Aug '11

29 Aug '11

Hi, Henrik... Please, never, ever, forget to copy the list in a thread initiated there... other may be "curious" on it... ;) About your configuration, it seems that you just need a 'simple forward' server. It's very simple to accomplish... Make a simple forward configuration script (you can start with the provided one and cut all registration and whistles). Just be aware of all security issues binded to this choice. Ok... you still can load any of the control modules (pike, p.ex.) and … [View More]have some prevention, but even so it's not your best choice. Edson. Em 29/08/2011 12:13, Henrik Aagaard Sørensen escreveu: > Hi Edson. > > I do know the different definitions as SIP-Proxy, SIP-Server etc. (I hope). > > That's why I want Kamailio to act as a SIP-Proxy, not a SIP-Server, as > it does currently. > > As I've understood Kamailio and SIP-Router is pretty much the same > thing. Going to sip-router.org <http://sip-router.org> for downloads > also refers to Kamailio. > > So shouldn't it be possible to use Kamailio as SIP-Proxy? Or have I > misunderstood that. > > I do not want my Kamailio to take registrations etc. It should just > forward SIP messages between client and SIP-Server, as I'm not allowed > to register towards my current SIP-Server with my current IP. But my > server, where the Kamailio is running, is allowed with it's IP. > > 2011/8/29 Edson - Lists <4lists(a)gmail.com <mailto:4lists@gmail.com>> > > Hi, Henrik... > > Sip-Router is "just" a SIP Proxy... but a dam good one... > > You have to take a step back... You are mixing concepts when > affirming that SR is playing SIP-Server roles... Please take a > better look on the definitions (SIP-Proxy, SIP-Server, Registrar, > etc) and find wich system play each role. > > In your actual scenario they are mixed concepts. > > Draw your target scenario with all the alternatives. You will find > many of them. Only them you will be able to point the roles each > component can and will have to play in the chosen scenario. Each of > this "play alternatives" (roles played by each component) will give > You Pros and Cons (security, scalability, velocity, etc, etc, etc) ... > > Make Your choice, try to implement it and this list will be able to > help you when problems on SR (SER or KAMAILIO flavors) configuration > come to play. > > Edson. > > Em 29/08/2011 11:46, Henrik Aagaard Sørensen escreveu: > > This is my current setup, which works: > SIP client <-> SIP server. > > As I've currently moved to another country I would like to be > able to do > the following setup: > SIP client <-> Kamailio SIP proxy <-> SIP server. > > As the SIP server only accept packets from specific IP's. > > Is it possible? > > I've managed to setup a simple Kamailio installation on my > server, which > does have a valid IP for the SIP server. > With the Kamailio installation I've also enabled NAT with RTPProxy. > > My problem is now, that the Kamailio installation doesn't work as a > simple proxy, but actually takes registrations etc. and acts as a > "full-blown" SIP server. > > It should only take SIP messages and transfer them between > clients and > server. > > Any ideas? > > > _________________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users > mailing list > sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org> > http://lists.sip-router.org/__cgi-bin/mailman/listinfo/sr-__users <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> > > [View Less]

1 0

Kamailio as simple SIP proxy between client and SIP server.
by Henrik Aagaard Sørensen 29 Aug '11

29 Aug '11

This is my current setup, which works: SIP client <-> SIP server. As I've currently moved to another country I would like to be able to do the following setup: SIP client <-> Kamailio SIP proxy <-> SIP server. As the SIP server only accept packets from specific IP's. Is it possible? I've managed to setup a simple Kamailio installation on my server, which does have a valid IP for the SIP server. With the Kamailio installation I've also enabled NAT with RTPProxy. My problem … [View More]

2 1

Hunting Groups In SIP-Router
by afshin afzali 29 Aug '11

29 Aug '11

Hi Guys, I'm newbie in SER. I need to implement a hunting group service in SER. After some digging in the modules I could only find AVPOPS which seems to be able to use. I'll appreciate every advise on this. BEST, -- afshin

2 1

enum_pv_query
by Andrew Melton 28 Aug '11

28 Aug '11

I would like to construct an enum query using more than one pseudo variable. The result will be a non-e.164 string which is ok for this application. Ideally, it would be something like: enum_pv_query("$rU + $rp"); This would result in the username from the R-URI appended to the port of the R-URI. However, this doen't seem to work and I wind up with 1.2.1.2.5.5.5.2.1.2.1.$rp.e164.arpa. Instead of using the $rp pseudo variable, the string "$rp" is used to form the query. It should be: 1.… [View More]

4 3

Persistent dialog
by Jon Bonilla 27 Aug '11

27 Aug '11

Hi all I'm playing with the dialog module to count the calls and limit the number of calls per subscriber. After configured the proxy I've tested a call and I realize that the dialog list is not empty after the call ends. Here's a sample call log and the dialog list while the call was being stablished and 3 minutes after that: root@proxy1a:~# grep "fbrzjkplmvlauty@quenya" /var/log/kamailio.log New request - M=INVITE Dialog set mark TOTAL Proxy authentication failed New request - M=INVITE … [View More]Dialog set mark TOTAL Dialog set mark user and userout to 'd5a78dafcb93' Dialog set mark type to outgoing Call to PSTN Dialog set mark peer and peerout to '4' Relaying request Request leaving server NAT-Reply - S=100 - Connecting NAT-Reply - S=488 - Not Acceptable Here Failure route for PSTN call Unset dialog mark peer and peerout '4' Filter reply code No failover routing needed for this response code New request - M=ACK While the call was being stablished or just after it was dropped (quickly) sercmd> dlg.list hash:3837:638039689 state:1 timestart:0 timeout:0 callid:fbrzjkplmvlauty@quenya from_tag:inobo to_tag: from_uri:sip:123456789@labsip.com to_uri:sip:0043987654321@labsip.com caller_contact:sip:123456789-sbhn1naovm6bc@111.111.111.164:5060;transport=udp caller_cseq:149 caller_route_set:<sip:111.111.111.135;r2=on;lr=on;ftag=inobo;nat=yes>,<sip:111.111.111.100;r2=on;lr=on;ftag=inobo;nat=yes> callee_contact: callee_cseq: callee_route_set: caller_bind_addr:udp:111.111.111.138:5062 callee_bind_addr: hash:3837:638039690 state:5 timestart:0 timeout:0 callid:fbrzjkplmvlauty@quenya from_tag:inobo to_tag: from_uri:sip:123456789@labsip.com to_uri:sip:0043987654321@labsip.com caller_contact:sip:123456789-sbhn1naovm6bc@111.111.111.164:5060;transport=udp caller_cseq:150 caller_route_set: <sip:111.111.111.135;r2=on;lr=on;ftag=inobo;nat=yes>,<sip:111.111.111.100;r2=on;lr=on;ftag=inobo;nat=yes> callee_contact: callee_cseq: callee_route_set: caller_bind_addr:udp:111.111.111.138:5062 callee_bind_addr: After a couple of minutes: hash:3837:638039689 state:1 timestart:0 timeout:0 callid:fbrzjkplmvlauty@quenya from_tag:inobo to_tag: from_uri:sip:123456789@labsip.com to_uri:sip:0043987654321@labsip.com caller_contact:sip:123456789-sbhn1naovm6bc@111.111.111.164:5060;transport=udp caller_cseq:149 caller_route_set: <sip:111.111.111.135;r2=on;lr=on;ftag=inobo;nat=yes>,<sip:111.111.111.100;r2=on;lr=on;ftag=inobo;nat=yes> callee_contact: callee_cseq: callee_route_set: caller_bind_addr:udp:111.111.111.138:5062 callee_bind_addr: sercmd> dlg.profile_get_size total Profile:total => profile:total value: count:1 sercmd> dlg.profile_get_size user Profile:user => profile:user value: count:0 sercmd> dlg.profile_get_size userout Profile:userout => profile:userout value: count:0 sercmd> dlg.profile_get_size peer Profile:peer => profile:peer value: count:0 sercmd> dlg.profile_get_size peerout Profile:peerout => profile:peerout value: count:0 Obviously, there's no such a call. But my dialog profile TOTOAL say's there's one, which is listed by sercmd. The dialog is persistent, I guess it will be destroyed after 12 hours, but I'd like to know why it's still there if I received a negative response and the ACK for it. Thanks, Jon [View Less]

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insert_hf in branch_route
by Gustavo Garcia Bernardo 26 Aug '11

26 Aug '11

Hi, I'm trying to add some headers in a branch_route making use of insert_hf method: insert_hf("Route: <sip:10.10.10.10:5080;lr=on>\r\n"); But the forwarded message doesn't include that new header. According to the doc, insert_hf can be used in a branch_route, Am I missing something? I'm using Kamailio 3.1. G. ________________________________ Este mensaje se dirige exclusivamente a su destinatario. Puede consultar nuestra pol?tica de env?o y recepci?n de correo electr?nico en el … [View More]

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Re: [SR-Users] How to get mem_summary (SIGUSR1) on a worker process?
by Andrade Ricardo (CI/AFU1) 26 Aug '11

26 Aug '11

Hi, I'd just like to say that I found the answer to my own question in the documentation: With 3.x versions, you have to load ctl and cfg_rpc modules and execute: sercmd cfg.set_now_int core mem_dump_pkg <pid> http://www.kamailio.org/dokuwiki/doku.php/troubleshooting:memory I did also write a patch to suppress the full memory dump, because I am only interested in the first lines of the status (total usage). This way I the server doesn't get blocked - full memory dump usually takes a … [View More]

2 2

issue with tm callbacks / premature cancel
by Andrew Pogrebennyk 26 Aug '11

26 Aug '11

I've got stuck today trying to implement some sequential forking scenario. When the call comes destined to the local user, I'm trying to reach it sequentially through two alternative extensions. For each extension, I'm calling t_set_fr with a timeout of 15sec. They reply only 180 and do not pick up the phone. After that the call should go to the "last resort" PSTN number. I'm calling then t_set_fr with a timeout of 30sec. But immediately after sending INVITE to PSTN gw Kamailio generates … [View More]

2 5

Problems with pseudo-variables
by Rocío Ruz 26 Aug '11

26 Aug '11

Hello, I am using kamailio ims extensions and I am observing some problems with pseudo-variable handling. On the s-cscf the pseudo-variables are not 'translated' into the real values. When using xlog like this: xlog("L_ERR", " found pai: $ai time [$Tf] \n"); I am finding : tail -f /var/log/syslog | grep found invlcscs /usr/sbin/kamailio[21689]: ERROR: <script>: found pai: $ai time [$Tf] However on the p-cscf the same case is correctly treated: xlog("L_ERR", " found pai: $ai time [$… [View More]

2 1

Status of t_relay()
by Fabian Borot 25 Aug '11

25 Aug '11

Hi, If t_relay() fails, is it possible to retrieve the status/reason on a variable or something? I am using t_grep_status("code") and trying to catch each case that I am interested but I was wondering if there was a better way. One could do something like this: If (result > 400) { do action 1 } else { do action 2 } txs a lot fborot

