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37457 discussions

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-router project you can find the latest version. If you need any guidance to probe this ebuild under your gentoo linux do not hesitate to contact me or provide some feedback here at the lists. Best regards, Claudio

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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 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; } }

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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: 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. ********************************************************************************

3 3

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

2 1

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

3 5

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 Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

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 ngrep (or wireshark) and past it here? > > Cheers, > Daniel > > > On 9/22/11 8:07 PM, Krishna Kurapati wrote: >> 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 <mailto:krish_kura@sip.org> >> is not allowed? >> >> Thanks >> Krish Kura >> >> >> >> _______________________________________________ >> 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

2 4

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 from scratch, incoming message is lost and no longer available to check the sip trace flag. Now, the question is how to decide whether the cancel is to be siptraced. I can lookup for incoming CANCEL and check if it has the siptrace flag set, but there are CANCELs in case of local timeout or parallel forking, so I think it is better to actually lookup the INVITE transaction and if the siptrace flag is set for it, record the cancel. If there is no INVITE transaction anymore, the cancel is forwarded stateless and can be captured in onsend route, although it should be normally discarded if it is known that the proxy works in statefull mode only. 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

2 2

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 filled? Can I achieve this scenario with the carrierroute module ? Thanks. Mino

1 0

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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