3 5

report
by zmailer＠nic.funet.fi 25 Aug '11

25 Aug '11

Spam detection software, running on the system "mail.iptel.org", has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see the administrator of that system for details. Content preview: Dear user serusers(a)lists.iptel.org, mail server administrator of lists.iptel.org would like to let you know the following, We have detected that your e-mail … [View More]

1 0

Dumping Online Users stops at 2000 records
by Sebastian Damm 25 Aug '11

25 Aug '11

Hi, we have a script running on our Kamailio servers which dumps all online users with kamctl ul show, parses it and then sends it to database. (For explanation, we started doing it this way because we noticed at least with an older (open)ser release that SER would completely stop working for a couple of seconds while writing registered users to database every 5 minutes. When dumping it, we don't see any disruption.) With Kamailio 1.5 this works, with Kamailio 3.1 it works in the beginning, … [View More]

4 5

Segmentation Fault in ACC module
by Bruno Bresciani 25 Aug '11

25 Aug '11

Hi all, ACC module is generating segmentation fault at acc_db_request function in kamailio 3.1.2 with database postgres. When module try to access the struct db_func_t acc_dbf.use_table (line 389) the child proccess that execute this operation is exited by a signal 11. Someone know why this segmentation fault is gererating? How I fix this problem? Best Regards

2 1

Kamailio 3.1.1 snmpstat help!!
by Stephen Dodge (Bistech) 25 Aug '11

25 Aug '11

Hello, I am having difficulty running the snmpstats module. We are running CentOS 5.6 and Kamailio version 3.1.1 (i386/linux). I've tried a fresh install of Kamailio, installing net-snmp from source but I can't seem to get it running - I've seen a couple of posts regarding the issue but nothing seems to work for us!! The issue I get when I start Kamailio is: 0(27523) ERROR: <core> [sr_module.c:523]: ERROR: load_module: could not open module </usr/local/lib/kamailio/modules_k/… [View More]

1 0

ACC with MySqsl problem after installing Siremis
by אורי שקד - חטיבת הנדסה ורשת - Uri Shacked 25 Aug '11

25 Aug '11

Hi, I managed to setup the ACC module with MySql. Now, after installing Siremis, the ACC table and the Missed_calls table are empty and do not populae the data from the call I make. Any ideas why? BR, Uri

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Re: [SR-Users] ACC with MySqsl problem after installing Siremis
by Daniel-Constantin Mierla 25 Aug '11

25 Aug '11

On 8/25/11 1:29 PM, אורי שקד - חטיבת הנדסה ורשת - Uri Shacked wrote: > > I managed… the softphones clients were configured with “multicast” so > they where “talking behind my back” J…. I should have noticed it earlier… > > Thanks, and sorry…. > Don't worry, it happens many time to all of us to overlook the simplest things. Whenever the solution is found it is good to keep the mailing list cc-ed, so people that will read in the future the archive for similar issues, can … [View More]get the resolution to this discussion. Cheers, Daniel > ------------------------------------------------------------------------ > > *From:*Daniel-Constantin Mierla [mailto:miconda@gmail.com] > *Sent:* Thursday, August 25, 2011 2:23 PM > *To:* SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List > *Cc:* אורי שקד- חטיבת הנדסה ורשת- Uri Shacked > *Subject:* Re: [SR-Users] ACC with MySqsl problem after installing Siremis > > Hello, > > do you get any error message in syslog from acc module. Depending on > your Linux OS, the syslog file should be /var/log/syslog or > /var/log/messages. > > Since you have acc to syslog enabled as well, do you see the ACC > records for syslog written? > > Cheers, > Daniel > > On 8/22/11 2:50 PM, אורי שקד- חטיבת הנדסה ורשת- Uri Shacked wrote: > > Hi, > > I checked and redo the changes… > > It doesn’t work. > > My config file is attached. > > I checked the DB tables and columns to see it is OK and it is….. > > Any Idea what else? > > Thanks, > > Uri > > ------------------------------------------------------------------------ > > *From:*Elena-Ramona Modroiu [mailto:ramona@asipto.com] > *Sent:* Monday, August 22, 2011 11:57 AM > *To:* SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List > *Cc:* אורי שקד- חטיבת הנדסה ורשת- Uri Shacked > *Subject:* Re: [SR-Users] ACC with MySqsl problem after installing Siremis > > Hi, > > if you installed Siremis 2.0 via the wizard, the structures of acc and > missed_calls tables were updated, so you have to change the parameters > of acc module in kamailio.cfg, like: > > http://kb.asipto.com/siremis:install20:accounting#config_file > > If you want to see the sql used to create the new acc and missed_calls > tables, see: > > http://siremis.git.sourceforge.net/git/gitweb.cgi?p=siremis/siremis;a=blob;… > > Regards, > Ramona > > > On 8/21/11 10:58 AM, אורי שקד- חטיבת הנדסה ורשת- Uri Shacked wrote: > > Hi, > > I managed to setup the ACC module with MySql. > > Now, after installing Siremis, the ACC table and the Missed_calls > table are empty and do not populae the data from the call I make. > > Any ideas why? > > BR, > > Uri > > > <http://robustus.impactia.com/AnalyticServer/redirect?cid=50f7cf7d3be2418b&m…><http://robustus.impactia.com/AnalyticServer/redirect?cid=50f7cf7d3be2418b&m…><http://robustus.impactia.com/AnalyticServer/redirect?cid=50f7cf7d3be2418b&m…><http://robustus.impactia.com/AnalyticServer/redirect?cid=50f7cf7d3be2418b&m…> > > ------------------------------------------------------------------------ > > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org <mailto:sr-users@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 <mailto:sr-users@lists.sip-router.org> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > > > -- > Daniel-Constantin Mierla --http://www.asipto.com > Kamailio Advanced Training, Oct 10-13,Berlin:http://asipto.com/u/kat > http://linkedin.com/in/miconda -- http://twitter.com/miconda -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda [View Less]

1 0

Float Comparison
by Brandon Armstead 25 Aug '11

25 Aug '11

Hello, Is there anything special that needs to be done for float comparison? For example: if([5.5 >= 4.3]) .... or if(5.5 > 4.3) .... The conditional does not seem to be coming back as true like it should? This is simply an example... the precision in the actual case is different (longer decimal places). Thank you for all and any help in advance!

4 9

[regression] Message flags set in branch_route not saved into transaction
by Alex Hermann 25 Aug '11

25 Aug '11

Hello, It seems kamailio 3 does not save message flags set in branch route into the transaction. In reply_route and failure_route the flag set in branch_route is unset. In 1.4 this used to work. I would like to get that behaviour back, is this possible? Testscenario: route { setflag(0); t_on_branch("1"); t_on_reply("1"); t_on_failure("1"); xlog("L_NOTICE", "[$rm] Request before relay: $mF"); t_relay(); } branch_route[1] { xlog("L_NOTICE", "[$rm] Branch begin: $mF"); setflag(8); … [View More]

2 5

Kamailio Freezes and no calls are processed
by Omar 25 Aug '11

25 Aug '11

The only difference is we have added some AVPs variables to process, and kamailio stops processing new calls, is not regular, but seems is related to the number of calls received. No additional calls can be processed after certain time, or maybe some amount of calls. we were unable to isolate the problem, but kamailio.fifo is removed, which never happened before. No core dump created. did anybody have a similar issue. Omar

4 5

add carrierroute module
by אורי שקד - חטיבת הנדסה ורשת - Uri Shacked 25 Aug '11

25 Aug '11

Hi, I am trying to add the carrierroute module. (I installed Kamailio 3.1.4 with mysql). I add to the cfg file the ‘loadmodule “carrierroute.so”’ and all the modparams needed. When I started Kamailio I encountered errors. One of them was that the carrierroute.so could not be found…. So, can I add this module without the need to reinstall everything again? Please advice, Thanks, Uri

3 2

SIP to SMS/HTTP
by Mark Anthony C. Delfin 24 Aug '11

24 Aug '11

Hello list, I am not sure if this has been ask before but is there a way that Kamailio can sip received SIP method MESSAGES and then call a URL (Kannel URL). Thank you. Regards, Mark Anthony C. Delfin

3 2

WARNING: timer: add_timeout: 0 expire timer added
by Alex Hermann 23 Aug '11

23 Aug '11

Hello, I get this warning message when a request is relayed over tcp: WARNING: <core> [timer_funcs.h:119]: WARNING: timer: add_timeout: 0 expire timer added The warning is displayed after send_route() has run. (relevant) settings: modparam("tm", "fr_timer", 500) modparam("tm", "fr_inv_timer", 80000) modparam("tm", "wt_timer", 20000) Every other timer related setting is at its default. Just before t_relay() is called, fr timer is set with: t_set_fr($avp(ringtime) * 1000); where $… [View More]

2 4

uac_req: pseudovariables, avps
by Jon Bonilla 23 Aug '11

23 Aug '11

Hi all I'm trying to generate a request using uac_req in a failure route when the call has been canceled. (Kamailio 3.1.3) Here's the code: failure_route[FAILURE_ROUTE_LALA] { if t_is_canceled() { $uac_req(method) = "INVITE"; $uac_req(ouri) = "sip:127.0.0.1:5090"; $uac_req(ruri) = $ru; $uac_req(furi) = $fu; $uac_req(turi) = $tu; $uac_req(hdrs) = "X-PUSH-Type: mytype\r\nX-PUSH-CLI: $avp(s:first_caller_cli)\r\nX-PUSH-DST: $rU\r\n"; uac_req_send(); route(… [View More]

2 2

max branches
by Jon Bonilla 23 Aug '11

23 Aug '11

Hi all Is there any configuration option in kamailio 3.1 to set the max number of branches? I have a system that needs to run over 30 times to failure_route and call append_brach() from there. At 11-12 iteractions I get the following errors: ERROR: tm [t_fwd.c:651]: ERROR: add_uac: maximum number of branches exceeded ERROR: tm [t_fwd.c:1528]: ERROR: t_forward_nonack: failure to add branches ERROR: tm [tm.c:1368]: ERROR: w_t_relay_to: t_relay_to failed ERROR: tm [t_reply.c:965]: ERROR: … [View More]

4 4

Testing upcoming version 3.2.0
by Daniel-Constantin Mierla 23 Aug '11

23 Aug '11

Hello, for those looking to test v3.2.0 and need some instructions, for Kamailio flavour is a wiki page showing step-by-step installation: * http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-devel-from-git Practically 3.2.0 is the current devel branch. Testing SER flavour the difference is just using FLAVOUR=ser when compiling the application. Developers as well as users are encouraged to add details about the new additions in v3.2.0 to wiki page: * http://sip-router.org/… [View More]

1 0

Development branch frozen for v3.2.0
by Daniel-Constantin Mierla 23 Aug '11

23 Aug '11

Hello, since several hours ago, the development of new features in master branch has been frozen, to allow proper testing for release of version 3.2.0. That means no new features should be committed to GIT master branch. Just before the release (expected in 1.5months) we will create a dedicate branch for 3.2.x series and master branch will be again open for new stuff. Meanwhile, developers can use personal branches to push code in the public space. GIT master branch will be used in … [View More]

1 0

RFC-5766 / RFC-6062 compliant TURN clients?
by Rafal Boni 23 Aug '11

23 Aug '11

I know this might not be totally on-topic for the list, but I figured the list participants might have experience in this domain, so I'm lobbing it out there... I've got some situations where I'd like to connect users who are oftentimes sitting on very locked-down networks (ie, corporate networks) to some SIP-based VoIP infrastructure, but the idea of opening random UDP ports on the networks in question is a no-go. As a result, I'm experimenting with TURN over TCP or TLS as a solution to … [View More]

2 1

E2E ACK is not logged with siptrace modul kamailio
by MÉSZÁROS Mihály 22 Aug '11

22 Aug '11

Hi, I want to see all incoming and outgoing sip messages in sip trace table, but i am missing messages. I am using TLS so i can't see really trace the conversation only this way. I am wondering if it is know limitation, or an issue: I am experiencing that E2E ACK is not logged in siptrace table. I opened a ticket with the details: http://sip-router.org/tracker/index.php?do=details&task_id=144&project=1&pa… Many Thanks, Misi

3 11

New developer - Matthew Williams
by Daniel-Constantin Mierla 22 Aug '11

22 Aug '11

Hello, a new person has joined our development squad - Matthew Williams, currently working for Flowroute, USA. He is already a developer of SEMS (SIP Express Media Server) and he going to add shortly, just in time for 3.2, some Json related extensions, including a JSON-RPC client module. His GIT commit user id is: mgw. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda … [View More]

1 0

Re: [SR-Users] decode_mime_type error
by Jijo 22 Aug '11

22 Aug '11

Hi Marius, The existing implementation is done considering the limitation of decode_mime_type but somehow missed the change in parse_content_type_hdr. I agree with you that its better to do the change in decode_mime_type as it is common for lot of other callers.. Regarding the example you pointed, According to my understanding from the code, I believe that existing implementation for parse_accept_body shall work as the mime types are predefined. So in the example you pointed for second … [View More]invocation 'some value' is not a defined mime type as it not matching with predefined types. The decode_mime_type returns end pointer and parse_accept_body shall process only with the valid mime type text/plain. another example with multiple mime types.., Accept: text/html, multipart/mixed In this case parse_accept_body returns 2 mime types as both are predefined in the list. Thanks Jijo On Thu, Aug 18, 2011 at 5:58 PM, Bucur Marius <bucur_marius_ovidiu(a)yahoo.com > wrote: > Hi Jijo, > > In my opinion, decode_mime_type is broken, and parse_accept_body does not > work as expected. > For example, this is a valid accept header: > > accept: text/plain;param=",some value". > > but the parse_accept_body will return -1 because the first return value of > decode_mime_type is ",some value\"". The second invocation will think "some > value\"" is a mime type, which obviously isn't. > > I must say I never saw such an accept header, but the standard permits it. > Indeed, a quick fix would be to ignore the return code when comma is found. > This must be done for other functions that call decode_mime_type > like has_body(mime) in textops. > > This is just my opinion, but I think this solution is a bit hackish since > the convention, the "promise" that decode_mime_type should respect is to > return a non-NULL, non-end pointer ONLY when there are more mime types in > the body. And it doesn't. > A lot of functions that call it rely on this. I think it is easier/more > elegant to fix the bug in the decode_mime_type then to change the function > convention in all callers. > Also, this might work for most callers, but parse_accept_body will still be > buggy, as I explained in the example above. > > Cheers, > Marius > > ________________________________ > From: Jijo <realjijo(a)gmail.com> > To: Bucur Marius <bucur_marius_ovidiu(a)yahoo.com> > Sent: Thursday, August 18, 2011 10:48 AM > Subject: Re: [SR-Users] decode_mime_type error > > > Hi Marius, > > Right, the same function is used for Accept and Content-Type. > > I don't think there is any change required for parsing Accept. In case of > Accept the function decode_mime_type is called in a loop until it find all > the media types. On the returning from decode_mime_type(), the main function > parse_accept_xxx() checks the pointer is 'comma' or not. if its comma, then > the function decode_mime_type called again and find the remaining media > types. So i think the existing implementation for Parsing Accept is fine. > > The only change required here is not return error for comma while parsing > the Content-Type. > > Thanks > Jijo > > > On Thu, Aug 18, 2011 at 12:32 PM, Bucur Marius < > bucur_marius_ovidiu(a)yahoo.com> wrote: > > Hello Jijo, > > > >You are right, multiple mime-types are not allowed. But it seems like > decode_mime_type had the purpose of also parsing the Accept header: > > > >Accept: text/html, image/jpeg, multipart > > > >hence the search for commas :). > >The only function that uses this feature and calls decode_mime_type in a > loop is parse_accept_body. The other functions that call decode_mime_type > just fail when returning ret != end, and since multiple mime types in > content-type are not allowed the behavior is fine. > >The bad thing is the Accept header can have mime parameters too (which can > contain comma). > > > >Accept = "Accept" HCOLON [ accept-range *(COMMA accept-range) > ] > >accept-range = media-range *(SEMI accept-param) > >media-range = ( "*/*" / ( m-type SLASH "*" ) / ( m-type SLASH > m-subtype ) ) *( SEMI m-parameter ) > > > >So I think the best thing would be to change decode_mime_type so that it > does a more thorough parsing (e.g. check for parameter) hence a parameter > value can contain commas (only if quoted). > > > > > >Cheers, > > > >Marius > >________________________________ > >From: Jijo <realjijo(a)gmail.com> > >To: Bucur Marius <bucur_marius_ovidiu(a)yahoo.com>; SIP Router - Kamailio > (OpenSER) and SIP Express Router (SER) - Users Mailing List < > sr-users(a)lists.sip-router.org> > >Sent: Thursday, August 18, 2011 8:04 AM > >Subject: Re: [SR-Users] decode_mime_type error > > > > > > > >Hi Marius, > > > >Thanks for the response.. I checked the other functions, but they don't > have the check for ret !=end, but they check the pointer and if it is comma > then loop through again until it find all the media types. > > > >As per the RFC3261 multiple media-types are not supoorted in the > Content-Type. So I'm not sure why the author used comma to determine > multiple media types . Probably just followed like Route or Record-Route > headers..?? > > > >Content-Type = ( "Content-Type" / "c" ) HCOLON media-type > >media-type = m-type SLASH m-subtype *(SEMI m-parameter) > >m-type = discrete-type / composite-type > >discrete-type = "text" / "image" / "audio" / "video" > >/ "application" / extension-token > >composite-type = "message" / "multipart" / extension-token > > > >Record-Route = "Record-Route" HCOLON rec-route *(COMMA rec-route) > > > >Thanks > >Jijo > > > > > > > >On Thu, Aug 18, 2011 at 2:23 AM, Bucur Marius < > bucur_marius_ovidiu(a)yahoo.com> wrote: > > > >Hello Jijo, > >> > >>It seems like the decode_mime_type is a somehow broken. The comma is very > well allowed in boundary, as you said. The BNF specified in RFC2046 permits > it. > >>But, the decode_mime_type function ignores everything coming after comma. > More than that, it notifies the function caller that this content type has > multiple mime types. > >>I think the author of the function thought of something like: > >> > >>Content-Type: text/html, image/jpeg // very weird, though > >> > >>This is wrong, hence the comma can be in a parameter (like in your case). > >>I think the best fix, as you said is to remove that check, hence a > non-NULL return value doesn't mean anything (that the content type has > multiple mime types). > >>Another fix is to write a "good" decode_mime_type, that checks if the > comma is inside a parameter. I don't know if effort is worth it. > >> > >>One thing we need to do is to check if there are any functions in > Kamailio that call decode_mime_type and also perform this check (ret != > end). > >> > >>Cheers, > >>Marius > >> > >> > >>________________________________ > >>From: Jijo <realjijo(a)gmail.com> > >>To: sr-users(a)lists.sip-router.org; sr-dev(a)lists.sip-router.org > >>Sent: Wednesday, August 17, 2011 10:54 AM > >>Subject: [SR-Users] decode_mime_type error > >> > >> > >> > >>Hi All, > >> > >>The function parse_content_type_hdr() is failing in decode_mime_type() > when Content-Type parameter "boundary" has value comma as below. The error > message is "ERROR:parse_content_type_hdr: CONTENT_TYPE hdr contains " > >> "more then one mime type :-(! > >> > >>Content Type Header is as below, It works fine if the boundary value is > without comma. > >> > >>Content-Type: multipart/mixed;boundary=",AW" > >> > >>The parse_content_type_hdr() is failing in the following code, > >> > >> ret = decode_mime_type(msg->content_type->body.s, end , &mime); > >> if (ret==0) > >> goto error; > >> if (ret!=end) { > >> > >> LOG(L_INFO,"ERROR:parse_content_type_hdr: CONTENT_TYPE hdr > contains " > >> "more then one mime type :-(!\n"); > >> goto error ; > >> } > >> > >>I thought of fixing this issue by commenting the code for condition ret > !=end. I'm not sure why we we need that check, as mime variable has the > information to process the content. > >> > >>Please let me know if you see any impact. > >> > >>Thanks > >>Jijo > >> > >> > >> > >> > >> > >>_______________________________________________ > >>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 > >> > > > [View Less]

2 2

min expires question
by Fabian Borot 22 Aug '11

22 Aug '11

Hello This is how my registrar module is configured, hoping that min-expires set to 30 wont allow registration attempts more often than 30 seconds, but if I configure a UAC to register every 15 seconds I see that kamailio allows it. Shouldn't it reply with "422 Session Interval Too Small" or "423 Interval Too Brief"? # ----- registrar params ----- modparam("registrar", "default_expires", 120) modparam("registrar", "min_expires", 30) modparam("registrar", "method_filtering", 1) /* uncomment … [View More]

3 4

Sip Router Issues (Topoh's module)
by Richard DEMONGEOT 22 Aug '11

22 Aug '11

Dear all, I've encountered two issues with Sip Router. Build : Kamailio 3.1.4 for Squeeze (Kamailio's debian Packages). When i'm using Topoh with Call-ID encoding, instruction dlg_bye("ALL") generate me an error on logfiles : Aug 4 17:46:26 kamtest /usr/sbin/kamailio[25267]: ERROR: topoh [th_mask.c:166]: invalid input string"dd04db6ccc7074deb9f8a1160946d331@0:0:0:0:0:0:0:0" Aug 4 17:46:26 kamtest /usr/sbin/kamailio[25267]: ERROR: topoh [th_msg.c:480]: cannot decode callid Aug 4 17:46:… [View More]

2 5

10 Years SER Conference, Berlin - full booked
by Daniel-Constantin Mierla 22 Aug '11

22 Aug '11

Hello, the 10 years SER conference in Berlin is full booked, meaning that new registrations will be accepted only if someone that registered previously will cancel his/her participation. Since we just discovered such a situation, in the case you did register and you haven't received a confirmation email (not the one automatically generated at registration time, but a second one), please reply to me or write to registration(a)lists.sip-router.org telling the date when you did the … [View More]

1 0

Freezing development for v3.2.0 today
by Daniel-Constantin Mierla 22 Aug '11

22 Aug '11

Hello, just a short reminder that we will close the development of new features for version 3.2.0 at the end of today, GMT. Then the focus should be on testing (e.g., bug fixing) and enhancements to documentation. If you have something to commit, do it today, or if it is impossible for you to make it in the GIT repo today, let us know and we will see what can be done. But don't forget, 3.3.0 will come as well in the no-so-far future. Cheers, Daniel -- Daniel-Constantin Mierla -- http:/… [View More]

1 0

masking carriers
by Anders 20 Aug '11

20 Aug '11

Hi, Background: Kamailio 1.5.x + rtpproxy I am not too crazy about telling our customers where we send our calls. But if they take a look at the sip dialog as it is right now, they can see our suppliers/carriers IP addresses. What can we do to mask this, so that the last piece of information stops at our IPs? We are running rtpproxy for the media - if that makes any difference. Someone told me I cannot mask anything without moving to Kamailio 3.x - is that the case? Thanks!! //Anders

2 2

Question about encryption with Kamailio
by margot.basa 19 Aug '11

19 Aug '11

Hello all, I would like to use Kamailio to encrypt contents of SIP messages (using SIP TLS) between 2 endpoints, i.e.: - To use 5061 port instead of 5060 port, - To use sips uri instead of sip uri... For example, T1 and T2 communicates with "Server A" like that: 1) T1 and T2 send REGISTER to "Server A" 2) T1 and T2 received 200 OK from "Server A" ... 3) "Server A" sends an INVITE message to T1 and T2 ... 4) RTP flow between T1 and T2 (this should not be encrypted) ... 5) "Server A" sends a … [View More]

2 2

New developer: Alexandr Dubovikov
by Daniel-Constantin Mierla 19 Aug '11

19 Aug '11

Hello, a new person has now developer access to GIT repository - Alexandr Dubovikov, contributor in the past with patches, is going to work now on extensions related to Homer project - turning the SIP server in a SIP capturing server. His developer id is adubovikov. You can read more about Homer project at: * http://code.google.com/p/homer/ * http://www.kamailio.org/events/2011-Cluecon/AD-Homerv2.pdf Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio … [View More]

3 2

new functionality: CDR-based accounting
by Timo Reimann 19 Aug '11

19 Aug '11

Hi all, as announced quite a while ago, I finally checked in code that allows to produce CDRs (Call Data Records) directly from SIP-Router and generate logs accordingly. The main code portion resides in modules_k/acc and provides a switch to enable basic CDR generation including start time, end time, and duration. Analogous to the existing logging approach, you may define an extra parameter covering to-be-included dialog pseudo-variables that must be assigned in the configuration script. The … [View More]

5 7

LCR Modules Comparison
by Spencer Thomason 19 Aug '11

19 Aug '11

Hello all, Can anyone shed some light into the differences between the available LCR modules? We have migrated our config from OpenSIPS where we used the drouting module to Kamailio where we are using the lcr module. Our ruleset is not crazy huge, about 100k entries. Are there compelling reasons to choose one module over the other? Is there one that has the most active development behind it and is a "standard" for the platform or is there one that will be deprecated in the foreseeable future? Thanks, Spencer

4 3

Re: [SR-Users] decode_mime_type error
by Bucur Marius 19 Aug '11

19 Aug '11

Hi Jijo, In my opinion, decode_mime_type is broken, and parse_accept_body does not work as expected. For example, this is a valid accept header: accept: text/plain;param=",some value". but the parse_accept_body will return -1 because the first return value of decode_mime_type is ",some value\"". The second invocation will think "some value\"" is a mime type, which obviously isn't. I must say I never saw such an accept header, but the standard permits it. Indeed, a quick fix would be … [View More]to ignore the return code when comma is found. This must be done for other functions that call decode_mime_type like has_body(mime) in textops. This is just my opinion, but I think this solution is a bit hackish since the convention, the "promise" that decode_mime_type should respect is to return a non-NULL, non-end pointer ONLY when there are more mime types in the body. And it doesn't. A lot of functions that call it rely on this. I think it is easier/more elegant to fix the bug in the decode_mime_type then to change the function convention in all callers. Also, this might work for most callers, but parse_accept_body will still be buggy, as I explained in the example above. Cheers, Marius ________________________________ From: Jijo <realjijo(a)gmail.com> To: Bucur Marius <bucur_marius_ovidiu(a)yahoo.com> Sent: Thursday, August 18, 2011 10:48 AM Subject: Re: [SR-Users] decode_mime_type error Hi Marius, Right, the same function is used for Accept and Content-Type. I don't think there is any change required for parsing Accept. In case of Accept the function decode_mime_type is called in a loop until it find all the media types. On the returning from decode_mime_type(), the main function parse_accept_xxx() checks the pointer is 'comma' or not. if its comma, then the function decode_mime_type called again and find the remaining media types. So i think the existing implementation for Parsing Accept is fine. The only change required here is not return error for comma while parsing the Content-Type. Thanks Jijo On Thu, Aug 18, 2011 at 12:32 PM, Bucur Marius <bucur_marius_ovidiu(a)yahoo.com> wrote: Hello Jijo, > >You are right, multiple mime-types are not allowed. But it seems like decode_mime_type had the purpose of also parsing the Accept header: > >Accept: text/html, image/jpeg, multipart > >hence the search for commas :). >The only function that uses this feature and calls decode_mime_type in a loop is parse_accept_body. The other functions that call decode_mime_type just fail when returning ret != end, and since multiple mime types in content-type are not allowed the behavior is fine. >The bad thing is the Accept header can have mime parameters too (which can contain comma). > >Accept = "Accept" HCOLON [ accept-range *(COMMA accept-range) ] >accept-range = media-range *(SEMI accept-param) >media-range = ( "*/*" / ( m-type SLASH "*" ) / ( m-type SLASH m-subtype ) ) *( SEMI m-parameter ) > >So I think the best thing would be to change decode_mime_type so that it does a more thorough parsing (e.g. check for parameter) hence a parameter value can contain commas (only if quoted). > > >Cheers, > >Marius >________________________________ >From: Jijo <realjijo(a)gmail.com> >To: Bucur Marius <bucur_marius_ovidiu(a)yahoo.com>; SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List <sr-users(a)lists.sip-router.org> >Sent: Thursday, August 18, 2011 8:04 AM >Subject: Re: [SR-Users] decode_mime_type error > > > >Hi Marius, > >Thanks for the response.. I checked the other functions, but they don't have the check for ret !=end, but they check the pointer and if it is comma then loop through again until it find all the media types. > >As per the RFC3261 multiple media-types are not supoorted in the Content-Type. So I'm not sure why the author used comma to determine multiple media types . Probably just followed like Route or Record-Route headers..?? > >Content-Type = ( "Content-Type" / "c" ) HCOLON media-type >media-type = m-type SLASH m-subtype *(SEMI m-parameter) >m-type = discrete-type / composite-type >discrete-type = "text" / "image" / "audio" / "video" >/ "application" / extension-token >composite-type = "message" / "multipart" / extension-token > >Record-Route = "Record-Route" HCOLON rec-route *(COMMA rec-route) > >Thanks >Jijo > > > >On Thu, Aug 18, 2011 at 2:23 AM, Bucur Marius <bucur_marius_ovidiu(a)yahoo.com> wrote: > >Hello Jijo, >> >>It seems like the decode_mime_type is a somehow broken. The comma is very well allowed in boundary, as you said. The BNF specified in RFC2046 permits it. >>But, the decode_mime_type function ignores everything coming after comma. More than that, it notifies the function caller that this content type has multiple mime types. >>I think the author of the function thought of something like: >> >>Content-Type: text/html, image/jpeg // very weird, though >> >>This is wrong, hence the comma can be in a parameter (like in your case). >>I think the best fix, as you said is to remove that check, hence a non-NULL return value doesn't mean anything (that the content type has multiple mime types). >>Another fix is to write a "good" decode_mime_type, that checks if the comma is inside a parameter. I don't know if effort is worth it. >> >>One thing we need to do is to check if there are any functions in Kamailio that call decode_mime_type and also perform this check (ret != end). >> >>Cheers, >>Marius >> >> >>________________________________ >>From: Jijo <realjijo(a)gmail.com> >>To: sr-users(a)lists.sip-router.org; sr-dev(a)lists.sip-router.org >>Sent: Wednesday, August 17, 2011 10:54 AM >>Subject: [SR-Users] decode_mime_type error >> >> >> >>Hi All, >> >>The function parse_content_type_hdr() is failing in decode_mime_type() when Content-Type parameter "boundary" has value comma as below. The error message is "ERROR:parse_content_type_hdr: CONTENT_TYPE hdr contains " >> "more then one mime type :-(! >> >>Content Type Header is as below, It works fine if the boundary value is without comma. >> >>Content-Type: multipart/mixed;boundary=",AW" >> >>The parse_content_type_hdr() is failing in the following code, >> >> ret = decode_mime_type(msg->content_type->body.s, end , &mime); >> if (ret==0) >> goto error; >> if (ret!=end) { >> >> LOG(L_INFO,"ERROR:parse_content_type_hdr: CONTENT_TYPE hdr contains " >> "more then one mime type :-(!\n"); >> goto error ; >> } >> >>I thought of fixing this issue by commenting the code for condition ret !=end. I'm not sure why we we need that check, as mime variable has the information to process the content. >> >>Please let me know if you see any impact. >> >>Thanks >>Jijo >> >> >> >> >> >>_______________________________________________ >>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 >> > [View Less]

2 1

The not so simple Kamailio Load Balancing of Asterisk
by Nick Khamis 19 Aug '11

19 Aug '11

Hello Everyone, I saw the simple load balancing of asterisk using x-SER documentation here "http://www.kamailio.org/dokuwiki/doku.php/asterisk:load-balancing-and-ha". Are there any up-tp-date not so simple documentation for setting up HA and LB asterisk instances using x-SER. Thanks in Advance, Nick Khamis Toronto Hydro Telecom

5 7

decode_mime_type error
by Jijo 18 Aug '11

18 Aug '11

Hi All, The function parse_content_type_hdr() is failing in decode_mime_type() when Content-Type parameter "boundary" has value comma as below. The error message is "ERROR:parse_content_type_hdr: CONTENT_TYPE hdr contains " "more then one mime type :-(! Content Type Header is as below, It works fine if the boundary value is without comma. Content-Type: multipart/mixed;boundary="*,*AW" The parse_content_type_hdr() is failing in the following code, ret = decode_mime_type(msg-… [View More]

2 2

Re: [SR-Users] mtree question
by Javier Gallart 18 Aug '11

18 Aug '11

Hi Juha based on my experience with the mtree module: -mt_match() will match the longest prefix in the tree, in your case 00358 -mt_ignore_duplicates makes sense when loading the data in the DB to the memory. There could be duplicate entries in the db and with that directive you tell kamailio how to handle that situation -I don't know about mt_tree_type...(never needed to test it) Hope it helps Regards Javi > ------------------------------ > > Message: 2 > Date: Fri, 12 Aug … [View More]

3 5

lookup for specified user
by Efelin Novak 17 Aug '11

17 Aug '11

Hi folks, I discussed this issue in previous email ( [SR-Users Modifying $rU after lookup() ) but the question is a bit different now. I would like to change uri like lookup() function does it, but for the specific user. Problem is I want to have one registration (X) for several numbers. When this number is called I have to perform lookup() for this one common registration (X) and send this call there. To do this I changed $rU just before lookup() for X, and after lookup() I … [View More]

3 5

siptrace loosing an ACK
by Timo Klecker 17 Aug '11

17 Aug '11

Hello We are using the siptrace module to save the sip-messages to our database. All the messages are logged except one message. After INVITE, 100 trying and 200 OK we loose the outbound ACK which is forwarded by Kamailio statelessly in t_funcs.c. Is there any way to get a siptrace of this message, too? I've made a picture of the sip-messages and the lost ACK. Kind regards Timo Klecker

3 3

Error message
by Jon Bonilla 16 Aug '11

16 Aug '11

I have an error message in kamailio's log repeated many times. I tried to google it but no results (I think it's forst time I see no results in a google search) /usr/sbin/kamailio[3769]: ERROR: ctl [binrpc_run.c:942]: ERROR: binrpc: rpc_printf: binrpc_addstr failed: buffer too small (overflow) (-2) What does it mean? Any ideas? thanks, Jon

2 1

openSIPS vs Kamailio vs SIP-Router
by Nick Khamis 15 Aug '11

15 Aug '11

Hello Everyone, I can only imagine how many times this question has come up since post 2008. Please forgive my reoccurring of the issue. We are looking to provide carrier grade sip services to our clients world wide. What we need is a lightweight, robust and scalable solution that will allow us to terminate sip calls to our different carriers. Performance, and high throughput are factors very important to my employer. Features such as caller authentication, database back-end, load balancing, … [View More]

4 5

Kamailio presentations at Cluecon 2011
by Daniel-Constantin Mierla 15 Aug '11

15 Aug '11

Hello, I uploaded my slides presented at Cluecon 2011, last week in Chicago. In the first part the focus was on what has been happening in the last year around the project, on the second part showing some example of using it to secure VoIP networks. The PDF is available at: * http://asipto.com/u/42 Alexandr Dubovikov presented the recently announced Homer project which turns Kamailio in a scalable capturing system for SIP traffic, the slides for this talk are available at: * http:… [View More]

3 2

10 Years SER Conference - update
by Daniel-Constantin Mierla 15 Aug '11

15 Aug '11

Hello, just wanted to let you know that only few seats are still available for the 10 Years SER Conference, on Sep 2, 2011, in Berlin, Germany -- more details at: * http://sip-router.org/10-years-ser/ If you plan to attend, do not wait any longer and register as soon as possible (form at: http://sip-router.org/registration-form/). The host FhG Fokus Institute has a medium size conference room, with no option to expand -- I expect the registration to be closed very soon. Cheers, … [View More]

1 0

freezing for major release 3.2.0
by Daniel-Constantin Mierla 15 Aug '11

15 Aug '11

Hello, based on the last devel IRC meeting, the next major release should happen sometime by end of September, therefore we should freeze the development and go into testing phase about one month before. I propose Monday, August 22, the day to freeze development for version 3.2.0. By then, I hope every developer will be able to push to master branch what he/she wants to have in 3.2.0. Write here to mailing lists if you thing about alternatives. The very good side of 3.2.0 is that we … [View More]

1 1

Wrong formatting of Request URI
by Mino Haluz 15 Aug '11

15 Aug '11

Hi, my kamailio is sometimes receiving ruri from remote clients which I assume are not right formatted. For ex.: sip:user@kamailio=3Buser=3Dphone Is there any mechanism which could transform this into something readable for kamailio or do I have to simply cut off everything after the host name of the RURI ? Mino

2 1

pua_xmpp questions
by laura testi 13 Aug '11

13 Aug '11

Hi all, we are trying to setup a kamailio presence server in SIP/SIMPLE domain to interwork with XMPP domains including the GTalk (see the config below). The architecture is like this: - SIP/SIMPLE server: kamailio3.14 with full configuration with XCAP integration - XMPP Gateway: another kamailio server + ejabberd server - component mode - xmpp moduleL: outbound proxy to the Kamailio SIP/SIMPLE server - pua_xmpp: no outbound proxy? - XMPP server: ejabberd The users from XMPP domain … [View More]can add the SIP account and see the presence of the SIP users, but not vice versa. The chat from both directions works fine. in the Presence Server, we have configured the kamailio with XCAP integration. We have the following doubts: - does pua_xmpp/pua modules need the xcap integration for the presence integration with xmpp domains? because to use xcap for presence authorization rules, it needs the sip clients support the xcap - does pua_xmpp/pua support xcap? otherwise how to works? - in case of multiple SIP/SIMPLE presence server, how we can configure the server_address of pua_xmpp and presence parameter in xmpp gw? Can you help us to clarify the doubts please? Many thanks in advanced! Best Regards, Laura PS: following are the main configuration of the xmpp GW: --------------------------------------------------------------------------------------------- ... modparam("xmpp", "domain_separator", "*") modparam("xmpp", "backend", "component") modparam("xmpp", "gateway_domain", "<mygwdomain>") modparam("xmpp", "xmpp_domain", "<mygwdomain>") modparam("xmpp", "xmpp_host", "127.0.0.1") modparam("xmpp", "xmpp_password", "secret") modparam("xmpp", "outbound_proxy", "<my oubound proxy uri>") modparam("pua", "outbound_proxy", "<my outbound proxy uri>") modparam("pua", "update_period", 60) modparam("pua", "default_expires", 1200) modparam("presence", "server_address", "<my presence server uri>") modparam("pua_xmpp", "server_address", "<my presence server uri>") route{ route(REQINIT); t_check_trans(); if (uri =~ "sip:.+@.*<myxmppdomain>") { route(PRESENCE); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } route(CHAT); } xlog("L_INFO", "*** xmpp: unhandled message type\n"); t_reply("503", "Service unavailable"); return; } ... route[CHAT] { if(!is_method("MESSAGE")) return; if (!t_newtran()) { sl_reply_error(); exit; } xlog("L_INFO", "*** xmpp-handled MESSAGE message."); if ($cT=~"^text/plain") { if (xmpp_send_message()) { t_reply("200", "Accepted"); } else { t_reply("404", "Not found"); } } else { xlog("L_INFO", "*** xmpp-handled MESSAGE, ignoring not text messages"); t_reply("200", "Accepted"); } t_release(); exit; } route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE|NOTIFY")) return; # create new transaction to catch retransmissions if (!t_newtran()) { sl_reply_error(); exit; } if( is_method("NOTIFY")) { xlog("L_INFO", "*** xmpp-handled NOTIFY message."); if(pua_xmpp_notify()) t_reply("200", "OK"); t_release(); exit; } else if( is_method("SUBSCRIBE")) { xlog("L_INFO", "*** xmpp-handled SUBSCRIBE message.\n"); handle_subscribe(); if($(hdr(Event))== "presence") { pua_xmpp_req_winfo("$ruri", "$hdr(Expires)"); } t_release(); exit; } else if(is_method("PUBLISH")) { handle_publish(); t_release(); exit; } } ... [View Less]

2 8

subscriber table reload
by Fabian Borot 12 Aug '11

12 Aug '11

Hello When one creates a user using the cli command like this: "kamctl add 200 abc", it is my understanding that the information goes to the subscriber table and into memory. Is that right? But if one populates the subscriber table manually [adding a record on the table w/o the /kamctlcli command], do we need to "load" those accounts into memory before a client tries to register or does kamailio looks it up when the registration request comes in at that time? What we are trying to do is to … [View More]

4 5

mtree question
by Juha Heinanen 12 Aug '11

12 Aug '11

after reading mtree readme, it is not clear to me if mt_match() matches to longest or any matching prefix in the three. for example, if i have tree with prefixes 00 and 00358 and string to be matched is 0035892345670, will mt_match() set pv to value associated with 00 or 00358? also, if mt_ignore_duplicates is set, will 00358 be considered as duplicate of 00 or just prefixes that are exactly same. finally, what are possible mt_tree_type values and what types they stand for? -- juha

1 0

kamailio - proxy/relay
by Sunny 11 Aug '11

11 Aug '11

Hi list, I'm trying to configure kamailio to work as a OBP (Outbound Proxy). Basically KAMAILIO receive requests from UAs, after that it acts like a client and send those requests to the IP-PBX. On the same way the responses are forwarded from IP-PBX to the UAs. [UA] --> [KAMAILIO] --> [IP-PBX ] [UA] <-- [KAMAILIO] <-- [IP-PBX ] Following possible call flow: [UA ] -> [KAMAILIO] SIP/SDP Request: INVITE sip:12345678909@[KAMAILIO], with session description [KAMAILIO] -&… [View More]

3 4

time for acc records
by Daniel-Constantin Mierla 11 Aug '11

11 Aug '11

Hello, it seems that is some parts of the world it is required to have call duration with two decimals precision. By default acc module does the timestamp in seconds. Thinking of adding an option to save the milliseconds, I see two option: - a new column to store the milliseconds part as integer - a new column to store the seconds.milliseconds as double For those using various billing engines, what would be easier to digest when rating the CDRs? There is already the option of doing db … [View More]

4 5

Re: [SR-Users] Default kamailio installation, syntax errors with Defined Values
by Gertjan Wolzak 11 Aug '11

11 Aug '11

Sorry, my bad.... Missing dependencies Regards, Gertjan Wolzak

1 0

Default kamailio installation, syntax errors with Defined Values
by Gertjan Wolzak 11 Aug '11

11 Aug '11

Hello, Just been installing Kamailio, version 3.1.x via Git, but when I use the default kamailio.cfg, Kamailio will not start and the debug gives the following errors. 0(25876) : <core> [cfg.y:3330]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 285, column 29-35: syntax error 0(25876) : <core> [cfg.y:3330]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 285, column 29-35: Invalid arguments 0(25876) : <core> [cfg.y:3333]… [View More]

1 0

Using Kamailio to convert TCP to TLS
by aur.hoffmann＠free.fr 10 Aug '11

10 Aug '11

Hello, I would like to use Kamailio as a TCP <=> TLS convertor ? Do you think it's possible ? I have developed a software (which simulate a SIP phone) which can use UDP or TCP (not TLS). I need a SIP proxy to convert TCP to TLS. For example: UA => TCP OPTION (5060) => Kamailio => TLS OPTION (5061) => OXE UA <= TCP 200 OK (5060) <= Kamailio <= TLS 200 OK (5061) <= OXE Do you think Kamailio can do that ? Thanks in advance, Regards

3 2

$hu is not documented
by Juha Heinanen 10 Aug '11

10 Aug '11

xcap_server examples refer to pseudo variable $hu, which is not documented on pseudo variables wiki page: http://sip-router.org/wiki/cookbooks/pseudo-variables/devel -- juha

2 2

Cancel issue
by Carl Wagner 09 Aug '11

09 Aug '11

Hi, I am having an issue getting Kamailio to properly cancel a call. I initiate the call with an INVITE from Asterisk and then hang up. According to rfc3665 the call flow should be: 3.8. Unsuccessful No Answer Alice Proxy 1 Proxy 2 Bob | | | | | INVITE F1 | | | |--------------->| INVITE F2 | | | 100 F3 |--------------->|… [View More] INVITE F4 | |<---------------| 100 F5 |--------------->| | |<---------------| | | | | 180 F6 | | | 180 F7 |<---------------| | 180 F8 |<---------------| | |<---------------| | | | CANCEL F9 | | | |--------------->| | | | 200 F10 | | | |<---------------| CANCEL F11 | | | |--------------->| | | | 200 F12 | | | |<---------------| | | | | CANCEL F13 | | | |--------------->| | | | 200 F14 | | | |<---------------| | | | 487 F15 | | | |<---------------| | | | ACK F16 | | | 487 F17 |--------------->| | |<---------------| | | | ACK F18 | | | 487 F19 |--------------->| | |<---------------| | | | ACK F20 | | | |--------------->| | | In my case I only have one proxy - the Kamailio box, and the call is coming from Asterisk instead of Alice. The is the call flow that I am seeing: Asterisk Kamailio SIP_phone | | | | INVITE | | |--------------->| INVITE | | 100 |--------------->| |<---------------| 100 | | |<---------------| | | | | | 180 | | 180 |<---------------| |<---------------| | | CANCEL | | |--------------->| | | | CANCEL | | |--------------->| | 200 Canceling | | |<---------------| | | | 200 | | |<---------------| | | | | | | | | 487 | | |<---------------| | | ACK | | 487 |--------------->| |<---------------| | | ACK | | |--------------->| | | CANCEL | | |--------------->| | | 200 Canceling | | |<---------------| | | CANCEL | | |--------------->| | | 200 Canceling | | |<---------------| | It looks like the problem is that that "200 Canceling" has a To: tag and it shouldn't. ====================================== # cancel from Asterisk: U 2011/05/24 15:18:56.123714 3.3.3.3:5060 -> 2.2.2.2:5060 CANCEL sip:SIP_13034445555@2.2.2.2 SIP/2.0. Via: SIP/2.0/UDP 3.3.3.3:5060;branch=z9hG4bK51c9602a;rport. From: "+13032223333" <sip:+13032223333@3.3.3.3>;tag=as4c415d87. To: <sip:SIP_13034445555@2.2.2.2>. Call-ID: 605b8de40a05924177085d71154bda77(a)3.3.3.3. CSeq: 102 CANCEL. User-Agent: VerbalWorld. Max-Forwards: 70. Content-Length: 0. # 200 canceling from Kamailio to asterisk: U 2011/05/24 15:18:56.127349 2.2.2.2:5060 -> 3.3.3.3:5060 SIP/2.0 200 canceling. Via: SIP/2.0/UDP 3.3.3.3:5060;branch=z9hG4bK51c9602a;rport=5060. From: "+13032223333" <sip:+13032223333@3.3.3.3>;tag=as4c415d87. To: <sip:SIP_13034445555@2.2.2.2>;tag=4ed8ab9f3ed29a77613bb14b3431ff43-2f34. Call-ID: 605b8de40a05924177085d71154bda77(a)3.3.3.3. CSeq: 102 CANCEL. Server: kamailio (3.1.3 (i386/linux)). Content-Length: 0. ====================================== Why is kamialio adding a To: tag? Portion of kamailio.cfg ===================== ... route { ... route(WITHINDLG); ### only initial requests (no To: tag) from here on if (is_method("CANCEL")) { if (t_check_trans()) { xlog("L_INFO", "MAIN: Recieved Cancel, t_check_trans is true\n"); ... t_relay(); exit; } ... Thanks! Carl [View Less]

3 9

Modifying $rU after lookup()
by Efelin Novak 09 Aug '11

09 Aug '11

Hi Folks, I would like to ask how can I change the $rU (user part of uri) in multiple appended branches? In my scenario I do lookup("location") and after this I try to modify an username using the $rU variable. This works fine until there are several records in the location table for the given user (registered from different locations). In this case only one branch is changed and unchanged uri is sent to an another location. Is there a way to change it in all branches together? Thanks for … [View More]

5 7

Re: [SR-Users] Using SR for Number Translation & Number Portabilty Applications
by Henning Westerholt 09 Aug '11

09 Aug '11

On Monday 08 August 2011, Yaron Nachum wrote: > I have seen the match_mode but I haven't seen how is it possible to set it > up per domain. > > I would appreciate if you could explain how. Hi Yaron, please always CC to the mailling list. Its not possible to set the match_mode per domain, its a global parameter. With regards to the domains, this was another option you could use, e.g. use domain 0 for calls without the 'D' prefix, and then domain D for calls with 'D' prefix. Best regards, Henning

1 0

Re: [SR-Users] Using SR for Number Translation & Number Portabilty Applications
by Henning Westerholt 08 Aug '11

08 Aug '11

> Yaron wrote: > Thanks for the quick response. I have already played with the sqlops - > this is an obvious way to do it. I will further investigate cr and sqlops > and decide which is best for my purposes. Hello Yaron, > I liked the idea of having several instances of SIP-ROUTERs which are > independent of each other- I understand that cr answers it. Indeed, this is one advantage of using an in-memory cache. But you probably need some more DB access on the servers for … [View More]

1 0

Using SR for Number Translation & Number Portabilty Applications
by Yaron Nachum 08 Aug '11

08 Aug '11

Hi Everyone, I have downloaded SR r 3.1.4 and have been playing around with it. I am considering implementing Number Translation services (1800...) and Number Portability Services on it. We have around 20K entries on the Number Translation Service and around 3M entries on the Number portability Service. The volume is around 100 caps. I have started working with the carrierroute module - it seems perfect for the Number translation Service, but I am not sure about the Number portability. I have … [View More]

2 1

suggested number of TCP / UDP workers for version 3.1.x
by Francesco Castellano 08 Aug '11

08 Aug '11

Dear sirs, searching on the web and in old ML threads about the suggested number of TCP and UDP workers, I just found: - for TCP children (i.e., TCP receivers, isn't it?) "As a rule of thumb, (maximum simultaneous connections)/2000 should be OK" (from doc/tcp_tunning.txt) - In general for workers (from a past thread by Henning Westerholt): "In my experience the number of children is not that important for performance, you may just choose the default size of 8." In the last post there was, … [View More]

2 2

Kamailio Redundant Systems
by HM Kias 08 Aug '11

08 Aug '11

Hi, Anyone tried pacemaker, heartbeat or OpenAIS for building redundant kamailio systems? Please share your experiences. Also auggest a better solution. Regards, -- HM Kias 91-9443467600

2 1

Homer Project
by Alexandr Dubovikov 05 Aug '11

05 Aug '11

Hello sr-users, I would like to present a new project called "Homer". What is Homer ? Homer is a capture solution for SIP logging. How does it work? Homer has 4 components: 1. Capture Agent 2. Capture Node 4. Capture DB 3. Web GUI. The capture agent takes your local SIP traffic and sends it to the defined capture node in HEP protocol. If you are user of FreeSWITCH or Kamailio, use build-in capture agent. In other case, you can use an external generic daemon, but only for SIP/UDP. The … [View More]

3 3

$du gets set automatically before branch_route
by Alex Hermann 05 Aug '11

05 Aug '11

Hello, while trying to upgrade from 1.5 to 3.1 i noticed that between t_relay() and branch_route, $du gets automatically set to $ru if it has not been set before. This is a change from 1.5 i could not find documentation for, so it might have been unintened. Can someone enlighten me on the purpose of this change and if there is a way to determine if $du is (was) unset other than comparing $ru to $du, wich seems a bit inefficient? Is it even guaranteed that $du gets set to $ru in every … [View More]

3 8

udp vs. tcp receiver processes
by Juha Heinanen 05 Aug '11

05 Aug '11

i have 8 children specified in default file and in cfg file i have: listen=192.98.101.10:5060; listen=udp:127.0.0.1:5070; listen=udp:127.0.0.1:5070; listen=tcp:127.0.0.1:5070; listen=udp:127.0.0.1:5080; listen=tcp:127.0.0.1:5080; listen=tcp:127.0.0.1:6060; that results in 8 tcp receiver processes but 3 x 8 udp receiver processes: # sip-proxy_ctl ps 30720 attendant 30721 udp receiver child=0 sock=192.98.101.10:5060 30722 udp receiver child=1 sock=192.98.101.10:5060 30723 udp receiver child=2 … [View More]

3 4

Kamailio on a server with different network cards
by Rocío Ruz 05 Aug '11

05 Aug '11

Hello, My server has two different network cards with different IPs. I want kamailio to use both; the network card to be used depends on the destination. My problem is that I have to add an IP in the configuration file kamailio.cfg: listen=109.239.50.67 Will Kamailio listen only to packets sent to this IP? Can I add the two IPs corresponding to my two network cards here? Thank you RR

3 5

480 Temporarily Unavailable. in kamailio 3.1.4
by hala alramli 04 Aug '11

04 Aug '11

i installed kamailio and freeswitch according to http://kb.asipto.com/freeswitch:kamailio-3.1.x-freeswitch-1.0.6d-sbc i installed also rtpproxy . i managed to create 101,102,103 and all of them registered , i am not able to make calls between them, 480 temporarily unavailable always the error . i cant find in the configuration file what cause this error. this is the track. i will be thankful for any help . Hala interface: eth0 (192.168.1.0/255.255.255.0) filter:… [View More] (ip or ip6) and ( port 5060 ) match: 101 U 2011/08/04 13:16:57.351872 192.168.1.175:5060 -> 192.168.1.233:5060 INVITE sip:102@192.168.1.233 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.175:5060;branch=z9hG4bK-6eabf9b3. From: <sip:101@192.168.1.233>;tag=43c22ff96586a71ao0. To: <sip:102@192.168.1.233>. Call-ID: 9acd1ca5-9162c25e(a)192.168.1.175. CSeq: 101 INVITE. Max-Forwards: 70. Contact: <sip:101@192.168.1.175:5060>. Expires: 240. User-Agent: Linksys/PAP2T-5.1.6(LS). Content-Length: 444. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: x-sipura, replaces. Content-Type: application/sdp. . v=0. o=- 986176 986176 IN IP4 192.168.1.175. s=-. c=IN IP4 192.168.1.175. t=0 0. m=audio 16450 RTP/AVP 8 0 2 4 18 96 97 98 100 101. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:4 G723/8000. a=rtpmap:18 G729a/8000. a=rtpmap:96 G726-40/8000. a=rtpmap:97 G726-24/8000. a=rtpmap:98 G726-16/8000. a=rtpmap:100 NSE/8000. a=fmtp:100 192-193. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:30. a=sendrecv. U 2011/08/04 13:16:57.352037 192.168.1.233:5060 -> 192.168.1.175:5060 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 192.168.1.175:5060;branch=z9hG4bK-6eabf9b3. From: <sip:101@192.168.1.233>;tag=43c22ff96586a71ao0. To: <sip:102@192.168.1.233>;tag=b27e1a1d33761e85846fc98f5f3a7e58.7d11. Call-ID: 9acd1ca5-9162c25e(a)192.168.1.175. CSeq: 101 INVITE. Proxy-Authenticate: Digest realm="192.168.1.233", nonce="TjqA1U46f6liNM8a6LX5HBJ2wXEBtOIX". Server: kamailio (3.1.4 (x86_64/linux)). Content-Length: 0. . U 2011/08/04 13:16:57.370381 192.168.1.175:5060 -> 192.168.1.233:5060 ACK sip:102@192.168.1.233 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.175:5060;branch=z9hG4bK-6eabf9b3. From: <sip:101@192.168.1.233>;tag=43c22ff96586a71ao0. To: <sip:102@192.168.1.233>;tag=b27e1a1d33761e85846fc98f5f3a7e58.7d11. Call-ID: 9acd1ca5-9162c25e(a)192.168.1.175. CSeq: 101 ACK. Max-Forwards: 70. Contact: <sip:101@192.168.1.175:5060>. User-Agent: Linksys/PAP2T-5.1.6(LS). Content-Length: 0. . U 2011/08/04 13:16:57.380065 192.168.1.175:5060 -> 192.168.1.233:5060 INVITE sip:102@192.168.1.233 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.175:5060;branch=z9hG4bK-eab8c613. From: <sip:101@192.168.1.233>;tag=43c22ff96586a71ao0. To: <sip:102@192.168.1.233>. Call-ID: 9acd1ca5-9162c25e(a)192.168.1.175. CSeq: 102 INVITE. Max-Forwards: 70. Proxy-Authorization: Digest username="101",realm="192.168.1.233",nonce="TjqA1U46f6liNM8a6LX5HBJ2wXEBtOIX",uri="sip:102@192.168.1.233",algorithm=MD5,response="a019828b7a7334539ffa6b668dfcfada". Contact: <sip:101@192.168.1.175:5060>. Expires: 240. User-Agent: Linksys/PAP2T-5.1.6(LS). Content-Length: 444. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: x-sipura, replaces. Content-Type: application/sdp. . v=0. o=- 986176 986176 IN IP4 192.168.1.175. s=-. c=IN IP4 192.168.1.175. t=0 0. m=audio 16450 RTP/AVP 8 0 2 4 18 96 97 98 100 101. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:4 G723/8000. a=rtpmap:18 G729a/8000. a=rtpmap:96 G726-40/8000. a=rtpmap:97 G726-24/8000. a=rtpmap:98 G726-16/8000. a=rtpmap:100 NSE/8000. a=fmtp:100 192-193. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:30. a=sendrecv. U 2011/08/04 13:16:57.380329 192.168.1.233:5060 -> 192.168.1.175:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 192.168.1.175:5060;branch=z9hG4bK-eab8c613. From: <sip:101@192.168.1.233>;tag=43c22ff96586a71ao0. To: <sip:102@192.168.1.233>. Call-ID: 9acd1ca5-9162c25e(a)192.168.1.175. CSeq: 102 INVITE. Server: kamailio (3.1.4 (x86_64/linux)). Content-Length: 0. . U 2011/08/04 13:16:57.488511 192.168.1.233:5060 -> 192.168.1.175:5060 SIP/2.0 480 Temporarily Unavailable. Via: SIP/2.0/UDP 192.168.1.175:5060;branch=z9hG4bK-eab8c613. From: <sip:101@192.168.1.233>;tag=43c22ff96586a71ao0. To: <sip:102@192.168.1.233>;tag=NDrNpDZ1g0jZe. Call-ID: 9acd1ca5-9162c25e(a)192.168.1.175. CSeq: 102 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-5d37bc9 2011-08-03 09-05-01 -0500. From: <sip:101@192.168.1.233>;tag=43c22ff96586a71ao0. To: <sip:102@192.168.1.233>. Call-ID: 9acd1ca5-9162c25e(a)192.168.1.175. CSeq: 102 INVITE. Server: kamailio (3.1.4 (x86_64/linux)). Content-Length: 0. . U 2011/08/04 13:16:57.488511 192.168.1.233:5060 -> 192.168.1.175:5060 SIP/2.0 480 Temporarily Unavailable. Via: SIP/2.0/UDP 192.168.1.175:5060;branch=z9hG4bK-eab8c613. From: <sip:101@192.168.1.233>;tag=43c22ff96586a71ao0. To: <sip:102@192.168.1.233>;tag=NDrNpDZ1g0jZe. Call-ID: 9acd1ca5-9162c25e(a)192.168.1.175. CSeq: 102 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-5d37bc9 2011-08-03 09-05-01 -0500. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, precondition, path, replaces. Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Reason: Q.850;cause=16;text="NORMAL_CLEARING". Content-Length: 0. Remote-Party-ID: "kb-102" <sip:kb-102@192.168.1.233>;party=calling;privacy=off;screen=no. . U 2011/08/04 13:16:57.502085 192.168.1.175:5060 -> 192.168.1.233:5060 ACK sip:102@192.168.1.233 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.175:5060;branch=z9hG4bK-eab8c613. From: <sip:101@192.168.1.233>;tag=43c22ff96586a71ao0. To: <sip:102@192.168.1.233>;tag=NDrNpDZ1g0jZe. Call-ID: 9acd1ca5-9162c25e(a)192.168.1.175. CSeq: 102 ACK. Max-Forwards: 70. Proxy-Authorization: Digest username="101",realm="192.168.1.233",nonce="TjqA1U46f6liNM8a6LX5HBJ2wXEBtOIX",uri="sip:102@192.168.1.233",algorithm=MD5,response="a019828b7a7334539ffa6b668dfcfada". Contact: <sip:101@192.168.1.175:5060>. User-Agent: Linksys/PAP2T-5.1.6(LS). Content-Length: 0. . [View Less]

2 1

How to reset t_on_failure/branch?
by Iñaki Baz Castillo 04 Aug '11

04 Aug '11

Hi, I remember that in Kamailio 1.5.X t_on_branch and t_on_failure still remain after rouning failure_route[XX], and I "think" there was a way to reset them. In 3.X it seems that t_on_branch and t_on_failure must be set again in failure_route to take effect in next t_relay() call, am I right? Thanks. -- Iñaki Baz Castillo <ibc(a)aliax.net>

2 2

AVPs missing in onreply_route
by Sebastian Damm 04 Aug '11

04 Aug '11

Hi, I'm still fighting problems in our Kamailio 1.5 to 3.1 migration. Right now I'm stuck with a strange behavior of AVPs in my dialplan. I set a lot of AVPs for accounting in the route directive, and I just inserted an avp_print() statement right before t_relay() is called, all values are set. Then I call avp_print() as the first statement of onreply_route(), and it doesn't print out anything. I call it again at the end of onreply_route, and the only AVP printed out is the one I set directly … [View More]

3 9

ERROR : when create database in kamailio 3.1.0
by hala alramli 04 Aug '11

04 Aug '11

hello i a trying to instal kamailio and free switch according to the tutorial in http://kb.asipto.com/freeswitch:kamailio-3.1.x-freeswitch-1.0.6d-sbc but when i try to create the database and the extra tables this error apear could you help me? thanks Hala root@noga1:~# nano /usr/local/etc/kamailio/kamctlrc root@noga1:~# /usr/local/sbin/kamdbctl create MySQL password for root: database engine 'mysql' loaded INFO: test server charset INFO: creating database kamailio .… [View More]

2 2

Re: [SR-Users] Chicago, ClueCon next week
by Daniel-Constantin Mierla 03 Aug '11

03 Aug '11

Hello, for questions about kamailio (openser), the best place to address them is still the mailing list for users. There are many people there that might be able to answer quickly. Messages sent personally to me for technical questions about kamailio (openser) are discarded, unless they were a result of a discussion on the mailing list where I asked for sensitive data. Cheers, Daniel On 8/3/11 1:47 AM, ifeanyi okoye wrote: > hi Daniel..i would like to discuss some of the issues am … [View More]

1 0

Chicago, ClueCon next week
by Daniel-Constantin Mierla 03 Aug '11

03 Aug '11

Hello, I'll be in Chicago few days next week for ClueCon conference. If you are in the area, but not attending the ClueCon conference, drop me a message in case you want to meet. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

4 5

Kamailio 3.1.4 Bitwise Operators?
by Brandon Armstead 02 Aug '11

02 Aug '11

Hello, It seems as if the bitwise operations in Kamailio 3.1.4 are broken? Please correct me if I am doing something wrong - however simple evaluations such as: $avp(s:my-test-bit) = 10; $avp(s:my-test-bit) = (int)$avp(s:my-test-bit) & ~ 2; Seems to result in $avp(s:my-test-bit) being equal to "2" rather than "8" Any thoughts / Suggestions / Fixes? I also wonder if this has any relation to !~ not working? Thanks for all of your help in advance! P.S. I sent this both to user list … [View More]

2 3

Re: [SR-Users] 10 years of SER - statistics
by Javier Gallart 02 Aug '11

02 Aug '11

Hi Daniel a few numbers here...we use Kamailio in a carrier environment (no subscribers, no nat...), basically as proxy and redirect server. We use it in production and also as one of the foundations of other features still under development. As a redirect server, on average it handles around 600k-700k calls a day. Regards Javi > ------------------------------ > > Message: 4 > Date: Tue, 02 Aug 2011 11:23:26 +0200 > From: Daniel-Constantin Mierla <miconda(a)gmail.com> … [View More]

1 0

Carrierroute Compilation error Kamailio 3.1.4
by Omar 01 Aug '11

01 Aug '11

I am Using Solaris 10 and GCC 4.2, i do have libconfuse everything goes fine until this error comes up. carrierroute.c:54: error: 'MODULE_INTERFACE_VER' undeclared here (not in a function) carrierroute.c:87: error: expected '=', ',', ';', 'asm' or '__attribute__' before 'cmds' carrierroute.c:132: error: variable 'exports' has initializer but incomplete type carrierroute.c:133: warning: excess elements in struct initializer carrierroute.c:133: warning: (near initialization for 'exports') … [View More]

3 8

Detect phone and fax
by Linux Guy 31 Jul '11

31 Jul '11

Hi, Is there any way to separate detect phone and fax calls in kamailio ? I understand that kamailio is a signaling proxy only but is there any way we can detect if a call is a fax or phone ? Thanks Linux Guy..

2 1

what is tcp_accept_no_cl core param?
by Juha Heinanen 31 Jul '11

31 Jul '11

xcap_server module requires tcp_accept_no_cl=yes i didn't find that param defined in core cookbook. what does it do? -- juha

2 1

Set CLI.
by Henrik Aagaard Sørensen 29 Jul '11

29 Jul '11

Is it possible to set CLI via the usr_preferences table in Kamailio? Or any other way? So that certain subscribers get there from-number overwritten with a specific one.

3 2

[Urgent] Double REGISTER and MESSAGE issue in Kamailio3.1
by Coca 29 Jul '11

29 Jul '11

Dear List, I have Kamailio 3.1 installed in Amazon EC 2 server. Everything is working fine except that every REGISTER / MESSAGE was looped for twice or 3 times. And I am sure this is not caused by UA. I have attached the ngrep log in this message. And also, my main modification to the kamailio.cfg is like below: Your help will be great great appreciated. Thank you!! Coca ---------------------------------- #!define WITH_MYSQL #!define WITH_NAT alias="mydomain.org" alias="xx.xx.xx.xx"… [View More]

1 0

Problem using textops API
by Timo Klecker 29 Jul '11

29 Jul '11

Hi, I'm trying to extend a Kamailio Module and need to add and remove headers in a SIP Message. To do this I want to use the textops module and I am importing the api and trying to bind the textops module. #include "../textops/api.h" [..] struct textops_binds tob; [..] /* register callbacks to TEXTOPS */ if (load_textops_api(&tob)!=0) { LM_ERR("can't load textops api. Is module textops loaded?\n"); return -1; } Later I use tob.append_hf … [View More]

2 3

problem with siptrace outgoing SIP message logging
by Peter Kaszas 29 Jul '11

29 Jul '11

Dear List members, Im quite new to kamailio, and i got a problem with the siptrace module. I read that siptrace module can capture and log all incoming and outgoing sip messages. So i put to the config file the following lines: loadmodule "siptrace.so" . . . #-------- siptrace paramas ----- modparam("siptrace", "db_url", DBURL) modparam("siptrace", "table", "sip_trace") modparam("siptrace", "trace_flag", 18) modparam("siptrace", "trace_on", 1) modparam("siptrace", "trace_sl_acks", 1) . . . … [View More]

2 3

Disk space went full in 3 days. Kamailio 3.1 @debian64
by Coca 29 Jul '11

29 Jul '11

Hi List , I have newly installed Kamailio 3.1 on a debian 64 server , replacing the OpenSIPs server And the subscribers are about 1000 people. After running 3 days, the disk with 8GB became full and caused Mysql stopped. I have checked /etc/my.cnf , and the setting about binary log is like: #log_bin = /var/log/mysql/mysql-bin.log expire_logs_days = 10 max_binlog_size = 100M It seems that binaly log for mysql is disabled. What else possible reason could … [View More]

2 2

Kamctl and Postgresql
by Nathaniel L Keeling 29 Jul '11

29 Jul '11

I was using kamctl on Solaris to add a table entry into a postgresql database table and received this error: # ./kamctl domain showdb psql: warning: extra command-line argument "-c" ignored psql: warning: extra command-line argument "select * FROM domain ; " ignored After some research, I found in kamctl.pgsql the functions: # input: sql query, optional pgsql command-line params pgsql_query() { # if password not yet queried, query it now prompt_pw "PgSQL password for user '… [View More]

2 1

About RTimer module
by Iñaki Baz Castillo 28 Jul '11

28 Jul '11

Hi RTimer module doc says: ---------- The module executes route blocks on a timer base. It can create new timer processes and execute many route blocks on same timer. ---------- What does it mean "It can create new timer processes"? Let's suppose I run a blocking call (i.e. sending an HTTP request which takes too long) from a route block called by a timer of RTimer. What would it occur? Thanks. -- Iñaki Baz Castillo <ibc(a)aliax.net>

3 5

Kamailio - ISC interface
by Rocío Ruz 28 Jul '11

28 Jul '11

Hello, I was wondering if it would be possible to plug an Application Server on Kamilio solution using the ISC interface (other than the AS module provided with Kamailio solution). Would it be mandatory to use IMS extensions to do so? Thank you. RR

3 4

Kamctl and Postgresql
by Nathaniel L Keeling 28 Jul '11

28 Jul '11

I was using kamctl on Solaris to add a table entry into a postgresql v9.0 database table and received this error: # ./kamctl domain showdb psql: warning: extra command-line argument "-c" ignored psql: warning: extra command-line argument "select * FROM domain ; " ignored After some research, I found in kamctl.pgsql the functions: # input: sql query, optional pgsql command-line params pgsql_query() { # if password not yet queried, query it now prompt_pw "PgSQL password for … [View More]

1 0

Filtering unregister messges
by Lucas Alvarez 28 Jul '11

28 Jul '11

Hi, how can I do to filter the unregister messages? method == "REGISTER" && ??? Thanks ins advance. Lucas

2 1

Kamailio - IMS extensions
by Rocío Ruz 27 Jul '11

27 Jul '11

Hello, I would like to use kamailio and its IMS extensions. Is it mandatory to install and run each module (s-cscf, i-cscf, p-cscf, Hss) on different machines? Could I just install/run all the modules on the same VM? Thank you. RR

2 1

issue tracking system
by Daniel-Constantin Mierla 27 Jul '11

27 Jul '11

Hello, sightly off-topic and cc-ed to users mailing lists, but hopefully there will be people giving valuable feedback based on their experience. When we started the integration of source code for kamailio and ser, we went for FlySpray as issue tracking system (aka bug tracker). The project seems to be dead, nothing happened to it for long time, no updates and there are warnings using latest php versions, that may result in malfunctionality or no-functionality with future versions. … [View More]

12 15

Content of $sndto(ip) is "ip:port"
by Andreas Granig 27 Jul '11

27 Jul '11

Hi, Looks like in kam-3.1 the value of $sndto(ip) in onsend_route is "ip:port", e.g. "1.2.3.4:5060" instead of just "1.2.3.4". Am I the only one experiencing this issue? Andreas

2 2

Understanding alias_db correctly.
by Henrik Aagaard Sørensen 27 Jul '11

27 Jul '11

Can someone help me by explaining what the alias_db exactly does: http://www.kamailio.org/docs/modules/stable/modules_k/alias_db.html As I've understood it, I can add several aliases to my subscribers, for example: 100(a)my-domain.com can have the alias 600(a)my-domain.com and therefore a client can login with the username 600(a)my-domain.com, but really have the preferences of 100(a)my-domain.com as that is the real subscriber. But it doesn't seems to work.

3 3

load_credentials into non-AVP PVs
by Alex Balashov 27 Jul '11

27 Jul '11

When I try this: modparam("auth_db", "load_credentials", "$var(x)=some_col"); I get a parse error: malformed or non AVP $var(customer_bg_id) AVP definition Is there any particular reason these days why auth_db columns have to be loaded to AVPs only? -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/

2 1

Kamailio Advanced Training
by Roman Dmytriv 27 Jul '11

27 Jul '11

Hello Daniel, Are you planning to organize Kamailio Advanced Training on American soil in the near future? Regards Roman

1 1

kamailio password length
by Jon Bonilla 27 Jul '11

27 Jul '11

I have a running kamailio with db backend and I'd like to use 40 character strings as subscriber passwords. Would be enought to increase the db field (actually varchar25) or is there any other lenght limit in kamailio's code? thanks Jon

2 1

Kamailio-IMS-Packages
by Rocío Ruz 27 Jul '11

27 Jul '11

Hello, I would like to know if Kamailio IMS extension RPM packages are available for Red Hat distributions. If they are available, I would like to know the repository. For the moment I have only found Debian packages: http://www.kamailio.org/w/2011/01/ims-extensions-available-for-testing/ but I am having problems with the repository (http://repository.ng-voice.comseems to be not available). Could anybody please help me? Thank you RR

2 1

Kamailio Multiple Source IP
by Linux Guy 27 Jul '11

27 Jul '11

Hi, How can I set a specific source ip for outbound calls when I have multiple Ips that kamailio is listening on. For Eg. I have a table with Caller ID ( Phone Numbers ) Call Enters Kamailio > Check the table >> If the phone number is found >> SOURCE IP : 1.1.1.1 If the phone number is not found >> Source IP : 2.2.2.2 Kamailio is listening on both 1.1.1.1 and 2.2.2.2 . Thanks. Linux Guy..

2 1

bypass rtp traffic.
by MingHon 26 Jul '11

26 Jul '11

Hi List, i would like to know is it possible to bypass the rtp traffic forwarding to asterisk server? my kamailio and rtpproxy is on the same box and asterisk is on the other box. can kamailio/rtpproxy handle the rtp traffic without forwarding to asterisk box? thanks in advance. -- Regards, MingHon

5 14

Kamailio without sound, what to do?
by Henrik Aagaard Sørensen 26 Jul '11

26 Jul '11

I'm a newbee in the world of Kamailio. I've managed to setup a fresh installation of Kamailio, with authentication etc. (with help from some great guys on this mailing-list). Everything seems to work with registers, calls etc. except that there is no sound on my calls. How do I figure out what the problem is?

2 3

Re: [SR-Users] Avoiding to send 603 DECLINED to un-trusted IPs - Is that possible with Kamilio?
by Andrade Ricardo (CI/AFU1) 25 Jul '11

25 Jul '11

Hello Bruce, Strangely, I've never gotten an Anonymous INVITE rejected by Asterisk with 603, but that was the version 1.4. Let me try to help with your request for a solution with Kamailio. These few lines could be added to your main route, after the place where your handle the registration. It will block any request from unregistered users: Route { ... if (!registered("location")) { { exit; } ... } I am assuming that … [View More]

2 1

How to get mem_summary (SIGUSR1) on a worker process?
by Andrade Ricardo (CI/AFU1) 25 Jul '11

25 Jul '11

Hi list, I'm running out of private memory in Kamailio 3.1.0. However, before simply increasing it from the current 4 MB to some higher value, I'd like to know how much is really in use for every process and possibly monitor it. My expectation is that I should be able to send the SIGUSR1 to any of the processes shown by the "sercmd ps" command, and get the trace. But it only succeeds if I send the signal to the "attendant" process. Some more info: # sercmd core.info { version: … [View More]

1 0

Fresh Kamailio installation doesn't require authorization.
by Henrik Aagaard Sørensen 25 Jul '11

25 Jul '11

I've installed Kamailio on Ubuntu 10.04 LTS via: http://www.kamailio.org/dokuwiki/doku.php/packages:debs Updated /etc/kamailio/kamctlrc with: SIP_DOMAIN=MY-IP DBENGINE=MYSQL I've also installed the kamailio-mysql-modules via apt-get. Then I've run kamdbctl create and created the MySQL database. After that I've started Kamailio via kamctl start. Success. At last I've added two users: kamctl add 100 PASS1 kamctl add 101 PASS2 Everything seems to work, but I've found out that everyone can … [View More]

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